Book-length PDF: Cisco IOS Voice, Video, and Fax

Book-length PDF: Cisco IOS Voice, Video, and Fax
Cisco IOS
Voice, Video, and Fax
Configuration Guide
Release 12.2
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Cisco IOS Voice, Video, and Fax Configuration Guide
Copyright © 2001–2006, Cisco Systems, Inc.
All rights reserved.
CONTENTS
About Cisco IOS Software Documentation
Documentation Objectives
Audience
xxxi
xxxi
xxxi
Documentation Organization xxxi
Documentation Modules xxxi
Master Indexes xxxiv
Supporting Documents and Resources
New and Changed Information
Document Conventions
xxxiv
xxxv
xxxv
Obtaining Documentation xxxvii
World Wide Web xxxvii
Ordering Documentation xxxvii
Documentation Feedback
xxxvii
Obtaining Technical Assistance xxxviii
Cisco.com xxxviii
Technical Assistance Center xxxviii
Contacting TAC by Using the Cisco TAC Website
Contacting TAC by Telephone xxxix
Using Cisco IOS Software
xxxviii
xli
Understanding Command Modes
xli
Getting Help xlii
Example: How to Find Command Options
xliii
Using the no and default Forms of Commands
xlv
Saving Configuration Changes
xlvi
Filtering Output from the show and more Commands
xlvi
Identifying Supported Platforms xlvii
Using Feature Navigator xlvii
Using Software Release Notes xlvii
Cisco IOS Voice, Video, and Fax Configuration Guide
iii
Contents
Voice, Video, and Fax Overview
VC-1
Configuration Guide Overview VC-1
Dial Peers VC-1
Voice Ports VC-2
Voice Technologies VC-3
Voice over IP VC-3
Voice over Frame Relay VC-4
Voice over ATM VC-4
H.323 Gateways VC-5
Media Gateway Control Protocol VC-5
Session Initiation Protocol VC-5
Interactive Voice Response VC-6
Multimedia Conference Manager VC-7
Video VC-7
Fax Gateways VC-8
Configuring Voice over IP
Voice over IP Overview
VoIP Benefits
VC-9
VC-9
VC-12
VoIP Call Processing
VoIP Prerequisite Tasks
VC-12
VC-13
VoIP Network Design Considerations VC-14
VoIP Quality of Service Tips VC-14
Delay VC-14
Jitter VC-15
Serialization VC-15
Bandwidth Consumption VC-15
VoIP Configuration Task List
VC-15
Configuring VoIP over Frame Relay
VC-17
VoIP Configuration Examples VC-18
VoIP over Frame Relay Configuration Example VC-18
VoIP for the Cisco 3600 Series Configuration Examples VC-19
FXS-to-FXS Connection Using RSVP VC-19
Linking PBX Users with E&M Trunk Lines VC-22
PSTN Gateway Access Using FXO Connection VC-24
PSTN Gateway Access Using FXO Connection (PLAR Mode)
VoIP for the Cisco AS5300 Configuration Example VC-26
Cisco IOS Voice, Video, and Fax Configuration Guide
iv
VC-25
Contents
Linking PBX Users to a T1 ISDN PRI Interface VC-26
VoIP for the Cisco AS5800 Configuration Example VC-29
Configuring the Cisco 3640 As a Gatekeeper VC-29
Configuring the Cisco 2600 As a Gateway VC-30
Configuring the Cisco AS5800 as a Gateway VC-30
BASIC VOICE CONFIGURATION
Configuring Voice Ports
VC-35
Voice Port Configuration Overview VC-36
Telephony Signaling Interfaces VC-37
FXS and FXO Interfaces VC-38
E&M Interfaces VC-39
Analog Voice Ports Configuration Task List VC-40
Prerequisites for Configuring Analog Voice Ports VC-41
Preparing to Configure Analog Voice Ports VC-41
Configuring Platform-Specific Analog Voice Hardware VC-43
Cisco 800 Series Routers VC-43
Cisco 1750 Modular Router VC-43
Cisco 2600 Series and Cisco 3600 Series Routers VC-44
Cisco MC3810 Multiservice Concentrator VC-44
Configuring Codec Complexity for Analog Voice Ports on the Cisco MC3810 with High-Performance
Compression Modules VC-45
Configuring Basic Parameters on Analog FXO, FXS, or E&M Voice Ports VC-46
Configuring Analog Telephone Connections on Cisco 803 and 804 Routers VC-50
Verifying Analog Telephone Connections on Cisco 803 and 804 Routers VC-52
Troubleshooting Tip for Cisco 803 and 804 Routers VC-54
Configuring Digital Voice Ports VC-54
Prerequisites for Configuring Digital Voice Ports VC-55
Preparing Information to Configure Digital Voice Ports VC-56
Platform-Specific Digital Voice Hardware VC-58
Cisco 2600 Series and Cisco 3600 Series Routers VC-58
Cisco MC3810 Multiservice Concentrator VC-59
Cisco AS5300 Universal Access Server VC-60
Cisco AS5800 Universal Access Server VC-60
Cisco 7200 and Cisco 7500 Series Routers VC-61
Configuring Basic Parameters on Digital T1/E1 Voice Ports VC-61
Configuring Codec Complexity for Digital T1/E1 Voice Ports VC-62
Cisco IOS Voice, Video, and Fax Configuration Guide
v
Contents
Configuring Controller Settings for Digital T1/E1 Voice Ports VC-65
Configuring Basic Voice Port Parameters for Digital T1/E1 Voice Ports VC-76
Fine-Tuning Analog and Digital Voice Ports VC-79
Auto Cut-Through Command VC-80
Bit Modification Commands for Digital Voice Ports VC-80
Calling Number Outbound Commands VC-82
Disconnect Supervision Commands VC-83
FXO Supervisory Disconnect Tone Commands VC-85
Timeouts Commands VC-87
Timing Commands VC-89
DTMF Timer Inter-Digit Command for Cisco AS5300 Access Servers VC-90
Voice Activity Detection Commands Related to Voice-Port Configuration Mode VC-91
Voice Quality Tuning Commands VC-92
Delay in Voice Networks VC-92
Jitter Adjustment VC-92
Echo Adjustment VC-94
Voice Level Adjustment VC-96
Verifying Analog and Digital Voice-Port Configurations VC-97
show voice port summary Command Examples VC-98
Cisco 3640 Router Analog Voice Port VC-99
Cisco MC3810 Multiservice Concentrator Digital Voice Port VC-99
show voice port Command Examples VC-99
Cisco 3600 Series Router Analog E&M Voice Port VC-99
Cisco 3600 Series Router Analog FXS Voice Port VC-100
Cisco 3600 Series Router Digital E&M Voice Port VC-101
Cisco AS5300 Universal Access Server T1 CAS Voice Port VC-101
Cisco 7200 Series Router Digital E&M Voice Port VC-102
show controller Command Examples VC-103
Cisco 3600 Series Router T1 Controller VC-103
Cisco MC3810 Multiservice Concentrator E1 Controller VC-103
Cisco AS5800 Universal Access Server T1 Controller VC-103
show voice dsp Command Examples VC-104
show voice call summary Command Examples VC-105
Cisco MC3810 Multiservice Concentrator Analog Voice Port VC-105
Cisco 3600 Series Router Digital Voice Port VC-105
show call active voice Command Example VC-105
show call history voice Command Example VC-106
Troubleshooting Analog and Digital Voice Port Configurations
Cisco IOS Voice, Video, and Fax Configuration Guide
vi
VC-108
Contents
Troubleshooting Chart VC-108
Voice Port Testing Commands VC-110
Detector-Related Function Tests VC-110
Loopback Function Tests VC-112
Tone Injection Tests VC-113
Relay-Related Function Tests VC-114
Fax/Voice Mode Tests VC-114
Configuring Dial Plans, Dial Peers, and Digit Manipulation
Dial Plan Overview
VC-117
VC-117
Dial Peer Overview VC-118
Inbound and Outbound Dial Peers VC-119
Destination Pattern VC-120
Fixed- and Variable-Length Dial Plans VC-122
Session Target VC-123
Digit Stripping on Outbound POTS Dial Peers VC-124
Configuring Dial Peers VC-124
Configuring Dial Peers for Call Legs VC-125
Creating a Dial Peer Configuration Table VC-127
Configuring POTS Dial Peers VC-128
Configuring Dial Plan Options for POTS Dial Peers VC-130
Configuring VoIP Dial Peers VC-131
Configuring Codec Selection Order VC-132
Creating a Voice Class to Define Codec Selection Order VC-133
Applying Codec Selection Order to a VoIP Dial Peer VC-133
Configuring Dial Plan Options for VoIP Dial Peers VC-133
Configuring VoFR Dial Peers VC-135
Configuring VoATM Dial Peers VC-135
Verifying POTS and VoIP Dial Peer Configurations VC-135
Troubleshooting Tips VC-136
Dial Peer Overview VC-137
Two-Stage Dialing VC-137
Variable-Length Matching VC-138
Matching Inbound Dial Peers VC-139
Inbound Dial Peers for IVR Applications VC-140
Matching Outbound Dial Peers VC-140
Default Routes for Outbound Call Legs VC-141
Cisco IOS Voice, Video, and Fax Configuration Guide
vii
Contents
Configuring Dial Peer Matching Features VC-141
Answer Address for VoIP VC-142
DID for POTS Dial Peers VC-142
Identifying Voice and Modem Calls VC-144
Hunt Groups and Preferences VC-144
Configuring Dial-Peer Hunting Options VC-146
Numbering Type Matching VC-147
Configuring Numbering-Type Matching VC-148
Class of Restrictions VC-148
Configuring Classes of Restrictions VC-150
Verifying Classes of Restrictions VC-150
Configuring Digit Manipulation VC-151
Digit Stripping and Prefixes VC-151
Forward Digits VC-154
Number Expansion VC-155
Creating a Number Expansion Table VC-156
Configuring Number Expansion VC-157
Verifying Number Expansion VC-157
Digit Translation Rules for VoIP VC-157
Configuring Digit Translation Rules VC-159
Creating Digit Translation Rules VC-159
Applying Translation Rules to Inbound POTS Calls VC-160
Applying Translation Rules to Inbound VoIP Calls VC-161
Applying Translation Rules to Outbound Call Legs VC-161
Verifying Digit Translation VC-162
Configuring Quality of Service for Voice
QoS for Voice Overview
VC-163
VC-163
QoS for Voice Tools VC-164
Edge Functions VC-165
Bandwidth Limitations VC-165
Real-Time Transport Protocol VC-165
Queueing VC-166
Packet Classification VC-167
IP Precedence VC-167
Policy Routing VC-167
RSVP VC-167
VoIP Call Admission Control VC-167
Cisco IOS Voice, Video, and Fax Configuration Guide
viii
Contents
IP RTP Priority
VC-169
Traffic Policing for Voice Networks
VC-169
Traffic Shaping for Voice Networks
VC-170
High-Speed Transport
VC-171
Congestion Avoidance
WRED VC-171
TCP VC-171
VC-171
QoS for Voice Configuration Prerequisites
VC-172
QoS for Voice Configuration Task List VC-172
Configuring Synchronization and the Reservation Timer VC-173
Configuring Slow Connect for VoIP Globally VC-173
Configuring Slow Connect for a Specific Dial Peer VC-174
Verifying the RSVP CAC Configuration VC-174
Monitoring and Maintaining RSVP Call Admission Control
VC-174
QoS for Voice Configuration Examples VC-175
RSVP Synchronization Examples VC-175
H.323 Slow Connect by Voice Service Example VC-176
H.323 Slow Connect by Dial Peer Example VC-176
H.323 SUPPORT AND OTHER VOIP CALL CONTROL SIGNALING
Configuring Media Gateway Control Protocol and Related Protocols
VC-179
MGCP Configuration Overview VC-180
Supported Gateways VC-182
Residential Gateway VC-182
Trunking Gateway VC-183
MGCP Prerequisite Tasks
VC-183
MGCP Configuration Task List VC-184
Configuring a TGW for MGCP VC-184
Configuring a TGW for SGCP VC-186
Configuring an RGW VC-187
Configuring the Cisco Voice Gateway 200 to Support Cisco CallManager
Verifying the TGW or RGW Configuration VC-190
Blocking New Calls and Gracefully Terminating Existing Calls VC-190
Monitoring and Maintaining MGCP VC-190
MGCP Configuration Examples VC-190
Configuring the Cisco AS5300 As a TGW with MGCP: Example
VC-188
VC-191
Cisco IOS Voice, Video, and Fax Configuration Guide
ix
Contents
Configuring the Cisco AS5300 As a TGW with SGCP: Example VC-192
Configuring the Cisco 3660 as a TGW with MGCP: Example VC-193
Configuring the Cisco uBR924 as an RGW: Example VC-194
Configuring the Cisco 2620 as an RGW: Example VC-196
Configuring the Cisco Voice Gateway 200 as an RGW: Example VC-197
H.323 Applications
VC-199
The H.323 Standard VC-200
H.323 Terminals VC-201
H.323 Gateways VC-201
Configuring ISDN Redirect Number Support VC-201
H.323 Proxies VC-202
H.323 Gatekeepers VC-202
Gatekeeper Zones VC-202
MCUs VC-202
How Terminals, Gatekeepers, and Proxies Work Together VC-203
Intrazone Call VC-203
Interzone Call Without Proxy VC-203
Interzone Call with Proxy VC-204
How Terminals, Gatekeepers, and Gateways Work Together VC-205
How Terminals, Gatekeepers, Proxies, and MCUs Work Together VC-206
Intrazone MCU Conference Call VC-207
Interzone MCU Conference Call Without Proxy VC-207
Interzone MCU Conference Call with Proxy VC-208
Call Signaling Procedures VC-209
Call Setup—Both Gateways Registered to the Same Gatekeeper VC-209
Call Termination VC-210
Call Clearing with a Gatekeeper VC-211
H.323 Feature Overview VC-211
Source Call Signal Address VC-212
H.323 Version 2 Support VC-213
Lightweight Registration VC-214
Improved Gateway Selection Process VC-214
Gateway Resource Availability Reporting VC-215
Support for Single-Proxy Configurations VC-215
Registration of E.164 Addresses for Gateway-Attached Devices
Tunneling of Redirecting Number Information Element VC-215
DTMF Relay VC-216
Cisco IOS Voice, Video, and Fax Configuration Guide
x
VC-215
Contents
H.245 Tunneling of DTMF Relay in Conjunction with Fast Connect VC-217
Translation of FXS Hookflash Relay VC-217
H.235 Security VC-219
GKTMP and RAS Messages VC-219
RAS Message Fields VC-220
Multizone Features VC-224
Codec Negotiation VC-225
Supported Codecs VC-225
H.245 Empty Capabilities Set VC-226
H.323 Version 2 Fast Connect VC-226
H.450.2 Call Transfer VC-227
H.450.3 Call Deflection VC-228
Gateway Support for Alternate Endpoints VC-228
Gatekeeper C Code Generic API for GKTMP in a UNIX Environment VC-228
Gateway Support for a Network-Based Billing Number VC-228
Gateway Support for Voice-Port Description VC-229
H.323 Signaling VC-229
In-Band Tones and Announcements VC-229
End-to-End Alerting VC-231
Cut-Through of Voice Path VC-231
H.245 Initiation VC-231
Overlap Dialing VC-232
Configurable Timers in H.225.0 VC-232
Answer Supervision Reporting VC-232
Gateway-to-Gatekeeper Billing Redundancy VC-233
Ecosystem Gatekeeper Interoperability VC-233
AltGKInfo in GRJ Messages VC-234
AltGKInfo in RRJ Messages VC-234
H.323 Restrictions VC-235
H.323 Version 2 Feature Restrictions VC-235
H.323 Signaling Enhancement Feature Restrictions VC-235
Configurable Timers in H.225.0 Restriction VC-236
Source Call Signal Address and H.245 Empty Capabilities Set Restrictions
Ecosystem Gatekeeper Interoperability Restrictions VC-236
H.323 Prerequisite Tasks
VC-236
VC-237
H.323 Configuration Task List VC-238
Configuring Timers in H.225.0 VC-238
Verifying the H.225.0 TCP Timeout Value
VC-239
Cisco IOS Voice, Video, and Fax Configuration Guide
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Contents
Configuring H.245 Tunneling of DTMF Relay in Conjunction with Fast Connect
Configuring H.450 VC-239
Configuring Call Deflection VC-240
Configuring Call Transfer Without Consultation VC-246
Configuring H.323 Gateways
VC-249
H.323 Gateway Prerequisite Tasks
VC-249
H.323 Gateway Configuration Task List VC-250
Identifying a Router Interface As an H.323 Gateway VC-250
Verifying Gateway Interface Configuration VC-252
Configuring Gateway RAS VC-252
Verifying RAS Configuration VC-255
Troubleshooting Tips VC-255
Configuring AAA Functionality on the Gateway VC-255
AAA Authentication VC-255
AAA Accounting VC-256
Verifying AAA and RADIUS Configuration VC-261
Configuring H.235 Gateway Security VC-261
Settlement with the Gatekeeper VC-263
Call Tracking VC-263
Downloading IVR Scripts VC-265
H.235 Gateway Security Configuration Tasks VC-266
Verifying H.235 Gateway Security Configuration VC-268
Configuring Alternate Gatekeeper Support VC-268
Gatekeeper Clustering VC-268
Verifying Configuration of the Alternate Gatekeeper VC-270
Configuring Dual Tone Multifrequency Relay VC-270
Configuring FXS Hookflash Relay VC-274
Configuring Multiple Codecs VC-276
Verifying Multiple Codecs Configuration VC-277
Configuring Rotary Calling Pattern VC-277
Configuring H.323 Support for Virtual Interfaces VC-278
Verifying the Source IP Address of the Gateway VC-279
H.323 Gateway Configuration Examples VC-279
H.323 Gateway RAS Configuration Example VC-280
AAA Functionality on the Gateway Configuration Example
H.323 Gateway Security Configuration Example VC-284
Cisco IOS Voice, Video, and Fax Configuration Guide
xii
VC-281
VC-239
Contents
H.235 Security Example VC-286
Alternate Gatekeeper Configuration Example VC-286
DTMF Relay Configuration Example VC-287
FXS Hookflash Relay Configuration Example VC-287
Multiple Codec Configuration Example VC-287
Rotary Calling Pattern Configuration Example VC-287
H.323 Support for Virtual Interfaces Configuration Example
Configuring H.323 Gatekeepers and Proxies
VC-288
VC-289
Multimedia Conference Manager Overview VC-289
Principal Multimedia Conference Manager Functions
VC-290
H.323 Gatekeeper Features VC-290
Zone and Subnet Configuration VC-291
Redundant H.323 Zone Support VC-291
Gatekeeper Multiple Zone Support VC-291
Gateway Support for Alternate Gatekeepers VC-291
Zone Prefixes VC-291
Technology Prefixes VC-292
Gatekeeper-to-Gatekeeper Redundancy and Load-Sharing Mechanism
Terminal Name Registration VC-293
Interzone Communication VC-293
RADIUS and TACACS+ VC-293
Accounting via RADIUS and TACACS+ VC-293
Interzone Routing Using E.164 Addresses VC-294
HSRP Support VC-296
VC-292
H.323 Proxy Features VC-297
Security VC-297
Proxy Inside the Firewall VC-298
Proxy in Co-Edge Mode VC-299
Proxy Outside the Firewall VC-300
Proxies and NAT VC-300
Quality of Service VC-301
Application-Specific Routing VC-301
H.323 Prerequisite Tasks and Restrictions VC-302
Redundant H.323 Zone Support VC-302
Gatekeeper-to-Gatekeeper Redundancy and Load-Sharing Mechanism
VC-302
Cisco IOS Voice, Video, and Fax Configuration Guide
xiii
Contents
H.323 Gatekeeper Configuration Task List VC-303
Configuring the Gatekeeper VC-303
Starting a Gatekeeper VC-304
Configuring Redundant H.323 Zone Support VC-308
Configuring Local and Remote Gatekeepers VC-309
Configuring Redundant Gatekeepers for a Zone Prefix VC-310
Configuring Redundant Gatekeepers for a Technology Prefix VC-311
Configuring Static Nodes VC-313
Configuring H.323 Users via RADIUS VC-314
Configuring a RADIUS/AAA Server VC-318
Configuring User Accounting Activity for RADIUS VC-320
Configuring E.164 Interzone Routing VC-321
Configuring H.323 Version 2 Features VC-322
Configuring Gatekeeper Triggers for Interaction with External Applications
Configuring the Proxy VC-332
Configuring a Proxy Without ASR VC-333
Configuring a Proxy with ASR VC-337
VC-327
H.323 Gatekeeper Configuration Examples VC-345
Configuring a Gatekeeper Example VC-346
Redundant Gatekeepers for a Zone Prefix Example VC-347
Redundant Gatekeepers for a Technology Prefix Example VC-347
E.164 Interzone Routing Example VC-347
Configuring HSRP on the Gatekeeper Example VC-349
Using ASR for a Separate Multimedia Backbone Example VC-350
Enabling the Proxy to Forward H.323 Packets VC-351
Isolating the Multimedia Network VC-351
Configuring a Co-Edge Proxy with ASR Without Subnetting Example VC-352
Co-Edge Proxy with Subnetting Example VC-354
Configuring an Inside-Edge Proxy with ASR Without Subnetting Example VC-356
Configuring a QoS-Enforced Open Proxy Using RSVP Example VC-357
Configuring a Closed Co-Edge Proxy with ASR Without Subnetting Example VC-359
Defining Multiple Zones Example VC-360
Defining One Zone for Multiple Gateways Example VC-360
Configuring a Proxy for Inbound Calls Example VC-361
Configuring a Proxy for Outbound Calls Example VC-361
Removing a Proxy Example VC-362
H.235 Security Example VC-362
Cisco IOS Voice, Video, and Fax Configuration Guide
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Contents
GKTMP and RAS Messages Example VC-363
Prohibiting Proxy Use for Inbound Calls Example VC-363
Disconnecting a Single Call Associated with an H.323 Gateway Example VC-363
Disconnecting All Calls Associated with an H.323 Gateway Example VC-363
Configuring Session Initiation Protocol for Voice over IP
VC-365
SIP Overview VC-366
Components of SIP VC-366
SIP Clients VC-367
SIP Servers VC-368
How SIP Works VC-368
Using a Proxy Server VC-369
Using a Redirect Server VC-372
SIP Enhancements VC-374
SIP Restrictions and Considerations
SIP Prerequisite Tasks
VC-375
VC-376
SIP Configuration Tasks List VC-376
Configuring SIP Support for VoIP Dial Peers VC-376
Changing the Configuration of the SIP User Agent VC-377
Configuring SIP Call Transfer VC-378
Configuring Gateway Accounting VC-379
Verifying SIP Configuration VC-380
SIP Configuration Examples
VC-381
VOICE OVER LAYER 2 PROTOCOLS
Configuring Voice over Frame Relay
VC-387
VoFR Overview VC-387
VoFR Dial Peers VC-388
Switched Calls VC-389
Tandem Switching VC-389
Dynamic-Switched Calls VC-389
Cisco Trunk Calls VC-389
Permanent Calls VC-390
Frame Relay Fragmentation VC-390
End-to-End FRF.12 Fragmentation VC-391
Frame Relay Fragmentation Using FRF.11 Annex C
Cisco Proprietary Voice Encapsulation VC-392
VC-392
Cisco IOS Voice, Video, and Fax Configuration Guide
xv
Contents
Map Classes and Voice Packet Queues
Traffic Shaping VC-392
VoFR Prerequisite Tasks
VC-392
VC-393
VoFR Configuration Task List VC-393
Configuring Frame Relay to Support Voice VC-393
Configuring a Map Class to Support Voice Traffic VC-394
Configuring a Map Class for Traffic-Shaping Parameters VC-395
Configuring VoFR Dial Peers VC-395
Configuring Switched Calls VC-400
Tandem Switching of Switched Calls VC-402
Configuring Cisco Trunk Calls VC-404
Configuring FRF.11 Trunk Calls VC-406
Verifying the Voice Connections VC-407
Verifying the Frame Relay Configuration VC-407
Troubleshooting Tips VC-408
Monitoring and Maintaining the VoFR Configuration VC-408
VoFR Configuration Examples VC-409
Two Routers Using Frame Relay Fragmentation Example VC-409
Two Routers Using a VoFR PVC Example VC-410
Router Using VoFR PVCs Connected to Cisco MC3810s Before 12.1(2)T Example
Cisco Trunk Calls Between Two Routers Example VC-411
FRF.11 Trunk Calls Between Two Routers Example VC-412
Tandem Configuration Examples VC-413
Cisco Trunk Call with Hunt Groups Example VC-418
Configuring Voice over ATM
VC-421
VoATM Overview VC-421
AAL Technology VC-422
Variable Bit Rate Real-Time Options for Traffic Shaping VC-422
Cisco Trunk Calls on Cisco MC3810 Multiservice Concentrators VC-423
VoATM Dial Peers VC-423
VoATM Restrictions
VC-425
VoATM Prerequisite Tasks
VC-425
VoATM Configuration Task List VC-426
Configuring ATM Interfaces for Voice Traffic Using AAL5
Verifying the ATM PVC Configuration VC-429
Configuring AAL2 Encapsulation for VoATM VC-429
Configuring T1/E1 Trunks VC-429
Cisco IOS Voice, Video, and Fax Configuration Guide
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VC-426
VC-410
Contents
Configuring Call Admission Control VC-431
Configuring Subcell Multiplexing VC-432
Configuring VoATM Dial Peers VC-433
Configuring VoATM Dial Peers to Support AAL2 VC-435
Configuring VoATM Dial Peers for Cisco Trunk Calls VC-437
Configuring Dial-Peer Hunting VC-438
Configuring Cisco Trunk Permanent Calls VC-439
Verifying the Voice Connection VC-440
Troubleshooting Tips VC-440
Verifying the ATM Interface Configuration VC-440
Verifying the VoATM Connection VC-442
Troubleshooting Tips VC-443
VoATM Configuration Examples VC-443
Back-to-Back VoATM PVCs Example VC-443
Voice and Data Traffic over ATM PVCs Example VC-444
VoATM for Cisco 3600 Series Routers Configuration Example VC-447
VoATM for the Cisco MC3810 Multiservice Concentrator Configuration Example
VC-451
TELEPHONY APPLICATIONS
Configuring TCL IVR Applications
TCL IVR Overview
VC-457
VC-457
TCL IVR Enhancements VC-458
MGCP Scripting VC-458
RTSP Client Implementation VC-459
TCL IVR Prompts Played on IP Call Legs
TCL Verbs VC-461
TCL IVR Prerequisite Tasks
VC-459
VC-463
TCL IVR Configuration Tasks List VC-464
Configuring the Call Application for the Dial Peer VC-464
Configuring TCL IVR on the Inbound POTS Dial Peer VC-466
Configuring TCL IVR on the Inbound VoIP Dial Peer VC-467
Configuring MGCP Scripting VC-468
Verifying TCL IVR Configuration VC-469
TCL IVR Configuration Examples VC-471
TCL IVR for Gateway1 (GW1) Configuration Example
TCL IVR for GW2 Configuration Example VC-474
MGCP Scripting Configuration Example VC-476
VC-471
Cisco IOS Voice, Video, and Fax Configuration Guide
xvii
Contents
Configuring Debit Card Applications
VC-479
Debit Card for Packet Telephony Overview VC-479
Debit Card Call Flow VC-482
RADIUS and H.323 Gateway-Specific Accounting
Audio File Prompts VC-487
Cisco-Provided Audio Files VC-487
Additional Miscellaneous Prompts VC-488
Audio Filenaming Convention VC-490
Creating Audio Index Files VC-490
Sample Index File VC-491
Debit Card Prerequisite Tasks
VC-487
VC-491
Debit Card for Packet Telephony Configuration Tasks List
Verifying the Debit Card Configuration VC-494
Debit Card Feature Configuration Example
Configuring Settlement Applications
VC-492
VC-494
VC-499
Settlement for Packet Telephony Overview VC-500
Settlement (OSP) Enhancements VC-501
Roaming VC-501
User Identification VC-502
Settlement Provider VC-502
Dial Peer VC-502
Dial Peer Settlement Option VC-502
Public Key Infrastructure Multiple Roots VC-503
User-Network Interface OSP VC-504
Click-to-Talk Functionality VC-505
Settlement for Packet Telephony Prerequisite Tasks
Restrictions
VC-506
VC-506
Settlement for Packet Telephony Configuration Task List VC-506
Configuring the Public Key Infrastructure VC-507
Configuring the Originating Gateway VC-508
Configuring the Settlement Provider VC-508
Configuring the Inbound POTS Dial Peer VC-509
Configuring the Outbound VoIP Dial Peer VC-510
Configuring the Terminating Gateway VC-511
Configuring the Settlement Provider VC-511
Configuring the Inbound VoIP Dial Peer VC-512
Configuring the Outbound POTS Dial Peer VC-513
Cisco IOS Voice, Video, and Fax Configuration Guide
xviii
Contents
Verifying Settlement Configuration VC-513
Configuring Settlement with Roaming VC-514
Configuring the Roaming Patterns on the Originating Gateway VC-514
Enabling the Roaming Feature for the Settlement Provider VC-514
Enabling the Roaming Feature in the Outbound Dial Peer VC-514
Configuring Settlement with PKI Multiple Roots VC-515
Configuring a Settlement Server with PKI Multiple Roots on the Originating Gateway VC-515
Configuring the Root Certificate for Token Validation on the Terminating Gateway VC-515
Defining the Token Validation on the Terminating Gateway VC-515
Configuring Settlement with Suggested Route VC-516
Settlement for Packet Telephony Configuration Examples VC-520
Settlement on the Originating Gateway Example VC-521
Settlement on the Terminating Gateway Example VC-522
Settlement with Roaming Example VC-523
Settlement with PKI Multiple Roots Example VC-526
Settlement with UNI-OSP Example VC-530
TRUNK MANAGEMENT AND CONDITIONING FEATURES
Configuring Trunk Connections and Conditioning Features
Trunking Overview VC-533
Simulated Lines and Trunks
VC-533
VC-534
Trunk Conditioning Signaling Attributes
VC-535
Congestion Monitoring and Management Features VC-535
T1/E1 Alarm Conditioning VC-536
PSTN Fallback VC-536
Calculated Impairment Planning Factor VC-538
Service Assurance Agent VC-538
Busyout VC-538
Local Voice Busyout VC-538
Advanced Voice Busyout VC-539
Busyout Monitor VC-539
Trunk Management Prerequisite Tasks VC-539
Configuring Trunk-Conditioning Signaling Attributes VC-540
Assigning Trunk-Conditioning Attributes to Network Dial Peers VC-543
Assigning Voice Classes to Voice Ports VC-544
Verifying the Signaling Attributes and Trunk Conditioning VC-544
Cisco IOS Voice, Video, and Fax Configuration Guide
xix
Contents
Configuring Trunk Connections VC-546
Configuring PLAR (Switched) Connections VC-546
Configuring Trunk/Tie-Line Connections VC-547
Configuring PLAR-OPX Connections VC-551
Configuring T1/E1 Alarm Generation Parameters VC-551
Verifying Alarm-Generation Parameters VC-553
Configuring PSTN Fallback VC-554
Configuring Fallback to Alternate Dial Peers VC-554
Configuring Destination Monitoring without Fallback to Alternate Dial Peers
Configuring Call Fallback Cache Parameters VC-555
Configuring Call Fallback Jitter-Probe Parameters VC-555
Configuring Call Fallback Probe-Timeout and Weight Parameters VC-555
Configuring Call Fallback Threshold Parameters VC-556
Configuring Call Fallback Map Parameters VC-556
Verifying PSTN Fallback Configuration VC-556
Troubleshooting Tips VC-556
Monitoring and Maintaining PSTN Fallback VC-557
Configuring Local Voice Busyout VC-557
Configuring the Busyout Trigger Event VC-558
Configuring Busyout of Voice Ports VC-558
Configuring a Voice Port to Monitor the Link to a Remote Interface VC-562
Configuring a Busyout Monitoring Voice Class VC-563
Trunk Connections and Conditioning Configuration Examples VC-565
Trunk Conditioning Configuration Example VC-565
Voice Class for VoFR and VoATM Dial Peers Configuration Example VC-566
Voice Class for Voice Ports Configuration Example VC-566
Voice Class for Default Signaling Patterns Configuration Example VC-566
Voice Class for Specified Signaling Patterns Configuration Example VC-567
PLAR (Switched Calls) Configuration Example VC-567
Permanent Trunks Configuration Example VC-568
Congestion Monitoring and Management Configuration Examples VC-570
Configuring PSTN Fallback for VoIP over Frame Relay Example VC-570
Configuring PSTN Fallback for VoIP over MLP Example VC-573
Local Voice Busyout Configuration Examples VC-578
Alarm Trigger for Busyout of Voice Ports Configuration Example VC-581
Cisco IOS Voice, Video, and Fax Configuration Guide
xx
VC-554
Contents
Configuring ISDN Interfaces for Voice
VC-583
ISDN Voice Interface Overview VC-584
QSIG Protocol Support VC-585
QSIG Protocol Stack VC-587
Switch-Type Configuration Options VC-588
Q.931 Support VC-588
ISDN Voice Interface Limitations VC-589
QSIG Support Limitations VC-589
ISDN Voice Interface Prerequisite Tasks
VC-590
ISDN Voice Interface Configuration Task List VC-590
Configuring ISDN BRI Interfaces VC-591
Verifying ISDN BRI Interface Configuration VC-594
Monitoring and Maintaining ISDN BRI Interfaces VC-597
Configuring ISDN PRI Interfaces VC-598
Configuring ISDN PRI Voice Ports VC-599
Verifying ISDN PRI Configuration VC-599
ISDN PRI Troubleshooting Tips VC-599
Configuring Global QSIG Support for BRI or PRI VC-600
Configuring Controllers for QSIG over PRI VC-601
Configuring BRI Interfaces for QSIG VC-601
Configuring PRI Interfaces for QSIG VC-603
Verifying the QSIG Configuration VC-605
QSIG Support Troubleshooting Tips VC-608
Configuring ISDN PRI Q.931 Support VC-609
ISDN Voice Interface Configuration Examples VC-610
ISDN to PBX and ISDN to PSTN Configuration Examples VC-610
ISDN Connection to a PBX Configuration Example VC-611
ISDN Connection to the PSTN Configuration Example VC-612
QSIG Support Configuration Examples VC-612
QSIG Support on Cisco 3600 Series Routers Example VC-613
QSIG Support on Cisco 7200 Series Routers Example VC-617
QSIG Support on Cisco MC3810 Multiservice Concentrators Example
Q.931 Support Configuration Examples VC-624
VC-622
Cisco IOS Voice, Video, and Fax Configuration Guide
xxi
Contents
Configuring PBX Interconnectivity Features
VC-627
Configuring QSIG PRI Signaling Support VC-627
Benefits of QSIG Voice Signaling VC-627
Configuring Voice over IP QSIG Network Transparency on the Cisco AS5300 VC-628
QSIG Prerequisite Tasks VC-629
QSIG Configuration Task List VC-629
Configuring VoIP QSIG VC-629
Configuring Fusion Call Control Signaling (NEC Fusion) on the Cisco AS5300 VC-632
Verifying VoIP QSIG Software on the Cisco AS5300 VC-633
Configuring QSIG PRI Signaling Support on the Cisco MC3810 VC-633
QSIG Prerequisite Tasks VC-634
Configuring T-CCS VC-637
T-CCS Overview VC-637
T-CCS Limitations VC-638
Related Documents for T-CCS VC-638
T-CCS Prerequisite Tasks VC-639
T-CCS Configuration Task List VC-639
Configuring T-CCS Cross-Connect VC-639
Configuring T-CCS Frame Forwarding VC-643
Configuring T-CCS for a Clear-Channel Codec VC-645
Verifying the T-CCS Configuration VC-650
Troubleshooting Tips for T-CCS VC-653
Monitoring and Maintaining T-CCS and Frame Forwarding VC-653
PBX Interconnectivity Configuration Examples VC-654
QSIG Configuration Examples VC-654
QSIG for VoIP Configuration Example VC-654
QSIG PRI Signaling on the Cisco MC3810 Configuration Example
T-CCS Configuration Examples VC-658
T-CCS over Frame Relay Configuration Example VC-658
T-CCS over IP Configuration Example VC-660
FAX, VIDEO, AND MODEM SUPPORT
Configuring Fax Applications
VC-665
Fax Applications Overview VC-665
On-Ramp Gateway VC-666
Off-Ramp Gateway VC-667
Cisco IOS Voice, Video, and Fax Configuration Guide
xxii
VC-656
Contents
Call Discrimination Process VC-668
POTS Dial Peers VC-668
MMoIP Dial Peers VC-669
On-Ramp Gateway Security VC-670
Attribute-Value Pairs for AAA VC-670
Access Control Lists VC-671
ESMTP Accounting Services VC-671
Message Delivery Notifications VC-672
Delivery Status Notifications VC-672
T.37 Store and Forward Fax VC-672
Modem Pooling VC-673
Fax Relay Packet Loss Concealment VC-673
Handling of Enclosures VC-674
T.37/T.38 Fax Gateway VC-675
Using Interactive Voice Response VC-676
T.38 Fax Relay for VoIP H.323 VC-676
Fax Applications Prerequisites VC-677
T.37 Store and Forward Fax Prerequisites VC-677
Configuring the SMTP Server VC-678
Configuring the MTAs VC-678
Configuring Fax Operation VC-679
Configuring All Mail Through One Mailer VC-679
Configuring Sendmail 8.8.5 for Single Recipients VC-679
Configuring the Redialers VC-682
Fax Relay Packet Loss Concealment Prerequisite Tasks VC-682
T.37/T.38 Fax Gateway Prerequisite Tasks VC-682
Downloading VCWare to the VFC VC-682
Copying Flash Files to the VFC VC-685
Unbundling VCWare VC-686
Adding Files to the Default File List VC-687
Adding Codecs to the Capability List VC-687
Deleting Files from VFC Flash Memory VC-688
Erasing the VFC Flash Memory VC-688
Configuring IVR VC-688
T.38 Fax Relay for VoIP H.323 Prerequisites VC-689
Fax Applications Configuration Tasks List
VC-689
Cisco IOS Voice, Video, and Fax Configuration Guide
xxiii
Contents
Configuring the On-Ramp Gateway VC-689
Configuring the Called Subscriber Number VC-690
Configuring the Sending MTA VC-690
Configuring POTS Dial Peers VC-691
Configuring MMoIP Dial Peers VC-691
Verifying the Gateway Configuration VC-692
Configuring the Off-Ramp Gateway VC-693
Configuring the Transmitting Subscriber Number VC-693
Configuring the Fax Transmission Speed VC-693
Configuring the Receiving Mail Transfer Agent VC-694
Configuring the POTS Dial Peer VC-694
Configuring the MMoIP Dial Peer VC-695
Configuring the Faxed Header Information VC-695
Configuring the Fax Cover Page Information VC-696
Verifying the Gateway Configuration VC-696
Configuring Gateway Security VC-697
Configuring On-Ramp Gateway Security VC-697
Configuring Off-Ramp Gateway Security VC-698
Configuring the Gateway Security for TCL Application Files
Verifying the Gateway Security Configuration VC-699
Configuring MDNs VC-699
Verifying MDN Configuration VC-700
Configuring DSNs VC-700
Verifying DSN Configuration VC-701
Configuring T.37 Store and Forward Fax VC-701
Configuring On-Ramp Modem Pooling VC-702
Configuring ECM VC-702
Configuring the T.37/T.38 Fax Gateway VC-702
Specifying the Interface Type for Fax Calls VC-703
Configuring IVR Functionality VC-703
Verify the IVR Configuration VC-704
Configuring T.38 Fax Relay for VoIP H.323 VC-705
Fax Applications Configuration Examples VC-707
T.37 Store and Forward Fax Configuration Examples VC-707
T.37/T.38 Fax Gateway Examples VC-714
T.38 Fax Relay for VoIP H.323 Configuration Example VC-717
Cisco IOS Voice, Video, and Fax Configuration Guide
xxiv
VC-699
Contents
Configuring Video Applications
VC-719
Video Applications Overview VC-719
Cisco Video Support by Platform VC-720
Cisco MC3810 Multiservice Concentrator VC-720
Cisco 2600 Series, 3600 Series, and 7200 Series Router and MC3810 Multiservice
Concentrator VC-720
Cisco 3600 Series Router VC-721
Multimedia Conference Manager with Voice Gateway Image and RSVP to ATM SVC
Mapping VC-721
ATM Nonreal-Time VBR SVC Support for Video VC-722
Video Applications Prerequisite Tasks and Restrictions
VC-722
Video Applications Configuration Task List VC-723
Configuring Video in Pass-Through Mode VC-723
Configuring Video over ATM AAL1 VC-725
Tuning Circuit Emulation Services Settings VC-728
Configuring Video over ATM PVCs and SVCs VC-728
Configuring Network Clocks and Controllers VC-731
Verifying Network Clock and Controller Configuration VC-734
Configuring Serial Interfaces to Support the Video Codec VC-735
Configuring ATM Interfaces to Support Video over PVCs and SVCs
Configuring Video Dial Peers VC-744
Verifying Video Dial-Peer Configuration VC-747
Troubleshooting Video over ATM SVCs and PVCs VC-747
Configuring the CES Clock VC-752
Configuring Structured CES VC-754
Configuring the Proxy and T.120 VC-757
Configuring the Gatekeeper to Support Zone Bandwidth VC-761
Configuring RSVP-ATM QoS Interworking VC-762
Verifying RSVP-ATM QoS Interworking Configuration VC-762
VC-736
Video Applications Configuration Examples VC-764
Video over ATM PVCs and SVCs Configuration Examples VC-764
CES Video Traffic on the Cisco MC3810 Multiservice Concentrator Configuration Example
Video Traffic on a Cisco 3600 Series Router Configuration Example VC-767
Cisco IP/VC 3510 Multipoint Control Unit with Cisco IOS Gatekeeper/Proxy Configuration
Example VC-769
CES Clock Configuration Examples VC-771
VC-766
Cisco IOS Voice, Video, and Fax Configuration Guide
xxv
Contents
Configuring Modem Transport Support for VoIP
VC-773
Modem Transport Support Overview VC-773
Monitoring and Maintaining Modem Call Status VC-773
DS-0 Busyout Traps VC-774
ISDN PRI-Requested Channel-Not-Available Traps VC-774
Modem Health Traps VC-774
show controllers timeslots Command VC-774
DS-1 Loopback Traps VC-774
Modem Passthrough over VoIP VC-775
Modem Tone Detection VC-775
Passthrough Switchover VC-776
Controlled Redundancy VC-776
Packet Size VC-776
Clock Slip Buffer Management VC-776
Modem Transport Support Prerequisite Tasks
VC-776
Modem Transport Support Configuration Task List VC-777
Configuring Modem Call Status VC-777
Enabling DS-0 Busyout Traps VC-777
Enabling ISDN PRI-Requested Channel-Not-Available Traps VC-777
Enabling Modem Health Traps VC-778
Enabling DS-1 Loopback Traps VC-778
Verifying Enabled Traps VC-778
Troubleshooting Tips VC-779
Configuring Modem Passthrough VC-779
Configuring Modem Passthrough Globally VC-780
Configuring Modem Passthrough for a Specific Dial Peer VC-781
Verifying Modem Passthrough VC-783
Troubleshooting Tips for Modem Passthrough VC-783
Monitoring and Maintaining Modem Passthrough VC-783
Modem Transport Support Configuration Examples VC-784
Modem Call Status Configuration Example VC-784
Modem Passthrough Configuration Example VC-786
Cisco IOS Voice, Video, and Fax Configuration Guide
xxvi
Contents
APPENDIXES
Configuring Synchronized Clocking
VC-791
Synchronized Clocking Overview VC-791
Configuring the Cisco MC3810 to a Synchronous Clocked Network
VC-792
Synchronized Clocking Configuration Task List VC-793
Configuring the Cisco MC3810 to Obtain Clocking from the Network VC-793
Configuring the Cisco MC3810 to Recover Clocking from a Network Device Attached to a T1/E1
Controller VC-794
Configuring a T1/E1 Controller to Loop-Time the Clocking Back to the Network Clock
Source VC-798
Configuring the Cisco MC3810 to Recover Clocking from a Network Device Attached to Serial
0 VC-801
Configuring the Cisco MC3810 to Use the Internal Clock Source VC-804
Configuring a Hierarchy of Clock Sources for Backup Purposes VC-805
Caller ID on Cisco 2600 and 3600 Series Routers and Cisco MC3810 Multiservice
Concentrators VC-811
Caller ID Overview VC-811
Calling Name and Number
Call Time Display VC-813
Caller ID Prerequisites Tasks
VC-812
VC-814
Caller ID Configuration Task List VC-815
Configuring Voice Ports to Support Caller ID VC-815
Configuring FXS and FXO Voice Ports to Support Caller ID
Verifying Caller ID on Voice Ports Configuration VC-822
Troubleshooting Tips VC-823
Cisco Hoot and Holler over IP
VC-818
VC-825
Hoot and Holler over IP Overview VC-825
Current Hoot and Holler Implementations VC-827
Cisco Hoot and Holler over IP Overview VC-827
Voice Multicasting VC-828
IP/TV Access VC-829
Interactive Voice Response VC-830
Migration Strategy VC-830
Technical Details of the Cisco Hoot and Holler over IP Solution
IP Multicast and DSP Arbitration and Mixing VC-832
VC-831
Cisco IOS Voice, Video, and Fax Configuration Guide
xxvii
Contents
Bandwidth Planning VC-832
Virtual Interface VC-834
Connection Trunk VC-834
Cisco Hoot and Holler over IP Restrictions
VC-835
Configuration Tasks VC-835
Configuring Multicast Routing VC-836
Configuring the Virtual Interface VC-836
Configuring VoIP Dial Peers VC-837
Configuring E&M Voice Ports VC-839
Configuring for Receive Only Mode VC-841
Configuring Relevant Interface (Serial/Ethernet) VC-842
Configuring Voice Ports in High-Density Voice Network Modules
VC-842
Configuration Examples VC-843
Voice Multicasting over an Ethernet LAN VC-844
Configuring the Second Router VC-845
Verifying the Configuration VC-845
High-Density Voice Modules VC-846
Dial-Peer Configuration VC-846
Ethernet Configuration VC-847
Voice Multicasting over a WAN VC-847
Quality of Service VC-848
Cisco Hoot and Holler over IP with Ethernet Topology (Two Hoot Groups) VC-849
Router-1 (E&M Four-Wire Ports) VC-849
Router-2 (FXS Ports) VC-850
Router-3 (FXO Ports) VC-851
Cisco Hoot and Holler over IP with Frame-Relay Topology (One Hoot Group) VC-852
Router-1 VC-852
Router-2 VC-853
Router-3 VC-854
Enhanced Voice Services for Japan for Cisco 800 Series Routers
Enhanced Voice Services Overview VC-857
Enhanced Voice Services Limitations VC-860
Related Documents for Enhanced Voice Services
Enhanced Voice Services Prerequisite Tasks
VC-861
Enhanced Voice Services Configuration Task List
Configuring Caller ID VC-862
Cisco IOS Voice, Video, and Fax Configuration Guide
xxviii
VC-861
VC-862
VC-857
Contents
Configuring Call Blocking on Caller ID VC-862
Configuring Nariwake VC-863
Configuring I Number VC-863
Monitoring and Maintaining Enhanced Voice Services
Enhanced Voice Services Configuration Examples
Caller ID Example VC-864
Call Blocking on Caller ID Example VC-864
Local Call Waiting Example VC-864
Nariwake Example VC-865
I Number Example VC-865
POTS Dial Example VC-865
POTS Disconnect Example VC-865
Managing Cisco AS5300 Voice Feature Cards
VFC Management Overview
VC-864
VC-864
VC-867
VC-867
VFC Management Task List VC-868
Downloading VCWare VC-868
Identifying the VFC Mode VC-869
Downloading Software (VCWare Mode) VC-869
Downloading Software (ROM Monitor Mode) VC-870
Copying Flash Files to the VFC VC-870
Downloading VCWare to the VFC from the Router Motherboard
Downloading VCWare to the VFC from a TFTP Server VC-871
Unbundling VCWare VC-871
Adding Files to the Default File List VC-872
Adding Codecs to the Capability List VC-872
Deleting Files from VFC Flash Memory VC-873
Erasing the VFC Flash Memory VC-873
VC-871
Global System for Mobile Communications Full Rate and Enhanced Full Rate Codecs
VC-875
Global System for Mobile Communications Full Rate and Enhanced Full Rate Codecs Overview
Prerequisite Tasks and Restrictions
GSM Configuration Tasks VC-876
Configuring Dial Peers VC-876
Verifying Gateway Configuration
GSM Configuration Example
VC-875
VC-876
VC-879
VC-880
Cisco IOS Voice, Video, and Fax Configuration Guide
xxix
Contents
Configuring the Cisco SS7/C7 Dial Access Solution System
Cisco SS7/C7 Dial Access Overview 884
Redundant Link Management 884
Continuity Test Subsystem 885
ISDN Module 886
RLM Configuration Task List 886
Configuring the Access Server for RLM
Verifying RLM 887
Troubleshooting RLM 888
886
Cisco SS7/C7 Dial Access Solution System Examples
INDEX
Cisco IOS Voice, Video, and Fax Configuration Guide
xxx
888
883
About Cisco IOS Software Documentation
This chapter discusses the objectives, audience, organization, and conventions of Cisco IOS software
documentation. It also provides sources for obtaining documentation from Cisco Systems.
Documentation Objectives
Cisco IOS software documentation describes the tasks and commands necessary to configure and
maintain Cisco networking devices.
Audience
The Cisco IOS software documentation set is intended primarily for users who configure and maintain
Cisco networking devices (such as routers and switches) but who may not be familiar with the tasks, the
relationship between tasks, or the Cisco IOS software commands necessary to perform particular tasks.
The Cisco IOS software documentation set is also intended for those users experienced with Cisco IOS
software who need to know about new features, new configuration options, and new software
characteristics in the current Cisco IOS software release.
Documentation Organization
The Cisco IOS software documentation set consists of documentation modules and master indexes. In
addition to the main documentation set, there are supporting documents and resources.
Documentation Modules
The Cisco IOS documentation modules consist of configuration guides and corresponding command
reference publications. Chapters in a configuration guide describe protocols, configuration tasks, and
Cisco IOS software functionality and contain comprehensive configuration examples. Chapters in a
command reference publication provide complete Cisco IOS command syntax information. Use each
configuration guide in conjunction with its corresponding command reference publication.
Cisco IOS Voice, Video, and Fax Configuration Guide
xxxi
About Cisco IOS Software Documentation
Documentation Organization
Figure 1 shows the Cisco IOS software documentation modules.
Note
Figure 1
The abbreviations (for example, FC and FR) next to the book icons are page designators, which are
defined in a key in the index of each document to help you with navigation. The bullets under each
module list the major technology areas discussed in the corresponding books.
Cisco IOS Software Documentation Modules
IPC
FC
Cisco IOS
Configuration
Fundamentals
Configuration
Guide
Cisco IOS
Configuration
Fundamentals
Command
Reference
FR
IP2R
Module FC/FR:
• Cisco IOS User
Interfaces
• File Management
• System Management
WC
WR
Cisco IOS
Wide-Area
Networking
Command
Reference
Cisco IOS
IP Command
Reference,
Volume 1 of 3:
Addressing
and Services
Cisco IOS
IP Command
Reference,
Volume 2 of 3:
Routing
Protocols
P2C
IP3R
Cisco IOS
IP Command
Reference,
Volume 3 of 3:
Multicast
Cisco IOS
Interface
Configuration
Guide
IR
Cisco IOS
Interface
Command
Reference
P3C
Cisco IOS
AppleTalk and
Novell IPX
Configuration
Guide
P2R
Cisco IOS
AppleTalk and
Novell IPX
Command
Reference
P3R
Module P2C/P2R:
• AppleTalk
• Novell IPX
MWC
Cisco IOS
Mobile
Wireless
Configuration
Guide
MWR
Module IC/IR:
• LAN Interfaces
• Serial Interfaces
• Logical Interfaces
Cisco IOS
Mobile
Wireless
Command
Reference
Module MWC/MWR:
• General Packet
Radio Service
Cisco IOS
Apollo Domain,
Banyan VINES,
DECnet, ISO
CLNS, and XNS
Configuration
Guide
SC
Cisco IOS
Apollo Domain,
Banyan VINES,
DECnet, ISO
CLNS, and XNS
Command
Reference
Module P3C/P3R:
• Apollo Domain
• Banyan VINES
• DECnet
• ISO CLNS
• XNS
Cisco IOS
Security
Configuration
Guide
SR
Cisco IOS
Security
Command
Reference
Module SC/SR:
• AAA Security Services
• Security Server Protocols
• Traffic Filtering and Firewalls
• IP Security and Encryption
• Passwords and Privileges
• Neighbor Router Authentication
• IP Security Options
• Supported AV Pairs
47953
Module WC/WR:
• ATM
• Broadband Access
• Frame Relay
• SMDS
• X.25 and LAPB
IP1R
Module IPC/IP1R/IP2R/IP3R:
• IP Addressing and Services
• IP Routing Protocols
• IP Multicast
IC
Cisco IOS
Wide-Area
Networking
Configuration
Guide
Cisco IOS
IP
Configuration
Guide
Cisco IOS Voice, Video, and Fax Configuration Guide
xxxii
About Cisco IOS Software Documentation
Documentation Organization
Cisco IOS
Dial
Technologies
Configuration
Guide
TC
BC
Cisco IOS
Terminal
Services
Configuration
Guide
Cisco IOS
Bridging and
IBM Networking
Configuration
Guide
B2R
B1R
DR
Cisco IOS
Dial
Technologies
Command
Reference
TR
Module DC/DR:
• Preparing for Dial Access
• Modem and Dial Shelf Configuration
and Management
• ISDN Configuration
• Signalling Configuration
• Dial-on-Demand Routing
Configuration
• Dial-Backup Configuration
• Dial-Related Addressing Services
• Virtual Templates, Profiles, and
Networks
• PPP Configuration
• Callback and Bandwidth Allocation
Configuration
• Dial Access Specialized Features
• Dial Access Scenarios
VC
Cisco IOS
Voice, Video,
and Fax
Configuration
Guide
VR
Cisco IOS
Voice, Video,
and Fax
Command
Reference
Module VC/VR:
• Voice over IP
• Call Control Signalling
• Voice over
Frame Relay
• Voice over ATM
• Telephony Applications
• Trunk Management
• Fax, Video, and
Modem Support
Cisco IOS
Terminal
Services
Command
Reference
Module TC/TR:
• ARA
• LAT
• NASI
• Telnet
• TN3270
• XRemote
• X.28 PAD
• Protocol Translation
QC
Cisco IOS
Quality of
Service
Solutions
Configuration
Guide
QR
Cisco IOS
Quality of
Service
Solutions
Command
Reference
Module QC/QR:
• Packet Classification
• Congestion Management
• Congestion Avoidance
• Policing and Shaping
• Signalling
• Link Efficiency
Mechanisms
Cisco IOS
Bridging
and IBM
Networking
Command
Reference,
Volume 1 of 2
Cisco IOS
Bridging
and IBM
Networking
Command
Reference,
Volume 2 of 2
Module BC/B1R:
• Transparent
Bridging
• SRB
• Token Ring
Inter-Switch Link
• Token Ring Route
Switch Module
• RSRB
• DLSw+
• Serial Tunnel and
Block Serial Tunnel
• LLC2 and SDLC
• IBM Network
Media Translation
• SNA Frame Relay
Access
• NCIA Client/Server
• Airline Product Set
XC
Module BC/B2R:
• DSPU and SNA
Service Point
• SNA Switching
Services
• Cisco Transaction
Connection
• Cisco Mainframe
Channel Connection
• CLAW and TCP/IP
Offload
• CSNA, CMPC,
and CMPC+
• TN3270 Server
Cisco IOS
Switching
Services
Configuration
Guide
XR
Cisco IOS
Switching
Services
Command
Reference
Module XC/XR:
• Cisco IOS
Switching Paths
• NetFlow Switching
• Multiprotocol Label Switching
• Multilayer Switching
• Multicast Distributed Switching
• Virtual LANs
• LAN Emulation
47954
DC
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About Cisco IOS Software Documentation
Documentation Organization
Master Indexes
Two master indexes provide indexing information for the Cisco IOS software documentation set:
an index for the configuration guides and an index for the command references. Individual books also
contain a book-specific index.
The master indexes provide a quick way for you to find a command when you know the command name
but not which module contains the command. When you use the online master indexes, you can click the
page number for an index entry and go to that page in the online document.
Supporting Documents and Resources
The following documents and resources support the Cisco IOS software documentation set:
•
Cisco IOS Command Summary (two volumes)—This publication explains the function and syntax
of the Cisco IOS software commands. For more information about defaults and usage guidelines,
refer to the Cisco IOS command reference publications.
•
Cisco IOS System Error Messages—This publication lists and describes Cisco IOS system error
messages. Not all system error messages indicate problems with your system. Some are purely
informational, and others may help diagnose problems with communications lines, internal
hardware, or the system software.
•
Cisco IOS Debug Command Reference—This publication contains an alphabetical listing of the
debug commands and their descriptions. Documentation for each command includes a brief
description of its use, command syntax, usage guidelines, and sample output.
•
Dictionary of Internetworking Terms and Acronyms—This Cisco publication compiles and defines
the terms and acronyms used in the internetworking industry.
•
New feature documentation—The Cisco IOS software documentation set documents the mainline
release of Cisco IOS software (for example, Cisco IOS Release 12.2). New software features are
introduced in early deployment releases (for example, the Cisco IOS “T” release train for 12.2,
12.2(x)T). Documentation for these new features can be found in standalone documents called
“feature modules.” Feature module documentation describes new Cisco IOS software and hardware
networking functionality and is available on Cisco.com and the Documentation CD-ROM.
•
Release notes—This documentation describes system requirements, provides information about
new and changed features, and includes other useful information about specific software releases.
See the section “Using Software Release Notes” in the chapter “Using Cisco IOS Software” for
more information.
•
Caveats documentation—This documentation provides information about Cisco IOS software
defects in specific software releases.
•
RFCs—RFCs are standards documents maintained by the Internet Engineering Task Force (IETF).
Cisco IOS software documentation references supported RFCs when applicable. The full text of
referenced RFCs may be obtained on the World Wide Web at http://www.rfc-editor.org/.
•
MIBs—MIBs are used for network monitoring. For lists of supported MIBs by platform and release,
and to download MIB files, see the Cisco MIB website on Cisco.com at
http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml.
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New and Changed Information
New and Changed Information
The Cisco IOS Voice, Video, and Fax Configuration Guide is the result of reorganizing and renaming the
Cisco IOS Multiservice Applications Configuration Guide.
The Cisco IOS Voice, Video, and Fax Configuration Guide contains information about configuring Voice
over IP, Voice over Frame Relay, Voice over ATM, and telephony applications using Interactive Voice
Response (IVR), fax, and video. This release adds the following new technologies:
•
Media Gateway Control Protocol/Simple Gateway Control Protocol
•
Session Initiation Protocol
•
T.38-compliant fax relay
•
Hoot and holler over IP
•
Caller ID
This release of the Cisco IOS Voice, Video, and Fax Configuration Guide deletes the following
technologies:
•
Broadband—covered in a separate configuration guide.
•
Voice over HDLC—no longer supported by Cisco routers.
Document Conventions
Within Cisco IOS software documentation, the term router is generally used to refer to a variety of Cisco
products (for example, routers, access servers, and switches). Routers, access servers, and other
networking devices that support Cisco IOS software are shown interchangeably within examples. These
products are used only for illustrative purposes; that is, an example that shows one product does not
necessarily indicate that other products are not supported.
The Cisco IOS documentation set uses the following conventions:
Convention
Description
^ or Ctrl
The ^ and Ctrl symbols represent the Control key. For example, the key combination ^D or Ctrl-D
means hold down the Control key while you press the D key. Keys are indicated in capital letters but
are not case sensitive.
string
A string is a nonquoted set of characters shown in italics. For example, when setting an SNMP
community string to public, do not use quotation marks around the string or the string will include the
quotation marks.
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Document Conventions
Command syntax descriptions use the following conventions:
Convention
Description
boldface
Boldface text indicates commands and keywords that you enter literally as shown.
italics
Italic text indicates arguments for which you supply values.
[x]
Square brackets enclose an optional element (keyword or argument).
|
A vertical line indicates a choice within an optional or required set of keywords or arguments.
[x | y]
Square brackets enclosing keywords or arguments separated by a vertical line indicate an optional
choice.
{x | y}
Braces enclosing keywords or arguments separated by a vertical line indicate a required choice.
Nested sets of square brackets or braces indicate optional or required choices within optional or required
elements. For example:
Convention
Description
[x {y | z}]
Braces and a vertical line within square brackets indicate a required choice within an optional element.
Examples use the following conventions:
Convention
Description
screen
Examples of information displayed on the screen are set in Courier font.
boldface screen
Examples of text that you must enter are set in Courier bold font.
<
Angle brackets enclose text that is not printed to the screen, such as passwords.
>
!
[
An exclamation point at the beginning of a line indicates a comment line. (Exclamation points are also
displayed by the Cisco IOS software for certain processes.)
]
Square brackets enclose default responses to system prompts.
The following conventions are used to attract the attention of the reader:
Caution
Means reader be careful. In this situation, you might do something that could result in equipment
damage or loss of data.
Note
Means reader take note. Notes contain helpful suggestions or references to materials not contained
in this manual.
Timesaver
Means the described action saves time. You can save time by performing the action described in the
paragraph.
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Obtaining Documentation
Obtaining Documentation
The following sections provide sources for obtaining documentation from Cisco Systems.
World Wide Web
The most current Cisco documentation is available on the World Wide Web at the following website:
http://www.cisco.com
Translated documentation is available at the following website:
http://www.cisco.com/public/countries_languages.html
Ordering Documentation
Cisco documentation can be ordered in the following ways:
•
Registered Cisco Direct Customers can order Cisco product documentation from the Networking
Products MarketPlace:
http://www.cisco.com/cgi-bin/order/order_root.pl
•
Registered Cisco.com users can order the Documentation CD-ROM through the online Subscription
Store:
http://www.cisco.com/go/subscription
•
Nonregistered Cisco.com users can order documentation through a local account representative by
calling Cisco corporate headquarters (California, USA) at 408 526-7208 or, in North America, by
calling 800 553-NETS(6387).
Documentation Feedback
If you are reading Cisco product documentation on the World Wide Web, you can submit technical
comments electronically. Click Feedback in the toolbar and select Documentation. After you complete
the form, click Submit to send it to Cisco.
You can e-mail your comments to [email protected]
To submit your comments by mail, use the response card behind the front cover of your document, or
write to the following address:
Cisco Systems, Inc.
Document Resource Connection
170 West Tasman Drive
San Jose, CA 95134-9883
We appreciate your comments.
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About Cisco IOS Software Documentation
Obtaining Technical Assistance
Obtaining Technical Assistance
Cisco provides Cisco.com as a starting point for all technical assistance. Customers and partners can
obtain documentation, troubleshooting tips, and sample configurations from online tools. For Cisco.com
registered users, additional troubleshooting tools are available from the TAC website.
Cisco.com
Cisco.com is the foundation of a suite of interactive, networked services that provides immediate, open
access to Cisco information and resources at anytime, from anywhere in the world. This highly
integrated Internet application is a powerful, easy-to-use tool for doing business with Cisco.
Cisco.com provides a broad range of features and services to help customers and partners streamline
business processes and improve productivity. Through Cisco.com, you can find information about Cisco
and our networking solutions, services, and programs. In addition, you can resolve technical issues with
online technical support, download and test software packages, and order Cisco learning materials and
merchandise. Valuable online skill assessment, training, and certification programs are also available.
Customers and partners can self-register on Cisco.com to obtain additional personalized information and
services. Registered users can order products, check on the status of an order, access technical support,
and view benefits specific to their relationships with Cisco.
To access Cisco.com, go to the following website:
http://www.cisco.com
Technical Assistance Center
The Cisco TAC website is available to all customers who need technical assistance with a Cisco product
or technology that is under warranty or covered by a maintenance contract.
Contacting TAC by Using the Cisco TAC Website
If you have a priority level 3 (P3) or priority level 4 (P4) problem, contact TAC by going to the TAC
website:
http://www.cisco.com/tac
P3 and P4 level problems are defined as follows:
•
P3—Your network performance is degraded. Network functionality is noticeably impaired, but most
business operations continue.
•
P4—You need information or assistance on Cisco product capabilities, product installation, or basic
product configuration.
In each of the above cases, use the Cisco TAC website to quickly find answers to your questions.
To register for Cisco.com, go to the following website:
http://www.cisco.com/register/
If you cannot resolve your technical issue by using the TAC online resources, Cisco.com registered users
can open a case online by using the TAC Case Open tool at the following website:
http://www.cisco.com/tac/caseopen
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Obtaining Technical Assistance
Contacting TAC by Telephone
If you have a priority level 1 (P1) or priority level 2 (P2) problem, contact TAC by telephone and
immediately open a case. To obtain a directory of toll-free numbers for your country, go to the following
website:
http://www.cisco.com/warp/public/687/Directory/DirTAC.shtml
P1 and P2 level problems are defined as follows:
•
P1—Your production network is down, causing a critical impact to business operations if service is
not restored quickly. No workaround is available.
•
P2—Your production network is severely degraded, affecting significant aspects of your business
operations. No workaround is available.
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Obtaining Technical Assistance
Cisco IOS Voice, Video, and Fax Configuration Guide
xl
Using Cisco IOS Software
This chapter provides helpful tips for understanding and configuring Cisco IOS software using the
command-line interface (CLI). It contains the following sections:
•
Understanding Command Modes
•
Getting Help
•
Using the no and default Forms of Commands
•
Saving Configuration Changes
•
Filtering Output from the show and more Commands
•
Identifying Supported Platforms
For an overview of Cisco IOS software configuration, refer to the Cisco IOS Configuration
Fundamentals Configuration Guide.
For information on the conventions used in the Cisco IOS software documentation set, see the chapter
“About Cisco IOS Software Documentation” located at the beginning of this book.
Understanding Command Modes
You use the CLI to access Cisco IOS software. Because the CLI is divided into many different modes,
the commands available to you at any given time depend on the mode you are currently in. Entering a
question mark (?) at the CLI prompt allows you to obtain a list of commands available for each command
mode.
When you log in to the CLI, you are in user EXEC mode. User EXEC mode contains only a limited
subset of commands. To have access to all commands, you must enter privileged EXEC mode, normally
by using a password. From privileged EXEC mode you can issue any EXEC command—user or
privileged mode—or you can enter global configuration mode. Most EXEC commands are one-time
commands. For example, show commands show important status information, and clear commands
clear counters or interfaces. The EXEC commands are not saved when the software reboots.
Configuration modes allow you to make changes to the running configuration. If you later save the
running configuration to the startup configuration, these changed commands are stored when the
software is rebooted. To enter specific configuration modes, you must start at global configuration mode.
From global configuration mode, you can enter interface configuration mode and a variety of other
modes, such as protocol-specific modes.
ROM monitor mode is a separate mode used when the Cisco IOS software cannot load properly. If a valid
software image is not found when the software boots or if the configuration file is corrupted at startup,
the software might enter ROM monitor mode.
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Using Cisco IOS Software
Getting Help
Table 1 describes how to access and exit various common command modes of the Cisco IOS software.
It also shows examples of the prompts displayed for each mode.
Table 1
Accessing and Exiting Command Modes
Command
Mode
Access Method
Prompt
Exit Method
User EXEC
Log in.
Router>
Use the logout command.
Privileged
EXEC
From user EXEC mode,
use the enable EXEC
command.
Router#
To return to user EXEC mode, use the disable
command.
Global
configuration
From privileged EXEC
mode, use the configure
terminal privileged
EXEC command.
Router(config)#
To return to privileged EXEC mode from global
configuration mode, use the exit or end command,
or press Ctrl-Z.
Interface
configuration
Router(config-if)#
From global
configuration mode,
specify an interface using
an interface command.
To return to global configuration mode, use the exit
command.
>
From privileged EXEC
mode, use the reload
EXEC command. Press
the Break key during the
first 60 seconds while the
system is booting.
To exit ROM monitor mode, use the continue
command.
ROM monitor
To return to privileged EXEC mode, use the end
command, or press Ctrl-Z.
For more information on command modes, refer to the “Using the Command-Line Interface” chapter in
the Cisco IOS Configuration Fundamentals Configuration Guide.
Getting Help
Entering a question mark (?) at the CLI prompt displays a list of commands available for each command
mode. You can also get a list of keywords and arguments associated with any command by using the
context-sensitive help feature.
To get help specific to a command mode, a command, a keyword, or an argument, use one of the
following commands:
Command
Purpose
help
Provides a brief description of the help system in any command mode.
abbreviated-command-entry?
Provides a list of commands that begin with a particular character string. (No space
between command and question mark.)
abbreviated-command-entry<Tab>
Completes a partial command name.
?
Lists all commands available for a particular command mode.
command ?
Lists the keywords or arguments that you must enter next on the command line.
(Space between command and question mark.)
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Getting Help
Example: How to Find Command Options
This section provides an example of how to display syntax for a command. The syntax can consist of
optional or required keywords and arguments. To display keywords and arguments for a command, enter
a question mark (?) at the configuration prompt or after entering part of a command followed by a space.
The Cisco IOS software displays a list and brief description of available keywords and arguments. For
example, if you were in global configuration mode and wanted to see all the keywords or arguments for
the arap command, you would type arap ?.
The <cr> symbol in command help output stands for “carriage return.” On older keyboards, the carriage
return key is the Return key. On most modern keyboards, the carriage return key is the Enter key. The
<cr> symbol at the end of command help output indicates that you have the option to press Enter to
complete the command and that the arguments and keywords in the list preceding the <cr> symbol are
optional. The <cr> symbol by itself indicates that no more arguments or keywords are available and that
you must press Enter to complete the command.
Table 2 shows examples of how you can use the question mark (?) to assist you in entering commands.
The table steps you through configuring an IP address on a serial interface on a Cisco 7206 router that
is running Cisco IOS Release 12.0(3).
Table 2
How to Find Command Options
Command
Comment
Router> enable
Password: <password>
Router#
Enter the enable command and
password to access privileged EXEC
commands. You are in privileged
EXEC mode when the prompt changes
to Router#.
Router# configure terminal
Enter configuration commands, one per line. End with CNTL/Z.
Router(config)#
Enter the configure terminal
privileged EXEC command to enter
global configuration mode. You are in
global configuration mode when the
prompt changes to Router(config)#.
Router(config)# interface serial ?
<0-6>
Serial interface number
Router(config)# interface serial 4 ?
/
Router(config)# interface serial 4/ ?
<0-3>
Serial interface number
Router(config)# interface serial 4/0
Router(config-if)#
Enter interface configuration mode by
specifying the serial interface that you
want to configure using the interface
serial global configuration command.
Enter ? to display what you must enter
next on the command line. In this
example, you must enter the serial
interface slot number and port number,
separated by a forward slash.
You are in interface configuration mode
when the prompt changes to
Router(config-if)#.
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Getting Help
Table 2
How to Find Command Options (continued)
Command
Comment
Router(config-if)# ?
Interface configuration commands:
.
.
.
ip
Interface Internet Protocol config commands
keepalive
Enable keepalive
lan-name
LAN Name command
llc2
LLC2 Interface Subcommands
load-interval
Specify interval for load calculation for an
interface
locaddr-priority
Assign a priority group
logging
Configure logging for interface
loopback
Configure internal loopback on an interface
mac-address
Manually set interface MAC address
mls
mls router sub/interface commands
mpoa
MPOA interface configuration commands
mtu
Set the interface Maximum Transmission Unit (MTU)
netbios
Use a defined NETBIOS access list or enable
name-caching
no
Negate a command or set its defaults
nrzi-encoding
Enable use of NRZI encoding
ntp
Configure NTP
.
.
.
Router(config-if)#
Enter ? to display a list of all the
interface configuration commands
available for the serial interface. This
example shows only some of the
available interface configuration
commands.
Router(config-if)# ip ?
Interface IP configuration subcommands:
access-group
Specify access control for packets
accounting
Enable IP accounting on this interface
address
Set the IP address of an interface
authentication
authentication subcommands
bandwidth-percent
Set EIGRP bandwidth limit
broadcast-address
Set the broadcast address of an interface
cgmp
Enable/disable CGMP
directed-broadcast Enable forwarding of directed broadcasts
dvmrp
DVMRP interface commands
hello-interval
Configures IP-EIGRP hello interval
helper-address
Specify a destination address for UDP broadcasts
hold-time
Configures IP-EIGRP hold time
.
.
.
Router(config-if)# ip
Enter the command that you want to
configure for the interface. This
example uses the ip command.
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Enter ? to display what you must enter
next on the command line. This
example shows only some of the
available interface IP configuration
commands.
Using Cisco IOS Software
Using the no and default Forms of Commands
Table 2
How to Find Command Options (continued)
Command
Comment
Router(config-if)# ip address ?
A.B.C.D
IP address
negotiated
IP Address negotiated over PPP
Router(config-if)# ip address
Enter the command that you want to
configure for the interface. This
example uses the ip address command.
Enter ? to display what you must enter
next on the command line. In this
example, you must enter an IP address
or the negotiated keyword.
A carriage return (<cr>) is not
displayed; therefore, you must enter
additional keywords or arguments to
complete the command.
Enter the keyword or argument you
want to use. This example uses the
172.16.0.1 IP address.
Router(config-if)# ip address 172.16.0.1 ?
A.B.C.D
IP subnet mask
Router(config-if)# ip address 172.16.0.1
Enter ? to display what you must enter
next on the command line. In this
example, you must enter an IP subnet
mask.
A <cr> is not displayed; therefore, you
must enter additional keywords or
arguments to complete the command.
Router(config-if)# ip address 172.16.0.1 255.255.255.0 ?
secondary
Make this IP address a secondary address
<cr>
Router(config-if)# ip address 172.16.0.1 255.255.255.0
Enter the IP subnet mask. This example
uses the 255.255.255.0 IP subnet mask.
Enter ? to display what you must enter
next on the command line. In this
example, you can enter the secondary
keyword, or you can press Enter.
A <cr> is displayed; you can press
Enter to complete the command, or
you can enter another keyword.
Router(config-if)# ip address 172.16.0.1 255.255.255.0
Router(config-if)#
In this example, Enter is pressed to
complete the command.
Using the no and default Forms of Commands
Almost every configuration command has a no form. In general, use the no form to disable a function.
Use the command without the no keyword to reenable a disabled function or to enable a function that is
disabled by default. For example, IP routing is enabled by default. To disable IP routing, use the no ip
routing command; to reenable IP routing, use the ip routing command. The Cisco IOS software
command reference publications provide the complete syntax for the configuration commands and
describe what the no form of a command does.
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Using Cisco IOS Software
Saving Configuration Changes
Configuration commands also can have a default form, which returns the command settings to the
default values. Most commands are disabled by default, so in such cases using the default form has the
same result as using the no form of the command. However, some commands are enabled by default and
have variables set to certain default values. In these cases, the default form of the command enables the
command and sets the variables to their default values. The Cisco IOS software command reference
publications describe the effect of the default form of a command if the command functions differently
than the no form.
Saving Configuration Changes
Use the copy system:running-config nvram:startup-config command to save your configuration
changes to the startup configuration so that the changes will not be lost if the software reloads or a power
outage occurs. For example:
Router# copy system:running-config nvram:startup-config
Building configuration...
It might take a minute or two to save the configuration. After the configuration has been saved, the
following output appears:
[OK]
Router#
On most platforms, this task saves the configuration to NVRAM. On the Class A Flash file system
platforms, this task saves the configuration to the location specified by the CONFIG_FILE environment
variable. The CONFIG_FILE variable defaults to NVRAM.
Filtering Output from the show and more Commands
In Cisco IOS Release 12.0(1)T and later releases, you can search and filter the output of show and more
commands. This functionality is useful if you need to sort through large amounts of output or if you want
to exclude output that you need not see.
To use this functionality, enter a show or more command followed by the “pipe” character (|); one of the
keywords begin, include, or exclude; and a regular expression on which you want to search or filter (the
expression is case-sensitive):
command | {begin | include | exclude} regular-expression
The output matches certain lines of information in the configuration file. The following example
illustrates how to use output modifiers with the show interface command when you want the output to
include only lines in which the expression “protocol” appears:
Router# show interface | include protocol
FastEthernet0/0 is up, line protocol is up
Serial4/0 is up, line protocol is up
Serial4/1 is up, line protocol is up
Serial4/2 is administratively down, line protocol is down
Serial4/3 is administratively down, line protocol is down
For more information on the search and filter functionality, refer to the “Using the Command-Line
Interface” chapter in the Cisco IOS Configuration Fundamentals Configuration Guide.
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Using Cisco IOS Software
Identifying Supported Platforms
Identifying Supported Platforms
Cisco IOS software is packaged in feature sets consisting of software images that support specific
platforms. The feature sets available for a specific platform depend on which Cisco IOS software images
are included in a release. To identify the set of software images available in a specific release or to find
out if a feature is available in a given Cisco IOS software image, see the following sections:
•
Using Feature Navigator
•
Using Software Release Notes
Using Feature Navigator
Feature Navigator is a web-based tool that enables you to quickly determine which Cisco IOS software
images support a particular set of features and which features are supported in a particular Cisco IOS
image.
Feature Navigator is available 24 hours a day, 7 days a week. To access Feature Navigator, you must have
an account on Cisco.com. If you have forgotten or lost your account information, e-mail the Contact
Database Administration group at [email protected] If you do not have an account on Cisco.com,
go to http://www.cisco.com/register and follow the directions to establish an account.
To use Feature Navigator, you must have a JavaScript-enabled web browser such as Netscape 3.0 or later,
or Internet Explorer 4.0 or later. Internet Explorer 4.0 always has JavaScript enabled. To enable
JavaScript for Netscape 3.x or Netscape 4.x, follow the instructions provided with the web browser. For
JavaScript support and enabling instructions for other browsers, check with the browser vendor.
Feature Navigator is updated when major Cisco IOS software releases and technology releases occur.
You can access Feature Navigator at the following URL:
http://www.cisco.com/go/fn
Using Software Release Notes
Cisco IOS software releases include release notes that provide the following information:
•
Platform support information
•
Memory recommendations
•
Microcode support information
•
Feature set tables
•
Feature descriptions
•
Open and resolved severity 1 and 2 caveats for all platforms
Release notes are intended to be release-specific for the most current release, and the information
provided in these documents may not be cumulative in providing information about features that first
appeared in previous releases.
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Identifying Supported Platforms
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Voice, Video, and Fax Overview
The Cisco IOS Voice, Video, and Fax Configuration Guide shows you how to configure your Cisco router
or access server to support voice, video, and fax applications. This chapter is an overview of some of the
concepts and technologies described in the guide.
Configuration Guide Overview
The Cisco IOS Voice, Video, and Fax Configuration Guide is the result of reorganizing and renaming the
Cisco IOS Multiservice Applications Configuration Guide. The reorganized publication is divided into
the following parts:
•
Basic Voice Configuration
•
H.323 Support and Other VoIP Call Control Signaling Protocols
•
Voice over Layer 2 Protocols
•
Telephony Applications
•
Trunk Management and Conditioning Features
•
Fax, Video, and Modem Support
Each part contains one or more chapters that describe configuration procedures for each respective
technology. The following sections describe some of the chapter contents for this configuration guide.
Dial Peers
Dial peers describe the entities to or from which a call is established and the key to understanding the
Cisco voice implementation. All voice technologies use dial peers to define the characteristics associated
with a call leg. A call leg is a discrete segment of a call connection that lies between two points in the
connection. An end-to-end call comprises four call legs, two from the perspective of the source route,
and two from the perspective of the destination route.
You use dial peers to apply specific attributes to call legs and to identify call origin and destination.
Attributes applied to a call leg include specific quality of service (QoS) features (such as IP RTP Priority
and IP Precedence), compression/decompression (codec), voice activity detection (VAD), and fax rate.
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Voice, Video, and Fax Overview
Configuration Guide Overview
There are basically two different kinds of dial peers with each voice implementation:
•
Plain old telephone service (POTS)—Dial peer describing the characteristics of a traditional
telephony network connection. POTS peers point to a particular voice port on a voice network
device.
When you configure POTS dial peers, the key commands that you must be configure are the port
and destination-pattern commands. The destination-pattern command defines the telephone
number associated with the POTS dial peer. The port command associates the POTS dial peer with
a specific logical dial interface, normally the voice port connecting the Cisco device to the local
POTS network.
Specific applications, such as interactive voice response (IVR), are configured on the POTS dial
peer as well.
•
Voice network (VoIP, VoATM, and VoFR)—Dial peer describing the characteristics of a packet
network connection; in the case of VoIP, for example, it is an IP network. Voice-network peers point
to specific voice-network devices.
When you configure voice-network dial peers, the key commands that you must configure are the
destination-pattern and session-target commands. The destination-pattern command defines the
telephone number associated with the voice-network dial peer. The session-target command
specifies a destination address for the voice-network peer.
Other applications (such as store-and-forward fax, which uses the infrastructure of VoIP but is not
strictly a voice technology) also use dial peers to assign attributes to call legs.
Voice Ports
Voice port commands define the characteristics associated with a particular voice-port signaling type.
The Cisco implementation of voice supports both analog and digital telephony connections. The
connection supported (and the associated signaling) depends on the type of voice network module
(VNM) or voice feature card (VFC) installed in your Cisco router or access server.
Voice ports provide support for three basic analog voice signaling formats:
•
FXO—Foreign Exchange Office interface. The FXO interface is an RJ-11 connector that allows a
connection to be directed at the Public Switched Telephone Network (PSTN) central office (CO) (or
to a standard PBX interface, if the local telecommunications authority permits). This interface is of
value for off-premises extension applications.
•
FXS—Foreign Exchange Station interface. The FXS interface is an RJ-11 connector that allows
connection for basic telephone equipment, keysets, and PBXs; FXS connections supply ring,
voltage, and dial tone.
•
E&M—Ear and mouth (or recEive and transMit) interface. The E&M interface is an RJ-48
connector that allows connection for PBX trunk lines (tie lines). It is a signaling technique for 2-wire
and 4-wire telephone and trunk interfaces.
The Cisco MC3810 multiservice concentrator also supports E&M Mercury Exchange Limited
channel-associated signaling (MEL CAS), which is used primarily in the United Kingdom.
Depending on the Cisco device you are configuring, the following digital signaling is supported:
•
ISDN PRI
•
ISDN BRI
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•
E1 R2
•
T1 CAS
The voice port syntax depends on the hardware platform on which it is being configured.
Voice Technologies
Cisco IOS Release 12.2 offers the following voice and voice-related technologies:
•
VoIP
•
Voice over Frame Relay (VoFR)
•
Voice over ATM (VoATM)
•
H.323 gateways
•
Media Gateway Control Protocol (MGCP) and related protocols
•
Session Initiation Protocol (SIP)
•
Tool Command Language (TCL) and interactive voice response (IVR)
•
Multimedia Conference Manager
•
Fax gateways
•
Video
Voice over IP
Cisco offers VoIP that uses IP to carry voice traffic. Because voice traffic is being transported via IP, you
need to configure signaling parameters as part of the voice-port configuration in addition to
feature-specific elements such as dial peers. VoIP is compliant with International Telecommunications
Union-Telecommunications (ITU-T) specifications H.323 and Cisco’s H.323 Version 2.
VoIP can be used to provide the following:
•
A central-site telephony termination facility for VoIP traffic from multiple voice-equipped remote
office facilities.
•
A PSTN gateway for Internet telephone traffic. VoIP used as a PSTN gateway leverages the
standardized use of H.323-based Internet telephone client applications. In the case of a device with
extensive capacity running VoIP (such as the Cisco AS5800 universal access server), it provides the
functionality of a carrier class switch.
VoIP enables Cisco routers and access servers to carry voice traffic (for example, telephone calls and
faxes) over an IP network. In VoIP, the digital signal processor (DSP) segments the voice signal into
frames that are then coupled in groups of two and stored in voice packets. The voice packets are
transported using IP in compliance with ITU-T specification H.323. Because VoIP is a delay-sensitive
application, you must have a well-engineered network end-to-end to use VoIP successfully. Fine-tuning
your network to adequately support VoIP involves a series of protocols and features geared toward QoS.
Traffic shaping considerations must be taken into account to ensure the reliability of the voice
connection.
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Voice over Frame Relay
VoFR uses Frame Relay to transport voice traffic. Because VoFR is transporting signals over Layer 2,
you must configure timing parameters in addition to feature-specific elements such as dial peers and
voice ports. VoFR is compliant with FRF.11 and FRF.12 specifications.
VoFR enables a Cisco device to carry voice traffic (for example, telephone calls and faxes) over a Frame
Relay network. When voice traffic is sent over Frame Relay, the voice traffic is segmented and
encapsulated for transit across the Frame Relay network. The segmentation engine uses FRF.12
fragmentation. FRF.12 (also known as FRF.11 Annex C) allows long data frames to be fragmented into
smaller pieces and interleaved with real-time frames. In this way, real-time voice and nonreal-time data
frames can be carried together on lower speed links without causing excessive delay to the real-time
traffic.
The segmentation size configured must match the line rate or port access rate. To ensure a stable voice
connection, you must configure the same data segmentation size on both sides of the voice connection.
When voice segmentation is configured, all priority queueing, custom queueing, and weighted fair
queueing are disabled on the interface.
When you configure voice and data traffic over the same Frame Relay data-link connection identifier
(DLCI), you must take traffic-shaping considerations into account to ensure the reliability of the voice
connection.
Cisco VoFR implementation supports the following types of VoFR calls:
•
Static FRF.11 trunks
•
Switched VoFR calls:
– Dynamic switched calls
– Cisco trunk (private line) calls
Voice over ATM
VoATM uses ATM adaptation layer 5 (AAL5) to route voice traffic. Because VoATM is transporting
signals over Layer 2, you must configure timing parameters in addition to feature-specific elements such
as dial peers and voice ports.
VoATM enables a Cisco MC3810 multiservice concentrator to carry voice traffic (for example, telephone
calls and faxes) over an ATM network. The Cisco MC3810 multiservice concentrator supports
compressed VoATM on ATM port 0 only.
When voice traffic is sent over ATM, the voice traffic is encapsulated using a special AAL5
encapsulation for multiplexed voice. The ATM permanent virtual circuit (PVC) must be configured to
support real-time voice traffic, and the AAL5 voice encapsulation must be assigned to the PVC. The
PVC must also be configured to support variable bit rate (VBR) for real-time networks for traffic shaping
between voice and data PVCs.
Traffic shaping is necessary so that the carrier does not discard the incoming calls from the
Cisco MC3810 multiservice concentrator. To configure voice and data traffic shaping, you must
configure the peak, average, and burst options for voice traffic. Configure the burst value if the PVC will
be carrying bursty traffic. The peak, average, and burst values are needed so the PVC can effectively
handle the bandwidth for the expected number of voice calls.
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H.323 Gateways
The H.323 standard provides for sending audio, video, and data conferencing data on an IP-based
internetwork. The Cisco functionality enables gateway H.323 terminals to communicate with terminals
running other protocols. Gateways provide protocol conversion between terminals running different
types of protocols. Gatekeepers are optional nodes that manage other nodes in an H.323 network.
Gateways communicate with gatekeepers using the registration, admission, and status (RAS) protocol.
The gatekeeper maintains resource computing information, which it uses to select the appropriate
gateway during the admission of a call.
Cisco software complies with the mandatory requirements and several of the optional features of the
H.323 Version 2 specification. Cisco H.323 Version 2 software enables gatekeepers, gateways, and
proxies to send and receive all the required fields in H.323 Version 2 messages. Cisco H.323 Version 2
features include the following:
•
Lightweight registration
•
Improved gateway selection process
•
Gateway resource availability reporting
•
Support for single proxy configurations
•
Tunneling of redirecting number information element
•
H.245 tunneling
•
Hookflash relay
•
H.235 security
•
Codec negotiation
•
H.245 empty capabilities set
Media Gateway Control Protocol
Media Gateway Control Protocol (MGCP) defines the call control relationship between VoIP gateways
that translate audio signals to and from the packet network and call agents (CAs). The CAs are
responsible for processing the calls. The MGCP gateways interact with a CA, also called a Media
Gateway Controller (MGC) that performs signal and call processing on gateway calls. In the MGCP
configurations supported by Cisco, the gateway can be a Cisco router, access server, or cable modem,
and the CA is a third-party server.
Session Initiation Protocol
Session Initiation Protocol (SIP) is an alternative protocol developed by the Internet Engineering Task
Force (IETF) for multimedia conferencing over IP. SIP features are compliant with IETF RFC 2543, SIP:
Session Initiation Protocol, published in March 1999.
The Cisco SIP functionality enables Cisco access platforms to signal the setup of voice and multimedia
calls over IP networks. The SIP feature also provides nonproprietary advantages in the following areas:
•
Protocol extensibility
•
System scalability
•
Personal mobility services
•
Interoperability with different vendors
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SIP is an ASCII-based, application-layer control protocol that can be used to establish, maintain, and
terminate calls between two or more endpoints.
Like other VoIP protocols, SIP is designed to address the functions of signaling and session management
within a packet telephony network. Signaling allows call information to be carried across network
boundaries. Session management provides the ability to control the attributes of an end-to-end call.
SIP provides the following capabilities:
•
Determining the location of the target endpoint—SIP supports address resolution, name mapping,
and call redirection.
•
Determining the media capabilities of the target endpoint—Through Session Description Protocol
(SDP), SIP determines the lowest level of common services between the endpoints. Conferences are
established using only the media capabilities that can be supported by all endpoints.
•
Determining the availability of the target endpoint—If a call cannot be completed because the target
endpoint is unavailable, SIP determines whether the called party is connected to a call already or did
not answer in the allotted number of rings. SIP then returns a message indicating why the target
endpoint was unavailable.
•
Establishing a session between the originating and target endpoints—If the call can be completed,
SIP establishes a session between the endpoints. SIP also supports midcall changes, such as the
addition of another endpoint to the conference or the changing of a media characteristic or codec.
•
Handling the transfer and termination of calls—SIP supports the transfer of calls from one endpoint
to another. During a call transfer, SIP simply establishes a session between the transferee and a new
endpoint (specified by the transferring party) and terminates the session between the transferee and
the transferring party. At the end of a call, SIP terminates the sessions among all parties.
Interactive Voice Response
IVR consists of simple voice prompting and digit collection to gather caller information for
authenticating the user and identifying the destination. IVR applications can be assigned to specific ports
or invoked on the basis of Dialed Number Identification Service (DNIS). An IP public switched
telephone network gateway can have several IVR applications to accommodate many different gateway
services, and you can customize the IVR applications to present different interfaces to the various
callers.
IVR systems provide information in the form of recorded messages over telephone lines in response to
user input in the form of spoken words, or more commonly dual tone multifrequency (DTMF) signaling.
For example, when a user makes a call with a debit card, an IVR application is used to prompt the caller
to enter a specific type of information, such as an account number. After playing the voice prompt, the
IVR application collects the predetermined number of touch tones and then places the call to the
destination phone or system.
IVR uses TCL scripts to gather information and to process accounting and billing. For example, a TCL
IVR script plays when a caller receives a voice-prompt instruction to enter a specific type of information,
such as a personal identification number (PIN). After playing the voice prompt, the TCL IVR application
collects the predetermined number of touch tones and sends the collected information to an external
server for user authentication and authorization.
Since the introduction of the Cisco IVR technology, the software has undergone several enhancements.
Cisco TCL IVR Version 2.0 is made up of separate components that are described individually in the
sections that follow. The enhancements are as follows:
•
MGCP scripting package implementation
•
Real Time Streaming Protocol (RTSP) client implementation
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•
TCL IVR prompt playout and digit collection on IP call legs
•
New TCL verbs to use RTSP and MGCP scripting features
The enhancements add scalability and enable the TCL IVR scripting functionality on VoIP legs. In
addition, support for RTSP enables VoIP gateways to play messages from RTSP-compliant
announcement servers. The addition of these enhancements also reduces the CPU load and saves
memory on the gateway because no packetization is involved. Larger prompts can be played, and the use
of an external audio server is allowed.
Multimedia Conference Manager
The Multimedia Conference Manager provides both gatekeeper and proxy capabilities, which are
required for service provisioning and management of H.323 networks. With Multimedia Conference
Manager you can configure your current internetwork to route bit-intensive data such as audio,
telephony, video and audio telephony, and data conferencing using existing telephone and ISDN links,
without degrading the current level of service in the network. In addition, you can implement
H.323-compliant applications on existing networks in an incremental fashion without upgrades.
With Multimedia Conference Manager, you can provide the following services:
•
Identification of H.323 traffic and application of appropriate policies
•
Limiting of H.323 traffic on LANs and WANs
•
User accounting for records based on service utilization
•
Insertion of QoS for the H.323 traffic generated by applications such as VoIP, data conferencing, and
video conferencing
•
Implementation of security for H.323 communications
Video
Cisco 2600 series, 3600 series, and 7200 series routers and the Cisco MC3810 multiservice concentrator
support the H.323 gatekeeper (sometimes referred to as Multimedia Conference Manager) with voice
gateway image with Resource Reservation Protocol (RSVP) to ATM SVC mapping. This feature delivers
H.323 gatekeeper, proxy, and voice gateway solutions with routing as a single Cisco IOS image. In
addition, it enables H.323 RSVP reservations to be mapped to ATM non-real-time variable bit rate
(nRTVBR) SVCs to guarantee quality of service (QoS) for video applications over ATM backbones.
Cisco supports video traffic within a data stream in three ways:
•
Video in pass-through mode—Using this method, video traffic received from a video codec
connected to a universal I/O serial port can be transported on a dedicated time slot between systems
using the time-division multiplexing (TDM) functionality of the T1/E1 trunk.
•
Video over ATM AAL1—A serial stream from a video codec connected to a serial port can be
converted to ATM and transported across an ATM network using AAL1 circuit emulation service
(CES) encapsulation.
•
Video over ATM PVCs and switched virtual circuits (SVCs)—A serial stream from a video codec
connected to a Cisco MC3810 multiservice concentrator using the plug-in Video Dialing Module
(VDM) can be converted to ATM and transported across an ATM network using AAL1 CES.
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Fax Gateways
Fax applications enable Cisco AS5300 universal access servers to send and receive faxes across
packet-based networks using modems or VFCs. Some of the benefits of the fax gateway are as follows:
•
Universal inbox for fax and e-mail—Faxes and e-mails can go to the same mailbox using DID
numbers. E-mail and fax recipients can be combined.
•
Toll bypass—In an enterprise environment in which offices in different cities are connected using a
WAN, toll charges can be bypassed by transmitting faxes over the network connection. Because a
fax message is stored on the mail server until Simple Mail Transfer Protocol (SMTP) forwards
messages to the recipient, SMTP can forward fax e-mail attachments during off-peak hours (for
example, during evenings and weekends), thereby reducing long-distance charges.
•
Broadcast to multiple recipients—E-mail fax attachments can be sent to multiple recipients
simultaneously.
•
Improve robustness—The Fax Relay Packet Loss Concealment feature improves the robustness of
the facsimile relay. It eliminates fax failures and lost data caused by excessive page errors. Field
diagnostics and troubleshooting capabilities are improved by available debug commands. Statistics
give better visibility into the real-time fax operation in the gateway, allowing for improved field
diagnostics and troubleshooting.
•
Cost savings and port density using T.37/T.38 Fax Gateway—The cost of maintaining one
architecture (either fax or voice) is eliminated. Service providers can do the following:
– Use a single port for voice, fax relay, and store-and-forward fax. For smaller points of presence
(POPs), the single-port configuration for these technologies is even more significant because
mixed traffic can be handled more efficiently, requiring only a single pool of ports versus
splitting traffic across two pools.
– Offer the new service of a single number for subscriber voice and fax access. The applications
that use a single number for voice and fax require only half as many DNIS numbers and dial
peers as would be required with separate voice and fax applications.
– Offer applications that require toggling from voice to fax. Applications such as never-busy fax
service can be addressed once the gateway can dynamically switch from fax relay to fax store
and forward.
•
Interoperability with T.37 fax relay for VoIP H.323—The Cisco 2600 and 3600 series routers and
Cisco MC3810 multiservice concentrator gateways with ITU-T T.38 fax relay capability can
interoperate with third-party gateways and gatekeepers over an IP H.323 network. The goal is to
work with third-party gateways and gatekeepers to provide ITU-T standards-based T.38 fax relay
services for multivendor networks.
The Cisco 2600 and 3600 series routers and Cisco MC3810 multiservice concentrator gateways provide
standards-based toll bypass for fax and voice calls. In addition to existing voice and fax toll bypass
capabilities, the multiservice gateways provide toll bypass for fax relay with the standards-based ITU-T
T.38 fax relay implementation.
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Configuring Voice over IP
This chapter provides an overview of Voice over IP (VoIP) technology and gives step-by-step
configuration tasks. The chapter contains the following sections:
•
VoIP Benefits, page 12
•
VoIP Call Processing, page 12
•
VoIP Prerequisite Tasks, page 13
•
VoIP Network Design Considerations, page 14
•
VoIP Configuration Task List, page 15
•
Configuring VoIP over Frame Relay, page 17
•
VoIP Configuration Examples, page 18
To identify the hardware platform or software image information associated with a feature in this
chapter, use the Feature Navigator on Cisco.com to search for information about the feature or refer to
the software release notes for a specific release. For more information, see the “Identifying Supported
Platforms” in the “Using Cisco IOS Software” chapter.
Voice over IP Overview
VoIP is a Layer 3 network protocol that uses various Layer 2 point-to-point or link-layer protocols such
as PPP, Frame Relay, or ATM for its transport. VoIP enables Cisco routers, access servers, and
multiservice access concentrators to carry and send voice and fax traffic over an IP network. In VoIP,
digital signal processors (DSPs) segment the voice signal into frames and store them in voice packets.
These voice packets are transported via IP in compliance with a voice communications protocol or
standard such as H.323, Media Gateway Control Protocol (MGCP), or Session Initiation Protocol (SIP).
Table 3 shows the relationship between the Open System Interconnection (OSI) reference model and the
protocols and functions of VoIP network elements.
Table 3
Relationship of OSI Reference Model to VoIP Protocols and Functions
OSI Layer Number
OSI Layer Name
VoIP Protocols and Functions
7
Application
NetMeeting/Applications
6
Presentation
Codecs
5
Session
H.323/MGCP/SIP
4
Transport
RTP/TCP/UDP
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Table 3
Relationship of OSI Reference Model to VoIP Protocols and Functions (continued)
OSI Layer Number
OSI Layer Name
VoIP Protocols and Functions
3
Network
IP
2
Data Link
Frame Relay, ATM, Ethernet, PPP, MLP, and
more
Cisco IOS software supports the following call control protocols and standards in Release 12.2:
•
H.323—the International Telecommunication Union-Telecommunications Standardization Sector
(ITU-T) specification for sending voice, video, and data across a network. The H.323 specification
includes several related standards, such as H.225 (call control), H.235 (security), H.245 (media path
and parameter negotiation), and H.450 (supplementary services). For more information, see the
“H.323 Overview” chapter in this configuration guide.
•
MGCP—Media Gateway Control Protocol, an Internet Engineering Task Force (IETF) draft
standard for controlling voice gateways through IP networks. For more information, see the
“Configuring MGCP and Related Protocols” chapter in this configuration guide.
•
SIP—Session Initiation Protocol, defined in IETF RFC 2543. For more information, see the
“Configuring SIP” chapter in this guide.
VoIP protocols typically use Real-time Transport Protocol (RTP) for the media stream or speech path.
RTP uses User Datagram Protocol (UDP) as its transport protocol. Voice signaling traffic often uses
Transmission Control Protocol (TCP) as its transport medium. The IP layer provides routing and
network-level addressing; the data-link layer protocols control and direct the transmission of the
information over the physical medium.
The main factor in choosing between VoIP and the Layer 2 VoFR and VoATM transport alternatives is
interworking with other voice or multimedia applications. Generally speaking, Voice over Frame Relay
(VoFR) and Voice over ATM (VoATM) are effective WAN transport technologies and are more
bandwidth-efficient than VoIP. But VoFR and VoATM cannot be deployed over LANs or to the desktop.
VoIP is the predominant form of voice-over-packet deployed today, and, for implementing voice
applications, it is usually the only choice even if the first step in network deployment is pure transport
between existing PBXs.
VoIP leverages the entire Internet and Intranet IP infrastructure for routing, making it easy to design
any-to-any calling in a VoIP network. VoIP also allows multivendor interworking, which is more difficult
to achieve with VoFR and VoATM applications because standards for those solutions have only recently
emerged.
Cisco VoIP is frequently used in two primary applications:
•
To provide a central-site telephony termination facility for voice traffic coming from multiple
voice-equipped remote office facilities. Figure 2 illustrates this application using Cisco AS5300
universal access servers as the central-site telephony termination devices.
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Figure 2
VoIP Used as a Central-Site Telephony Termination Facility
729 411-5002
729 411-5003
408 555-1001
729 411-5001
729 411-5004
T1
ISDN PRI
Voice port
0:D
408 555-2001
MGW
Voice port 0:D
IP
cloud
WAN
10.1.1.1
WAN
10.1.1.2
MGW
10351
1:D
T1 ISDN PRI
408 555-3001
•
To provide Public Switched Telephone Network (PSTN) gateway functionality for Internet
telephone traffic. Cisco VoIP used in this scenario leverages the standardized use of H.323-based
Internet telephone client applications. In the case of a device with extensive capacity running VoIP
(such as the Cisco AS5800 universal access server), the functionality provided is equivalent to that
of a carrier-class switch.
Figure 3 illustrates this application, using a Cisco AS5300 as the PSTN gateway.
Figure 3
VoIP Used as a PSTN Gateway for Internet Telephone Traffic
408 526-4000
408 526-4001
408 526-4002
PSTN
408 526-4003
310 520-1001
310 520-1002
Central
office
310 520-1000
310 520-1003
Voice port
1/0/0
Cisco AS5300
10.1.1.1
10.1.1.2
Cisco 3640
10352
IP
cloud
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VoIP Benefits
To use VoIP, you must install the appropriate hardware in your Cisco device: for example, a
voice-specific port adapter or network module. The specific voice hardware required depends on the
router or access server used. The number of ports or channels available for sending VoIP data depends
on the capacity of the specific voice hardware installed. For more information about the physical
characteristics, capacity, installation, or configuration of voice hardware, refer to the online
documentation for your router or access server.
VoIP Benefits
VoIP offers the following benefits:
•
Toll bypass (either one- or two-stage toll bypass, depending on the environment in which VoIP is
deployed)
•
Remote PBX presence over WANs
•
PSTN voice-traffic and fax-traffic offload
•
Universally accessible voice-mail and fax-mail services
•
Unified voice and data trunking
•
Plain old telephone service (POTS)-Internet telephony gateways
•
Support for Microsoft NetMeeting when a Cisco router is used as a voice gateway
VoIP Call Processing
Before configuring VoIP on a Cisco router or access server, it helps to have a high-level understanding
of what happens when you place a VoIP call. The following sequence outlines the general flow of a
two-party VoIP voice call using H.323:
1.
The caller picks up the handset, signaling an off-hook condition to the signaling application layer
of VoIP.
2.
The session application layer of VoIP issues a dial tone and waits for the caller to dial.
3.
When the caller dials the number, the dialed digits are accumulated and stored by the session
application.
4.
After enough digits are accumulated to match a configured destination pattern, the telephone
number is mapped to an IP host via the dial plan mapper. The IP host has a direct connection to the
destination telephone number or a PBX that is responsible for completing the call to the configured
destination pattern.
5.
The session application runs the H.323 session protocol to establish a transmission and a reception
channel for each direction over the IP network. If the call is being handled by a PBX, the PBX
forwards the call to the destination telephone. If Resource Reservation Protocol (RSVP) has been
configured, the RSVP reservations are put into effect to achieve the desired quality of service (QoS)
over the IP network.
6.
The coder-decoders (codecs) are enabled for both ends of the connection and the conversation
proceeds using RTP/UDP/Internet Protocol (IP) as the protocol stack. Voice signals are digitized,
compressed, packaged into discrete packets, and transported over the network.
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VoIP Prerequisite Tasks
7.
Any call-progress indications or other signals that can be carried in-band are cut through the voice
path as soon as the end-to-end audio channel is established. Signaling that can be detected by the
voice ports (for example, in-band dual tone multifrequency [DTMF] digits after the call setup is
complete) is also trapped by the session application at either end of the connection and carried over
the IP network encapsulated in Real Time Conferencing Protocol (RTCP) using the RTCP APP
extension mechanism.
8.
When either end of the call hangs up, the RSVP reservations are torn down (if RSVP is used) and
the session ends. Each end becomes idle, waiting for the next off-hook condition to trigger another
call setup.
VoIP Prerequisite Tasks
Before configuring a Cisco router, access server, or gateway to use VoIP, complete the following tasks:
•
Establish a working IP network in which delay (as measured by ping tests) and jitter are minimized.
For more information about configuring IP, refer to the “IP Overview,” “Configuring IP
Addressing,” and “Configuring IP Services” chapters in the Cisco IOS IP Configuration Guide,
Release 12.2.
•
Install a voice network module (VNM), voice feature card (VFC), or universal port dial feature card
into the appropriate slot of your Cisco router, access server, or gateway. For more information about
the physical characteristics, capacity, memory requirements, and installation instructions for the
hardware you are installing, refer to the appropriate platform-specific hardware documentation.
•
Make sure your router, access server, or gateway has sufficient DRAM installed to support VoIP, and
make sure you are running a version and image of Cisco IOS software that supports VoIP. For more
information, refer to the release notes for the platform you are using and the version of Cisco IOS
you are running, or use the Feature Navigator tool on Cisco.com.
•
Complete basic configuration of your router, access server, or gateway. For more information about
these basic configuration tasks, refer to the “Configuring H.323 Gateways,” “Configuring Voice
Ports,” and “Configuring Dial Plans, Dial Peers, and Digit Manipulation” chapters of this
configuration guide.
•
Formulate the beginning of a dial plan that includes the following:
– Logical network diagram showing voice ports and components to which they connect, including
telephones, fax machines, PBX or key systems, other voice devices that require connection, and
voice-enabled routers.
– Connection details, including physical interfaces, relevant LAN and WAN ports, and all voice
ports; for each WAN, the type (Frame Relay, PPP, etc.); for Frame Relay, relevant PVCs and
link access rates.
– Phone numbers or extensions for each voice port, logically laid out and consistent with existing
private dial plans and external dialing schemes.
•
Establish a working telephony network based on your company dial plan.
•
Integrate your dial plan and telephony network into your existing IP network topology. In general,
we recommend the following practices:
– Make routing or dialing transparent to users; for example, avoid secondary dial tones from
secondary switches, where possible.
– Contact your PBX vendor for instructions about how to reconfigure the appropriate PBX
interfaces.
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VoIP Network Design Considerations
VoIP Network Design Considerations
You must have a well-engineered network end-to-end when running delay-sensitive applications such as
VoIP. Fine-tuning your network to adequately support VoIP involves a series of protocols and features
geared toward improving quality of service (QoS).
Quality of service refers to the ability of a network to provide differentiated service to selected network
traffic over various underlying technologies. QoS is not inherent in a network infrastructure. Rather, you
institute QoS by strategically enabling appropriate QoS features throughout your network.
Cisco IOS software provides many tools for enabling QoS on your backbone, such as Random Early
Detection (RED), Weighted Random Early Detection (WRED), fancy queueing (meaning custom,
priority, or weighted fair queueing), IP RTP priority, low-latency queueing (LLQ), and IP precedence.
To configure your IP network for real-time voice traffic, you must take into consideration the entire
scope of your network and then select the appropriate QoS tool or tools. For complete information about
any of these topics, refer to the Cisco IOS Quality of Service Solutions Configuration Guide, Release
12.2. In addition, refer to the “Configuring QoS for Voice” chapter in this configuration guide.
Remember that to improve voice network performance, QoS must be configured throughout your
network, not just on the Cisco devices running VoIP. Not all QoS techniques are appropriate for all
network routers. Edge routers and backbone routers in your network do not necessarily perform the same
operations; the QoS tasks they perform might differ as well. To configure your IP network for real-time
voice traffic, you must consider the functions of both edge and backbone routers in your network and
then select the appropriate QoS tool or tools.
VoIP Quality of Service Tips
This section explains the quality issues that you should consider when building VoIP networks and offers
a few tips about configuring VoIP with the appropriate QoS. For detailed information on these topics,
refer to “Voice Quality Tuning Commands” in the “Configuring Voice Ports” chapter.
Voice traffic differs from data traffic in the following ways:
•
Data is often bursty by nature; voice is deterministic (smooth).
•
Data applications resend dropped packets; voice applications can only conceal dropped packets.
•
Data applications can usually tolerate some delay; voice applications must minimize delay, so that
the recipient does not hear clips in the transmission.
These differences mandate the use of QoS strategies to give strict priority to voice traffic, ensuring
reliable delivery and minimal delay for networks that carry both voice and data.
Delay
Delay is the time it takes for VoIP packets to travel between two endpoints. Because of the speed of
network links and the processing power of intermediate devices, some delay is expected; however, you
should attempt to minimize this delay.
The human ear normally accepts a delay of about 150 milliseconds (ms) without noticing problems. (The
ITU G.114 standard recommends no more than 150 ms of one-way delay.) When delay exceeds 150 ms,
a conversation becomes more and more like a citizens band (CB) radio interchange in which one person
must wait for the other to stop speaking before beginning to talk. This type of delay is often evident on
international long-distance calls. You can measure delay fairly easily by using ping tests at various times
of the day with different network traffic loads. If network delay is excessive, reduce it before deploying
VoIP in your network.
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VoIP Configuration Task List
Jitter
Although delay can cause unnatural starting and stopping of conversations, variable-length delays (also
known as jitter) can cause a conversation to break and become unintelligible. Jitter is not usually a
problem with PSTN calls because the bandwidth of calls is fixed. However, in VoIP networks in which
existing data traffic might be bursty, jitter can become a problem. Cisco voice gateways have built-in
dejitter buffering to compensate for a certain amount of jitter, but if jitter is constant on a network,
identify the source and control it before deploying a VoIP network.
Serialization
Serialization is a term that describes what happens when a router attempts to send both voice and data
packets through an interface. In general, voice packets are very small (80 to 256 bytes), and data packets
can be very large (1500 to 18,000 bytes). On relatively slow links, such as WAN connections, large data
packets can take a long time to send onto the wire. When these large packets are mixed with smaller
voice packets, the excessive transmission time can lead to both delay and jitter. You can use
fragmentation to reduce the size of the data packets so that the delay and jitter also decrease.
Bandwidth Consumption
Traditional voice conversations consume 64 kbps of network bandwidth. When this voice traffic is run
though a VoIP network, it can be compressed and digitized by digital signal processors (DSPs built into
the routers. This compression can reduce the calls to sizes as small as 5.3 kbps for voice samples. After
the packets go onto the IP network, the appropriate IP/UDP/RTP headers must be added. This can add a
substantial amount of bandwidth to each call (about 40 bytes per packet). Technologies such as RTP
header compression, however, can reduce the IP header overhead to about two bytes. In addition, VAD
does not send any packets unless there is active speech.
VoIP Configuration Task List
To configure VoIP on a Cisco router or access server, complete the following tasks:
Step 1
Configure your IP network for real-time voice traffic. Fine-tuning your network to adequately support
VoIP involves a series of protocols and features designed to improve QoS. To configure your IP network
for real-time voice traffic, consider the entire scope of your network. Then select and configure the
appropriate QoS tool or tools.
Refer to “Configuring VoIP over Frame Relay” section on page 17, and the “Configuring QoS for Voice”
chapter for information about how to select and configure the appropriate QoS tools to optimize voice
traffic on your network.
Step 2
If you plan to run VoIP over Frame Relay, you must take certain factors into consideration when
configuring VoIP for it to run smoothly over Frame Relay. For example, a public Frame Relay cloud
provides no guarantees for QoS. Refer to the “Configuring VoIP over Frame Relay” section on page 17
for information about deploying VoIP over Frame Relay.
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VoIP Configuration Task List
Step 3
Configure dial peers. Use the dial-peer voice command to define dial peers and switch to the dial-peer
configuration mode. Each dial peer defines the characteristics associated with a call leg. A call leg is a
discrete segment of a call connection that lies between two points in the connection. An end-to-end call
consists of four call legs, two from the perspective of the source access server, and two from the
perspective of the destination access server. Dial peers are used to apply attributes to call legs and to
identify call origin and destination. There are two types of dial peers used for VoIP:
•
POTS—Dial peer describing the characteristics of a traditional telephony network connection.
POTS dial peers point to a particular voice port on a voice network device. To configure a POTS dial
peer, you must configure the associated telephone number and the logical interface. Use the
destination-pattern command to associate a telephone number with a POTS peer. Use the port
command to associate a specific logical interface with a POTS peer. In addition, you can specify
direct inward dialing for a POTS peer by using the direct-inward-dial command.
•
VoIP—Dial peer describing the characteristics of the IP network connection. VoIP dial peers point
to specific VoIP devices. To configure a VoIP dial peer, you must configure the associated
destination telephone number and a destination IP address. Use the destination-pattern command
to define the destination telephone number associated with a VoIP peer. Use the session target
command to specify a destination IP address for a VoIP peer.
Refer to the “Configuring Dial Plans, Dial Peers, and Digit Manipulation” chapter in this
configuration guide for additional information about dial-peer characteristics and configuring dial
peers.
Step 4
Configure number expansion. Use the num-exp command to configure number expansion if your
telephone network is configured so that you can reach a destination by dialing only a portion (an
extension number) of the full E.164 telephone number. Refer to the “Configuring Digit Manipulation
Features” section of the “Configuring Dial Plans, Dial Peers, and Digit Manipulation” chapter of this
guide for information about number expansion.
Step 5
Optimize dial peer and network interface configurations. You can use VoIP dial peers to define
characteristics such as codec, voice activity detection (VAD), and additional QoS parameters (when
RSVP is configured). If you have configured RSVP, use either the req-qos or acc-qos command to
configure QoS parameters. Use the codec command to configure specific voice coder rates. Use the vad
command to disable voice activation detection and the transmission of silence packets. Refer to the
“Configuring Dial Plan Options for VoIP Dial Peers” section of the “Configuring Dial Plans, Dial Peers,
and Digit Manipulation” chapter in this guide for additional information about optimizing dial-peer
characteristics.
Step 6
Configure voice ports. In general, voice-port commands define the characteristics associated with a
particular voice-port signaling type. The following voice signaling types are supported:
•
FXO—Foreign Exchange Office interface
•
FXS—The Foreign Exchange Station interface
•
E&M—The “ear and mouth” interface (also called the “earth and magnet interface, or the “recEive
and transMit” interface)
Under most circumstances, the default voice-port command values are adequate to configure FXO and
FXS ports to transport voice data over your existing IP network. Because of the inherent complexities
involved with PBX networks, E&M ports might need specific voice-port values configured, depending
on the specifications of the devices in your telephony network. For information about configuring voice
ports, refer to the “Configuring Voice Ports” chapter in this guide.
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Configuring VoIP over Frame Relay
Configuring VoIP over Frame Relay
You must consider certain factors when configuring VoIP to ensure that it runs smoothly over Frame
Relay. A public Frame Relay cloud provides no guarantees for QoS. For real-time traffic to be sent in a
timely manner, the data rate must not exceed the committed information rate (CIR) or packets may be
dropped. In addition, Frame Relay traffic shaping and RSVP are mutually exclusive. Remembering this
is particularly important if multiple data link connection identifiers (DLCIs) are carried on a single
interface.
For Frame Relay links with slow output rates (less than or equal to 64 kbps) in which data and voice are
being sent over the same permanent virtual circuit (PVC), we recommend the following solutions:
•
Separate DLCIs for voice and data—By providing a separate subinterface for voice and data, you
can use the appropriate QoS tool for each line. For example, with each DLCI using 32 kbps of a
64-kbps line, you could do the following:
– Apply adaptive traffic shaping to both DLCIs.
– Use RSVP or IP Precedence to prioritize voice traffic.
– Use compressed RTP to minimize voice packet size.
– Use weighted fair queueing to manage voice traffic.
•
•
Lower the maximum transmission unit (MTU) size—Voice packets are generally small. With a
lower MTU size (for example, 300 bytes), large data packets can be broken up into smaller data
packets that can more easily be interwoven with voice packets.
Note
Some applications do not support a smaller MTU size. If you decide to lower the MTU
size, use the ip mtu command; this command affects only IP traffic.
Note
Lowering the MTU size affects data throughput speed.
CIR equal to line rate—Make sure that the data rate does not exceed the CIR. One way you can make
sure that the data rate does not exceed the CIR is through generic traffic shaping. For example, you
could do the following:
– Use IP precedence to prioritize voice traffic.
– Use compressed RTP to minimize voice packet header size.
•
Traffic shaping—Use adaptive traffic shaping to throttle back the output rate based on the backward
explicit congestion notification (BECN). If the feedback from the switch is ignored, both data and
voice packets might be discarded. Because the Frame Relay switch does not distinguish between
voice and data packets, voice packets could be discarded, resulting in a deterioration of voice
quality. For example, you could do the following:
– Use compressed RTP, reduced MTU size, and adaptive traffic shaping based on BECN to hold
the data rate to the CIR.
– Use generic traffic shaping to obtain a low interpacket wait time. For example, set Bc to 4000
to obtain an interpacket wait of 125 ms.
Note
We recommend FRF.12 fragmentation setup rules for VoIP connections over Frame Relay. For more
information, refer to the “Configuring Voice over Frame Relay” chapter.
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VoIP Configuration Examples
VoIP Configuration Examples
This section contains the following configuration examples:
•
VoIP over Frame Relay Configuration Example, page 18
•
VoIP for the Cisco 3600 Series Configuration Examples, page 19
•
VoIP for the Cisco AS5300 Configuration Example, page 26
•
VoIP for the Cisco AS5800 Configuration Example, page 29
VoIP over Frame Relay Configuration Example
For Frame Relay, it is customary to configure a main interface and one subinterface per permanent
virtual circuit (PVC). The following example configures a Frame Relay main interface and a subinterface
so that voice and data traffic can be successfully transported:
interface Serial0/0
ip mtu 300
no ip address
encapsulation frame-relay
no ip route-cache
no ip mroute-cache
fair-queue 64 256 1000
frame-relay ip rtp header-compression
interface Serial0/0.1 point-to-point
ip mtu 300
ip address 40.0.0.7 255.0.0.0
no ip route-cache
no ip mroute-cache
bandwidth 64
traffic-shape rate 32000 4000 4000
frame-relay interface-dlci 16
frame-relay ip rtp header-compression
In this configuration example, the main interface has been configured as follows:
•
Maximum Transmission Unit (MTU) size of IP packets is 300 bytes.
•
No IP address is associated with this serial interface. The IP address must be assigned for the
subinterface.
•
Encapsulation method is Frame Relay.
•
Fair queueing is enabled.
•
IP RTP header compression is enabled.
The subinterface has been configured as follows:
•
MTU size is inherited from the main interface.
•
IP address for the subinterface is specified.
•
Bandwidth is set to 64 kbps.
•
Generic traffic shaping is enabled with 32 kbps CIR where Bc = 4000 bits and Be = 4000 bits.
•
Frame Relay DLCI number is specified.
•
IP RTP header compression is enabled.
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VoIP Configuration Examples
Note
When traffic bursts over the CIR, the output rate is held at the speed configured for the CIR (for
example, traffic will not go beyond 32 kbps if CIR is set to 32 kbps).
For more information about Frame Relay, refer to the “Configuring Frame Relay” chapter in the
Cisco IOS Wide-Area Networking Configuration Guide.
VoIP for the Cisco 3600 Series Configuration Examples
The actual VoIP configuration procedure you complete depends on the topology of your voice network.
The following configuration examples are a starting point. Of course, these configuration examples must
be customized to reflect your network topology.
Configuration examples are supplied for the following sections:
•
FXS-to-FXS Connection Using RSVP, page 19
•
Linking PBX Users with E&M Trunk Lines, page 22
•
PSTN Gateway Access Using FXO Connection, page 24
•
PSTN Gateway Access Using FXO Connection (PLAR Mode), page 25
FXS-to-FXS Connection Using RSVP
The following example shows how to configure VoIP for simple FXS-to-FXS connections.
In this example, a very small company of two offices has decided to integrate VoIP into its existing IP
network. One basic telephony device is connected to Router RLB-1; therefore Router RLB-1 is
configured for one POTS dial peer and one VoIP dial peer. Router RLB-w and Router R12-e establish
the WAN connection between the two offices. Because one POTS telephony device is connected to
Router RLB-2, it is also configured for only one POTS peer and one VoIP peer.
In this example, only the calling end (Router RLB-1) is requesting RSVP. Figure 4 illustrates the
topology of this FXS-to-FXS connection example.
Figure 4
FXS-to-FXS Connection Example
64 kbps
Voice port
1/0/0
IP cloud
Serial port
1/3
1/0
Router
RLB-w
128 kbps
Router
R12-e
Serial port
0/0
Router
RLB-1
Dial peer 1
POTS
Serial port
1/0
1/3
64 kbps
Voice port
1/0/0
Serial port
1/0
Router
RLB-2
Dial peer 2
POTS
S6612
Note
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VoIP Configuration Examples
Router RLB-1
hostname rlb-1
! Create voip dial peer 10
dial-peer voice 10 voip
! Define its associated telephone number and IP address
destination-pattern +4155554000
session target ipv4:40.0.0.1
! Request RSVP
req-qos guaranteed-delay
! Create pots dial peer 1
dial-peer voice 1 pots
! Define its associated telephone number and voice port
destination-pattern +4085554000
port 1/0/0
! Configure serial interface 0/0
interface Serial0/0
ip address 10.0.0.1 255.0.0.0
no ip mroute-cache
! Configure RTP header compression
ip rtp header-compression
ip rtp compression-connections 25
! Enable RSVP on this interface
ip rsvp bandwidth 48 48
fair-queue 64 256 36
clockrate 64000
router igrp 888
network 10.0.0.0
network 20.0.0.0
network 40.0.0.0
Router RLB-w
hostname rlb-w
! Configure serial interface 1/0
interface Serial1/0
ip address 10.0.0.2 255.0.0.0
! Configure RTP header compression
ip rtp header-compression
ip rtp compression-connections 25
! Enable RSVP on this interface
ip rsvp bandwidth 96 96
fair-queue 64 256 3
! Configure serial interface 1/3
interface Serial1/3
ip address 20.0.0.1 255.0.0.0
! Configure RTP header compression
ip rtp header-compression
ip rtp compression-connections 25
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! Enable RSVP on this interface
ip rsvp bandwidth 96 96
fair-queue 64 256 3
! Configure IGRP
router igrp 888
network 10.0.0.0
network 20.0.0.0
network 40.0.0.0
Router R12-e
hostname r12-e
! Configure serial interface 1/0
interface Serial1/0
ip address 40.0.0.2 25.0.0.0
! Configure RTP header compression
ip rtp header-compression
ip rtp compression-connections 25
! Enable RSVP on this interface
ip rsvp bandwidth 96 96
fair-queue 64 256 3
! Configure serial interface 1/3
interface Serial1/3
ip address 20.0.0.2 255.0.0.0
! Configure RTP header compression
ip rtp header-compression
ip rtp compression-connections 25
! Enable RSVP on this interface
ip rsvp bandwidth 96 96
fair-queue 64 256 3
clockrate 128000
! Configure IGRP
router igrp 888
network 10.0.0.0
network 20.0.0.0
network 40.0.0.0
Router RLB-2
hostname r1b-2
! Create pots dial peer 2
dial-peer voice 2 pots
! Define its associated telephone number and voice port
destination-pattern +4155554000
port 1/0/0
! Create voip dial peer 20
dial-peer voice 20 voip
!Define its associated telephone number and IP address
destination-pattern +4085554000
session target ipv4:10.0.0.1
! Configure serial interface 0/0
interface Serial0/0
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VoIP Configuration Examples
ip address 40.0.0.1 255.0.0.0
no ip mroute-cache
! Configure RTP header compression
ip rtp header-compression
ip rtp compression-connections 25
! Enable RSVP on this interface
ip rsvp bandwidth 96 96
fair-queue 64 256 3
clockrate 64000
! Configure IGRP
router igrp 888
network 10.0.0.0
network 20.0.0.0
network 40.0.0.0
Linking PBX Users with E&M Trunk Lines
The following example shows how to configure VoIP to link PBX users with E&M trunk lines.
In this example, a company wants to connect two offices: one in San Jose, California, and the other in
Salt Lake City, Utah. Each office has an internal telephone network using a PBX that is connected to the
voice network by an E&M interface. Both the Salt Lake City and the San Jose offices are using E&M
Port Type II with 4-wire operation and Immediate Start signaling. Each E&M interface connects to the
router using two voice interface connections. Users in San Jose dial 8569 and then the extension number
to reach a destination in Salt Lake City. Users in Salt Lake City dial 4527 and then the extension number
to reach a destination in San Jose.
Figure 5 illustrates the topology of this connection example.
Linking PBX Users with E&M Trunk Lines Example
172.16.1.123
Dial peer
1 POTS
PBX
Dial peer
2 POTS
Voice port
1/0/0
Router SJ
Voice port
1/0/1
San Jose
(408)
Note
IP cloud
Voice port Dial peer
1 POTS
1/0/0
PBX
Router SLC
Voice port
1/0/1
Dial peer
2 POTS
Salt Lake City
(801)
This example assumes that the company already has working IP connection between its two remote
offices.
Router SJ
hostname sanjose
!Configure pots dial peer 1
dial-peer voice 1 pots
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172.16.65.182
S6616
Figure 5
Configuring Voice over IP
VoIP Configuration Examples
destination-pattern 555....
port 1/0/0
!Configure pots dial peer 2
dial-peer voice 2 pots
destination-pattern 555....
port 1/0/1
!Configure voip dial peer 3
dial-peer voice 3 voip
destination-pattern 119....
session target ipv4:172.16.65.182
!Configure the E&M interface
voice-port 1/0/0
signal immediate
operation 4-wire
type 2
voice-port 1/0/1
signal immediate
operation 4-wire
type 2
!Configure the serial interface
interface serial 0/0
description serial interface type dce (provides clock)
clock rate 2000000
ip address 172.16.1.123
no shutdown
Router SLC
hostname saltlake
!Configure pots dial peer 1
dial-peer voice 1 pots
destination-pattern 119....
port 1/0/0
!Configure pots dial peer 2
dial-peer voice 2 pots
destination-pattern 119....
port 1/0/1
!Configure voip dial peer 3
dial-peer voice 3 voip
destination-pattern 555....
session target ipv4:172.16.1.123
!Configure the E&M interface
voice-port 1/0/0
signal immediate
operation 4-wire
type 2
voice-port 1/0/0
signal immediate
operation 4-wire
type 2
!Configure the serial interface
interface serial 0/0
description serial interface type dte
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ip address 172.16.65.182
no shutdown
Note
PBXs should be configured to pass all DTMF signals to the router. We recommend that you do not
configure store and forward tone.
Note
If you change the gain or the telephony port, make sure that the telephony port still accepts DTMF
signals.
PSTN Gateway Access Using FXO Connection
The following example shows how to configure VoIP to link users with the PSTN gateway using an FXO
connection.
In this example, users connected to Router SJ in San Jose, California, can reach PSTN users in Salt Lake
City, Utah, via Router SLC. Router SLC in Salt Lake City is connected directly to the PSTN through an
FXO interface.
Figure 6 illustrates the topology of this connection example.
Figure 6
PSTN Gateway Access Using FXO Connection Example
PSTN user
IP cloud
Router SJ
Router SLC
PSTN
cloud
1(408) 555-4000
San Jose
Note
Voice port
1/0/0
Voice port
Salt Lake City
1/0/0
This example assumes that the company already has a working IP connection between its two remote
offices.
Router SJ
! Configure pots dial peer 1
dial-peer voice 1 pots
destination-pattern +14085554000
port 1/0/0
! Configure voip dial peer 2
dial-peer voice 2 voip
destination-pattern 9...........
session target ipv4:172.16.65.182
! Configure the serial interface
interface serial 0/0
clock rate 2000000
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172.16.1.123
Configuring Voice over IP
VoIP Configuration Examples
ip address 172.16.1.123
no shutdown
Router SLC
! Configure pots dial peer 1
dial-peer voice 1 pots
destination-pattern 9...........
port 1/0/0
! Configure voip dial peer 2
dial-peer voice 2 voip
destination-pattern +14085554000
session target ipv4:172.16.1.123
! Configure serial interface
interface serial 0/0
ip address 172.16.65.182
no shutdown
PSTN Gateway Access Using FXO Connection (PLAR Mode)
The following example shows how to configure VoIP to link users with the PSTN gateway using an FXO
connection in private line auto-ringdown (PLAR) mode.
In this example, PSTN users in Salt Lake City, Utah, can dial a local number and establish a private-line
connection in a remote location. As in the preceding example, Router SLC in Salt Lake City is connected
directly to the PSTN through an FXO interface.
Figure 7 illustrates the topology of this connection example.
Figure 7
PSTN Gateway Access Using FXO Connection (PLAR Mode)
PLAR connection
PSTN user
IP cloud
Router SJ
Router SLC
PSTN
cloud
1(408) 555-4000
Note
Voice port
1/0/0
172.16.65.182
Voice port
1/0/0
Salt Lake City
S6618
172.16.1.123
San Jose
This example assumes that the company already has a working IP connection between its two remote
offices.
Router SJ
! Configure pots dial peer 1
dial-peer voice 1 pots
destination-pattern +14085554000
port 1/0/0
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! Configure voip dial peer 2
dial-peer voice 2 voip
destination-pattern 9...........
session target ipv4:172.16.65.182
! Configure the serial interface
interface serial 0/0
clock rate 2000000
ip address 172.16.1.123
no shutdown
Router SLC
! Configure pots dial peer 1
dial-peer voice 1 pots
destination-pattern 9...........
port 1/0/0
! Configure voip dial peer 2
dial-peer voice 2 voip
destination-pattern +14085554000
session target ipv4:172.16.1.123
! Configure the voice-port
voice-port 1/0/0
connection plar 14085554000
! Configure the serial interface
interface serial 0/0
ip address 172.16.65.182
no shutdown
VoIP for the Cisco AS5300 Configuration Example
This configuration example should give you a starting point in your configuration process. The actual
VoIP configuration procedure you complete depends on the topology of your voice network. These
configuration examples must be customized to reflect your network topology.
Linking PBX Users to a T1 ISDN PRI Interface
This example describes how to configure VoIP to link PBX users with T1 channels configured for ISDN
PRI signaling. In this example, the company has already established a working IP connection between
its two remote offices, one in San Jose, California, and the other in Research Triangle Park (RTP), North
Carolina. Figure 8 illustrates the topology of this example.
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Figure 8
Linking PBX Users to a T1 ISDN PRI Interface Example
729 555-1001
729 555-1002
408 115-1001
729 555-1000
408 116-1002
0:D
WAN
10.1.1.1
IP network
WAN
10.1.1.2
V
Router B
36850
1:D
0:D
Router A
V
729 555-1003
408 117-1003
Each office has an internal telephone network using a PBX that is connected to the voice network by T1
interfaces. The San Jose office, located to the left of the IP cloud, has two T1 connections; the RTP
office, located to the right of the IP cloud, has only one. Both offices are using PRI signaling for the T1
connections.
To reach a destination in RTP, callers in San Jose pick up the handset, hear a primary dial tone, and dial
9, 411, and the destination extension number. To reach a destination in San Jose, callers in RTP pick up
the handset, hear a primary dial tone, and dial 4. After dialing 4, callers hear a secondary dial tone. They
then dial 555 and the extension number.
Configuration for San Jose Access Server
The first part of this configuration example defines dial-in access, including configuring the T1 lines and
the ISDN D-channel parameters:
hostname sanjose
!
! Define the telephone company’s switch type
isdn switch-type primary-5ess
!
! Configure T1 PRI for line 1
controller T1 0
framing esf
clock source line primary
linecode b8zs
pri-group timeslots 1-24
!
! Configure T1 PRI for line 2
controller T1 1
framing esf
clock source line secondary
linecode b8zs
pri-group timeslots 1-24
!
! Configure the ISDN D channel for each ISDN PRI line
! Serial interface 0:23 is the D channel for controller T1 0
!
interface Serial0:23
isdn incoming-voice modem
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!
! Serial interface 1:23 is the D channel for controller T1 1
interface Serial1:23
isdn incoming-voice modem
The next part of this example configures number expansion:
! Configure number expansion.
num-exp 555.... 1408555....
num-exp 4115... 17294115...
The next part of this example configures the POTS and VoIP dial peers:
! Configure POTS dial peer 1 using the first T1
dial-peer voice 1 pots
prefix 6
dest-pat 1408555....
port 0:D
!
! Configure POTS dial-peer 2 using the first T1
dial-peer voice 2 pots
prefix 7
dest-pat 1408555....
port 0:D
!
! Configure POTS dial-peer 3 using the second T1
dial-peer voice 3 pots
prefix 5
dest-pat 1408555....
port 1:D
!
! Configure VoIP dial-peer 4
dial-peer voice 4 voip
dest-pat 17294115...
session-target ipv4:10.1.1.2
Configuration for RTP Access Server
The first part of this configuration example defines dial-in access, including configuring the T1 line and
the ISDN D-channel parameters:
hostname rtp
! Define the telephone company’s switch type
isdn switch-type primary-5ess
! Configure T1 PRI for line 1
controller T1 0
framing esf
clock source line primary
linecode b8zs
pri-group timeslots 1-24
!
! Configure the ISDN D channel for ISDN PRI line 1
! Serial interface 0:23 is the D channel for controller T1 0
interface Serial0:23
ip address 7.1.1.10 255.255.255.0
encapsulation ppp
isdn incoming-voice modem
dialer-group 1
ppp authentication chap
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The next part of this example configures number expansion:
! Configure number expansion.
num-exp 555.... 1408555....
num-exp 4115... 17294115...
The next part of this configuration example defines the POTS and VoIP peers:
! Configure POTS dial-peer 1
dial-peer voice 1 pots
dest-pat 17294115...
port 0:D
!
! Configure VoIP dial-peer 5
dial-peer voice 4 voip
dest-pat 1408555....
session-target ipv4:10.1.1.1
VoIP for the Cisco AS5800 Configuration Example
The following configuration example shows an abbreviated configuration using a Cisco 2600 router and
a Cisco AS5800 universal access server as gateways and a Cisco 3600 router as a gatekeeper. Figure 9
shows the network diagram for this particular scenario.
Figure 9
Cisco AS5800 Universal Access Server Acting As a Gateway
AS5800 VoIP
H.323 gateway
100BASE-T
Cisco 2600
10BASE-T
10BASE-T
10BASE-T
Catalyst
5000
NT Server
Cisco CallManager
10BASE-T
Cisco 3640
gatekeeper
30460
Cisco 2600
5000
Configuring the Cisco 3640 As a Gatekeeper
The following example shows how to configure a Cisco 3640 router as a gatekeeper:
! Configure the Ethernet interface to be used at the gatekeeper interface.
interface Ethernet0/1
ip address 172.30.00.00 255.255.255.0
no ip directed-broadcast
no logging event link-status
no keepalive
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!
! Configure the gatekeeper interface and enable the interface.
gatekeeper
zone local gk3.gg-dn1 gg-dn1 173.50.00.00
zone prefix gk3.gg-dn1 21*
gw-type-prefix 9#* gw ipaddr 173.60.0.0 1720
gw-type-prefix 6#* gw ipaddr 173.60.0.199 1720
no use-proxy gk3.gg-dn1 default inbound-to terminal
no shutdown
!
Configuring the Cisco 2600 As a Gateway
The following example shows how to configure a Cisco 2600 series router as a gateway:
! Configure POTS and VoIP dial peers.
dial-peer voice 88 voip
destination-pattern 11111
tech-prefix 9#
session ras
!
dial-peer voice 11 pots
incoming called-number 11111
destination-pattern 6#12345
port 1/1/1
prefix 12345
!
! Configure the gateway interface.
interface Ethernet0/0
ip address 173.60.0.199 255.255.255.0
no ip directed-broadcast
no ip mroute-cache
no logging event link-status
no keepalive
no cdp enabled
h323-gateway voip interface
h323-gateway voip id gk3.gg-dn1 ipaddr 173.30.0.0 1719
h323-gateway voip h323-id [email protected]
h323-gateway voip tech-prefix 6#
!
Configuring the Cisco AS5800 as a Gateway
The following example shows how to configure the Cisco AS5800 universal access server as a gateway:
! Configure the T1 controller. (This configuration is for a T3 card.)
controller T1 1/0/0:1
framing esf
linecode b8zs
pri-group timeslots 1-24
!
! Configure POTS and VoIP dial peers.
dial-peer voice 11111 pots
incoming called-number 12345
destination-pattern 9#11111
direct-inward-dial
port 1/0/0:1:D
prefix 11111
!
dial-peer voice 12345 voip
destination-pattern 12345
tech-prefix 6#
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session target ras
!
! Enable gateway functionality.
gateway
!
! Enable Cisco Express Forwarding.
ip cef
!
! Configure and enable the gateway interface.
interface FastEthernet0/3/0
ip address 173.60.0.0.255.255.255.0
no ip directed-broadcast
no keepalive
full-duplex
no cdp enable
h323-gateway voip interface
h323-gateway voip id gk3.gg-dn1 ipaddr 173.30.0.0 1719
h323-gateway voip h323-id [email protected]
h323-gateway voip tech-prefix 9#
!
! Configure the serial interface.(This configuration is for a T3 serial interface.)
interface Serial1/0/0:1:23
no ip address
no ip directed-broadcast
ip mroute-cache
isdn switch-type primary-5ess
isdn incoming-voice modem
no cdp enable
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Basic Voice Configuration
Configuring Voice Ports
Voice ports are found at the intersections of packet-based networks and traditional telephony networks,
and they facilitate the passing of voice and call signals between the two networks. Physically, voice ports
connect a router or access server to a line from a circuit-switched telephony device in a PBX or the public
switched telephone network (PSTN).
Basic software configuration for voice ports describes the type of connection being made and the type
of signaling to take place over this connection. Additional commands provide fine-tuning for voice
quality, enable special features, and specify parameters to match those of proprietary PBXs.
This chapter includes the following sections:
•
Voice Port Configuration Overview, page 36
•
Analog Voice Ports Configuration Task List, page 40
•
Configuring Digital Voice Ports, page 54
•
Fine-Tuning Analog and Digital Voice Ports, page 79
•
Verifying Analog and Digital Voice-Port Configurations, page 97
•
Troubleshooting Analog and Digital Voice Port Configurations, page 108
Not all voice-port commands are covered in this chapter. Some are described in the “Configuring Trunk
Connections and Conditioning Features” chapter or the “Configuring ISDN Interfaces for Voice” chapter
in this configuration guide. The voice-port configuration commands included in this chapter are fully
documented in the Cisco IOS Voice, Video, and Fax Command Reference.
To identify the hardware platform or software image information associated with a feature in this
chapter, use the Feature Navigator on Cisco.com to search for information about the feature or refer to
the software release notes for a specific release. For more information, see the “Identifying Supported
Platforms” section in the “Using Cisco IOS Software” chapter.
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Voice Port Configuration Overview
Voice Port Configuration Overview
Voice ports on routers and access servers emulate physical telephony switch connections so that voice
calls and their associated signaling can be transferred intact between a packet network and a
circuit-switched network or device.
For a voice call to occur, certain information must be passed between the telephony devices at either end
of the call, such as the devices’ on-hook status, the line’s availability, and whether an incoming call is
trying to reach a device. This information is referred to as signaling, and to process it properly, the
devices at both ends of the call segment (that is, those directly connected to each other) must use the
same type of signaling.
The devices in the packet network must be configured to convey signaling information in a way that the
circuit-switched network can understand. They must also be able to understand signaling information
received from the circuit-switched network. This is accomplished by installing appropriate voice
hardware in the router or access server and by configuring the voice ports that connect to telephony
devices or the circuit-switched network.
The illustrations below show examples of voice port usage.
•
In Figure 10, one voice port connects a telephone to the wide-area network (WAN) through the
router.
•
In Figure 11, one voice port connects to the PSTN and another to a telephone; the router acts like a
small PBX.
•
Figure 12 shows how two PBXs can be connected over a WAN to provide toll bypass.
Figure 10
Telephone to WAN
WAN
V
Figure 11
37754
Voice port
Serial or
1/0/0
Ethernet port
Telephone to PSTN
PSTN
V
PBX-to-PBX over a WAN
Voice port Serial or
1/0/0 Ethernet port
Serial or Voice port
Ethernet port 1/0/0
PBX
PBX
V
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WAN
V
37756
Figure 12
37755
Voice port Voice port
1/0/0
0/0/1
Configuring Voice Ports
Voice Port Configuration Overview
Cisco provides a variety of Cisco IOS commands for flexibility in programming voice ports to match the
physical attributes of the voice connections that are being made. Some of these connections are made
using analog means of transmission, while others use digital transmission. Table 4 shows the analog and
digital voice-port connection support of the router platforms discussed in this chapter.
Table 4
Analog and Digital Voice-port Support on Cisco Routers and Access Servers
Platform
Analog
Digital
Cisco 803 and 804
Yes
No
Cisco 1750
Yes
No
Cisco 2600 series
Yes
Yes
Cisco 3600 series
Yes
Yes
Cisco MC3810
Yes
Yes
Cisco AS5300
No
Yes
Cisco AS5800
No
Yes
Cisco 7200 series
No
Yes
Cisco 7500 series
No
Yes
Telephony Signaling Interfaces
Voice ports on routers and access servers physically connect the router or access server to telephony
devices such as telephones, fax machines, PBXs, and PSTN central office (CO) switches. These devices
may use any of several types of signaling interfaces to generate information about on-hook status,
ringing, and line seizure.
The router’s voice-port hardware and software need to be configured to transmit and receive the same
type of signaling being used by the device with which they are interfacing so that calls can be exchanged
smoothly between the packet network and the circuit-switched network.
The signaling interfaces discussed in this chapter include foreign exchange office (FXO), foreign
exchange station (FXS), and receive and transmit (E&M), which are types of analog interfaces. Some
digital connections emulate FXO, FXS, and E&M interfaces, and they are discussed in the second half
of this chapter. It is important to know which signaling method the telephony side of the connection is
using, and to match the router configuration and voice interface hardware to that signaling method.
The next three illustrations show how the different signaling interfaces are associated with different uses
of voice ports. In Figure 13, FXS signaling is used for end-user telephony equipment, such as a telephone
or fax machine. Figure 14 shows an FXS connection to a telephone and an FXO connection to the PSTN
at the far side of a WAN; this might be a telephone at a local office going over a WAN to a router at
headquarters that connects to the PSTN. In Figure 15, two PBXs are connected across a WAN by E&M
interfaces. This illustrates the path over a WAN between two geographically separated offices in the
same company.
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Voice Port Configuration Overview
FXS Signaling Interfaces
FXS
WAN
V
FXS
FXS and FXO Signaling Interfaces
Voice port Serial or
1/0/0 Ethernet port
FXS
Figure 15
V
Serial or Voice port
Ethernet port 1/0/0
WAN
V
V
FXO
PSTN
37758
Figure 14
Serial or Voice port
Ethernet port 1/0/0
37757
Voice port Serial or
1/0/0 Ethernet port
E&M Signaling Interfaces
Voice port Serial or
1/0/0 Ethernet port
Serial or Voice port
Ethernet port 1/0/0
PBX
PBX
E&M
V
WAN
V
E&M
37759
Figure 13
FXS and FXO Interfaces
An FXS interface connects the router or access server to end-user equipment such as telephones, fax
machines, or modems. The FXS interface supplies ring, voltage, and dial tone to the station and includes an
RJ-11 connector for basic telephone equipment, keysets, and PBXs.
An FXO interface is used for trunk, or tie line, connections to a PSTN CO or to a PBX that does not
support E&M signaling (when local telecommunications authority permits). This interface is of value
for off-premise station applications. A standard RJ-11 modular telephone cable connects the FXO voice
interface card to the PSTN or PBX through a telephone wall outlet.
FXO and FXS interfaces indicate on-hook or off-hook status and the seizure of telephone lines by one
of two access signaling methods: loop start or ground start. The type of access signaling is determined
by the type of service from the CO; standard home telephone lines use loop start, but business telephones
can order ground start lines instead.
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Voice Port Configuration Overview
Loop-start is the more common of the access signaling techniques. When a handset is picked up (the
telephone goes off-hook), this action closes the circuit that draws current from the telephone company
CO and indicates a change in status, which signals the CO to provide dial tone. An incoming call is
signaled from the CO to the handset by sending a signal in a standard on/off pattern, which causes the
telephone to ring.
Loop-start has two disadvantages, however, that usually are not a problem on residential telephones but
that become significant with the higher call volume experienced on business telephones. Loop-start
signaling has no means of preventing two sides from seizing the same line simultaneously, a condition
known as glare. Also, loop start signaling does not provide switch-side disconnect supervision for FXO
calls. The telephony switch (the connection in the PSTN, another PBX, or key system) expects the
router’s FXO interface, which looks like a telephone to the switch, to hang up the calls it receives through
its FXO port. However, this function is not built into the router for received calls; it only operates for
calls originating from the FXO port.
Another access signaling method used by FXO and FXS interfaces to indicate on-hook or off-hook status
to the CO is ground start signaling. It works by using ground and current detectors that allow the network
to indicate off-hook or seizure of an incoming call independent of the ringing signal and allow for
positive recognition of connects and disconnects. For this reason, ground start signaling is typically used
on trunk lines between PBXs and in businesses where call volume on loop start lines can result in glare.
See the “Disconnect Supervision Commands” section on page 83 and “FXO Supervisory Disconnect
Tone Commands” section on page 85 for voice port commands that configure additional recognition of
disconnect signaling.
In most cases, the default voice port command values are sufficient to configure FXO and FXS voice
ports.
E&M Interfaces
Trunk circuits connect telephone switches to one another; they do not connect end-user equipment to the
network. The most common form of analog trunk circuit is the E&M interface, which uses special
signaling paths that are separate from the trunk’s audio path to convey information about the calls. The
signaling paths are known as the E-lead and the M-lead. The name E&M is thought to derive from the
phrase Ear and Mouth or rEceive and transMit although it could also come from Earth and Magnet. The
history of these names dates back to the days of telegraphy, when the CO side had a key that grounded
the E circuit, and the other side had a sounder with an electromagnet attached to a battery. Descriptions
such as Ear and Mouth were adopted to help field personnel determine the direction of a signal in a wire.
E&M connections from routers to telephone switches or to PBXs are preferable to FXS/FXO
connections because E&M provides better answer and disconnect supervision.
Like a serial port, an E&M interface has a data terminal equipment/data communications equipment
(DTE/DCE) type of reference. In the telecommunications world, the trunking side is similar to the DCE,
and is usually associated with CO functionality. The router acts as this side of the interface. The other
side is referred to as the signaling side, like a DTE, and is usually a device such as a PBX. Five distinct
physical configurations for the signaling part of the interface (Types I-V) use different methods to signal
on-hook/off-hook status, as shown in Table 5. Cisco voice implementation supports E&M Types I, II,
III, and V.
The physical E&M interface is an RJ-48 connector that connects to PBX trunk lines, which are classified
as either two-wire or four-wire. This refers to whether the audio path is full duplex on one pair of wires
(two-wire) or on two pair of wires (four-wire). A connection may be called a four-wire E&M circuit
although it actually has six to eight physical wires. It is an analog connection although an analog E&M
circuit may be emulated on a digital line. For more information on digital voice port configuration of
E&M signaling, see the “DS0 Groups on Digital T1/E1 Voice Ports” section on page 70.
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Analog Voice Ports Configuration Task List
PBXs built by different manufacturers can indicate on-hook/off-hook status and telephone line seizure
on the E&M interface by using any of three types of access signaling that are as follows:
•
Immediate-start is the simplest method of E&M access signaling. The calling side seizes the line by
going off-hook on its E-lead and sends address information as dual-tone multifrequency (DTMF)
digits (or as dialed pulses on Cisco 2600 series routers and Cisco 3600 series routers) following a
short, fixed-length pause.
•
Wink-start is the most commonly used method for E&M access signaling, and is the default for
E&M voice ports. Wink-start was developed to minimize glare, a condition found in immediate-start
E&M, in which both ends attempt to seize a trunk at the same time. In wink-start, the calling side
seizes the line by going off-hook on its E-lead, then waits for a short temporary off-hook pulse, or
“wink,” from the other end on its M-lead before sending address information. The switch interprets
the pulse as an indication to proceed and then sends the dialed digits as DTMF or dialed pulses.
•
In delay-dial signaling, the calling station seizes the line by going off-hook on its E-lead. After a
timed interval, the calling side looks at the status of the called side. If the called side is on-hook, the
calling side starts sending information as DTMF digits; otherwise, the calling side waits until the
called side goes on-hook and then starts sending address information.
Table 5
E&M Wiring and Signaling Methods
M-Lead
E&M Type E-Lead Configuration Configuration
Signal Battery Lead
Configuration
Signal Ground Lead
Configuration
I
Output, relay to
ground
Input, referenced to
ground
—
—
II
Output, relay to SG
Input, referenced to
ground
Feed for M,
connected to –48V
Return for E,
galvanically isolated
from ground
III
Output, relay to
ground
Input, referenced to
ground
Connected to –48V
Connected to ground
V
Output, relay to
ground
Input, referenced to
–48V
—
—
Analog Voice Ports Configuration Task List
Analog voice port interfaces connect routers in packet-based networks to analog two-wire or four-wire
analog circuits in telephony networks. Two-wire circuits connect to analog telephone or fax devices, and
four-wire circuits connect to PBXs. Typically, connections to the PSTN CO are made with digital
interfaces.
This section describes how to configure analog voice ports and covers the following topics:
•
Configuring Codec Complexity for Analog Voice Ports on the Cisco MC3810 with
High-Performance Compression Modules, page 45
•
Configuring Basic Parameters on Analog FXO, FXS, or E&M Voice Ports, page 46
•
Configuring Analog Telephone Connections on Cisco 803 and 804 Routers, page 50
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Three other sections later in the chapter provide help with fine-tuning and troubleshooting:
•
Fine-Tuning Analog and Digital Voice Ports, page 79
•
Verifying Analog and Digital Voice-Port Configurations, page 97
•
Troubleshooting Analog and Digital Voice Port Configurations, page 108
Prerequisites for Configuring Analog Voice Ports
•
Obtain two- or four-wire line service from your service provider or from a PBX.
•
Complete your company’s dial plan.
•
Establish a working telephony network based on your company’s dial plan.
•
Install at least one other network module or WAN interface card to provide the connection to the
network LAN or WAN.
•
Establish a working IP and Frame Relay or ATM network. For more information about configuring
IP, refer to the Cisco IOS IP Configuration Guide, Release 12.2.
•
Install appropriate voice processing and voice interface hardware on the router. See the
“Configuring Platform-Specific Analog Voice Hardware” section on page 43.
Preparing to Configure Analog Voice Ports
Before configuring an analog voice port, assemble the following information about the telephony
connection that the voice port will be making. If connecting to a PBX, it is important to understand the
PBX’s wiring scheme and timing parameters. This information should be available from your PBX
vendor or the reference manuals that accompany your PBX.
•
Telephony signaling interface: FXO, FXS, or E&M
•
Locale code (usually the country) for call progress tones
•
If FXO, type of dialing: DTMF (touch-tone) or pulse
•
If FXO, type of start signal: loop-start or ground-start
•
If E&M, type: I, II, III, or V
•
If E&M, type of line: two-wire or four-wire
•
If E&M, type of start signal: wink, immediate, delay-dial
Table 6 should help you determine which hardware and configuration instructions are appropriate for
your situation. Table 7 on page 42 shows slot and port numbering, which differs for each of the
voice-enabled routers. More current information may be available in the release notes that accompany
the Cisco IOS software you are using.
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Table 6
Telephony
Signaling
Interface
Analog Voice Port Configurations
Router Platform
End user:
Cisco 803
telephone or Cisco 804
fax
FXO
“Configuring Analog Telephone
Connections on Cisco 803 and
804 Routers”
“Configuring Basic Parameters
on Analog FXO, FXS, or E&M
Voice Ports”
MC3810-AVM6
MC3810-APM-FXO
MC3810-AVM6
MC3810-APM-FXS
VIC-2E/M
Cisco 1750
Cisco 2600 series
Cisco 3600 series
Cisco MC3810
Table 7
Section Containing Voice Port
Configuration Instructions
Cisco 1750
VIC-2FXS
Cisco 2600 series
Cisco 3600 series
Cisco MC3810
E&M
—
Cisco 1750
VIC-2FXO, VIC-2FXO-EU
Cisco 2600 series
Cisco 3600 series
Cisco MC3810
FXS
Voice Hardware Required
MC3810-AVM6
MC3810-APM-EM
Analog Voice Slot/Port Designations
Router Platform
Voice Hardware
Chassis Slot
Numbers
Voice NM Slot Voice Port
Numbers
Numbers
Cisco 803, 804
Analog POTS
—
—
—
Cisco 1750
Analog VIC
0 to 1
—
0 to 1
Cisco 2600 series
Voice/fax network module
with two-port VIC
Varies, based
on router
1
0 to 1
Cisco 3600 series
Voice/fax network module
with two-port voice over
interface cards (VICs)
1
3620: 0 to 1
0 to 1
3640: 0 to 3
3660: 1 to 6
Cisco MC3810
Analog voice module (AVM)
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—
1 to 6
Configuring Voice Ports
Analog Voice Ports Configuration Task List
Configuring Platform-Specific Analog Voice Hardware
This section describes the general types of analog voice port hardware available for the router platforms
included in this chapter:
Note
•
Cisco 800 Series Routers, page 43
•
Cisco 1750 Modular Router, page 43
•
Cisco 2600 Series and Cisco 3600 Series Routers, page 44
•
Cisco MC3810 Multiservice Concentrator, page 44
For current information about supported hardware, see the release notes for the platform and
Cisco IOS release being used.
Cisco 800 Series Routers
Cisco 803 and Cisco 804 routers support data and voice applications. The data applications on these
routers are implemented through the ISDN port, and the voice applications are implemented with ISDN
Basic Rate Interface (BRI) through the telephone ports. If a Cisco 803 or 804 router is being used,
connect two devices, such as an analog touch-tone telephone, fax machine, or modem through two fixed
telephone ports, the gray PHONE 1 and PHONE 2 ports that have RJ-11 connectors. Each device is
connected to basic telephone services through the ISDN line.
For more information, refer to the Cisco 800 Series Routers Hardware Installation Guide.
Cisco 1750 Modular Router
The Cisco 1750 modular router provides Voice over IP (VoIP) functionality and can carry voice traffic
(for example, telephone calls and faxes) over an IP network. To make a voice connection, the router must
have a supported VIC installed. The Cisco 1750 router supports two slots for either WAN interface cards
(WICs) or VICs and supports one VIC-only slot. For analog connections, two-port VICs are available to
support FXO, FXS, and E&M signaling. VICs provide direct connections to telephone equipment
(analog phones, analog fax machines, key systems, or PBXs) or to a PSTN.
For more information, refer to the Cisco 1750 Voice-over-IP Quick Start Guide.
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Cisco 2600 Series and Cisco 3600 Series Routers
The Cisco 2600 and 3600 series routers are modular, multifunction platforms that combine dial access,
routing, local area network-to-local area network (LAN) services, and multiservice integration of voice,
video, and data in the same device.
Voice network modules installed in Cisco 2600 series or Cisco 3600 series routers convert telephone
voice signals into data packets that can be transmitted over an IP network. The voice network modules
have no connectors; VICs installed in the network modules provide connections to the telephone
equipment or network. VICs work with existing telephone and fax equipment and are compatible with
H.323 standards for audio and video conferencing.
The Cisco 2600 series router can house one network module. In the Cisco 3600 series, the Cisco 3620
router has slots for up to two network modules; the Cisco 3640 router has slots for up to four network
modules; and the Cisco 3660 router has slots for up to six network modules. (Typically, one of the slots
is used for LAN connectivity.)
For analog telephone connections, low-density voice/fax network modules that contain either one or two
VIC slots are installed in the network module slots. Each VIC is specific to a particular telephone
signaling interface (FXS, FXO, or E&M); therefore, the VIC determines the type of signaling on that
module.
For more information, refer to the following:
•
Cisco 2600 Series Hardware Installation Guide
•
Cisco 3600 Series Hardware Installation Guide
•
Cisco Network Module Hardware Installation Guide
Cisco MC3810 Multiservice Concentrator
To support analog voice circuits, a Cisco MC3810 multiservice concentrator must be equipped with an
AVM, which supports six analog voice ports. By installing specific signaling modules known as analog
personality modules (APMs), the analog voice ports may be equipped for the following signaling types
in various combinations: FXS, FXO, and E&M. For FXS, the analog voice ports use an RJ-11 connector
interface to connect to analog telephones or fax machines (two-wire) or to a key system (four-wire). For
FXO, the analog voice ports use an RJ-11 physical interface to connect to a CO trunk. For E&M
connections, the analog voice ports use an RJ-1CX physical interface to connect to an analog PBX
(two-wire or four-wire).
Optional high-performance voice compression modules (HCMs) can replace standard voice compression
modules (VCMs) to operate according to the voice compression coding algorithm (codec) specified
when the Cisco MC3810 concentrator is configured. The HCM2 provides four voice channels at high
codec complexity and eight channels at medium complexity. The HCM6 provides 12 voice channels at
high complexity and 24 channels at medium complexity. One or two HCMs can be installed in a
Cisco MC3810 multiservice concentrator, but an HCM may not be combined with a VCM in one chassis.
For more information, refer to the Cisco MC3810 Multiservice Concentrator Hardware Installation
Guide.
Note
For current information about supported hardware, see the release notes for the platform and
Cisco IOS release being used.
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Analog Voice Ports Configuration Task List
Configuring Codec Complexity for Analog Voice Ports on the Cisco MC3810
with High-Performance Compression Modules
The term codec stands for coder-decoder. A codec is a particular method of transforming analog voice
into a digital bit stream (and vice versa) and also refers to the type of compression used. Several different
codecs have been developed to perform these functions, and each one is known by the number of the
International Telecommunication Union-Telecommunication Standardization Sector (ITU-T) standard
in which it is defined. For example, two common codecs are the G.711 and the G.729 codecs. The various
codecs use different algorithms to encode analog voice into digital bit-streams and have different bit
rates, frame sizes, and coding delays associated with them. The codecs also differ in the amount of
perceived voice quality they achieve. Specialized hardware and software in the digital signal processors
(DSPs) perform codec transformation and compression functions, and different DSPs may offer different
selections of codecs.
Select the same type of codec as the one that is used at the other end of the call. For instance, if a call
was coded with a G.729 codec, it must be decoded with a G.729 codec. Codec choice is configured on
dial peers. For more information, see the “Configuring Dial Plans, Dial Peers, and Digit Manipulation”
chapter in this configuration guide.
Codec complexity refers to the amount of processing power that a codec compression technique requires:
some require more processing power than others. Codec complexity affects call density, which is the
number of calls that can take place on the DSP interfaces, which can be HCMs, port adapter DSP farms,
or voice cards, depending on the type of router (in this case, the Cisco MC3810 multiservice
concentrator). The greater the codec complexity, the fewer the calls that can be handled.
Codec complexity is either medium or high. The difference between medium- and high-complexity
codecs is the amount of CPU power necessary to process the algorithm and, therefore, the number of
voice channels that can be supported by a single DSP. All medium-complexity codecs can also be run in
high-complexity mode, but fewer (usually half as many) channels will be available per DSP.
For details on the number of calls that can be handled simultaneously using each of the codec standards,
refer to the entries for the codec and codec complexity commands in the Cisco IOS Voice, Video, and
Fax Command Reference.
On a Cisco MC3810 concentrator, only a single codec complexity setting is used, even when two HCMs
are installed. The value that is specified in this task affects the choice of codecs available when the codec
dial-peer configuration command is configured. See the “Configuring Dial Plans, Dial Peers, and Digit
Manipulation” chapter in this configuration guide.
Note
On the Cisco MC3810 with high-performance compression modules, check the DSP voice channel
activity with the show voice dsp command. If any DSP voice channels are in the busy state, the codec
complexity cannot be changed. When all the DSP channels are in the idle state, changes can be made
to the codec complexity selection.
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Analog Voice Ports Configuration Task List
To configure codec complexity on the Cisco MC3810 multiservice concentrator using HCMs, use the
following commands beginning in privileged EXEC mode:
Step 1
Command
Purpose
Router# show voice dsp
Checks the DSP voice channel activity. If any DSP
voice channels are in the busy state, the codec
complexity cannot be changed.
When all the DSP channels are in the idle state,
continue to Step 2.
Step 2
Router# configure terminal
Enters global configuration mode.
Step 3
Router(config)# voice-card 0
Enters voice-card configuration mode and
specifies voice card 0.
Step 4
Router(config-voicecard)# codec complexity {high |
medium}
(For analog voice ports) Specifies codec
complexity based on the codec standard being
used. This setting restricts the codecs available in
dial peer configuration. All voice cards in a router
must use the same codec complexity setting.
The keywords are as follows:
•
high—Specifies two voice channels encoded
in any of the following formats:
G.711ulaw, G.711alaw, G.723.1(r5.3),
G.723.1 Annex A(r5.3), G.723.1(r6.3),
G.723.1 Annex A(r6.3), G.726(r16),
G.726(r24), G.726(r32), G.728, G.729, G.729
Annex B, and fax relay.
•
medium—(default) Specifies four voice
channels encoded in any of the following
formats: G.711ulaw, G.711alaw, G.726(r16),
G.726(r24), G.726(r32), G.729 Annex A,
G.729 Annex B with Annex A, and fax relay.
Note
If two HCMs are installed, this command
configures both HCMs at once.
Configuring Basic Parameters on Analog FXO, FXS, or E&M Voice Ports
This section describes commands for basic analog voice port configuration. All the data recommended
in the “Preparing to Configure Analog Voice Ports” section on page 41 should be gathered before
starting this procedure.
If configuring a Cisco MC3810 multiservice concentrator that has HCMs, codec complexity should also
be configured, following the steps in the “Configuring Codec Complexity for Analog Voice Ports on the
Cisco MC3810 with High-Performance Compression Modules” section on page 45.
Note
If you have a Cisco MC3810 multiservice concentrator or Cisco 3660 router, the compand-type
a-law command must be configured on the analog ports only. The Cisco 2660, 3620, and 3640 routers
do not require the configuration of th compand-type a-law command, however, if you request a list
of commands, the compand-type a-law command will display.
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Analog Voice Ports Configuration Task List
In addition to the basic voice port parameters described in this section, there are commands that allow
voice port configurations to be fine tuned. In most cases, the default values for fine-tuning commands
are sufficient for establishing FXO and FXS voice port configurations. E&M voice ports are more likely
to require some configuration. If it is necessary to change some of the voice port values to improve voice
quality or to match parameters on proprietary PBXs to which you are connecting, use the commands in
the current section and also in the “Fine-Tuning Analog and Digital Voice Ports” section on page 79.
After the voice-port has been configured, make sure that the ports are operational by following the steps
described in the following sections:
•
Verifying Analog and Digital Voice-Port Configurations, page 97
•
Troubleshooting Analog and Digital Voice Port Configurations, page 108
For more information on these and other voice port commands, see the Cisco IOS Voice, Video, and Fax
Command Reference.
Note
The commands, keywords, and arguments that you are able to use may differ slightly from those
presented here, based on your platform, Cisco IOS release, and configuration. When in doubt, use
Cisco IOS command help (command ?) to determine the syntax choices that are available.
To configure basic analog voice port parameters on Cisco 1750, Cisco 2600 series, Cisco 3600 series,
and Cisco MC3810 routers, use the following commands beginning in global configuration mode:
Step 1
Command
Purpose
Cisco 1750 and MC3810
Enters voice-port configuration mode.
Router(config)# voice-port slot/port
The arguments are as follows:
Cisco 2600 and 3600 series
•
slot—Specifies the number of the router slot
where the voice network module is installed
(Cisco 2600 and Cisco 3600 series routers) or
the router slot number where the analog voice
module is installed (Cisco MC3810
multiservice concentrator).
•
port—Indicates the voice port. Valid entries
are 0 or 1.
•
subunit—Specifies the location of the VIC.
Router(config)# voice-port slot/subunit/port
Note
The slash must be entered between slot
and port.
Valid entries vary by router platform; see Table 7
on page 42 or enter the show voice port
summary command for available values.
Step 2
FXO or FXS
Router(config-voiceport)# signal {loop-start |
ground-start}
Selects the access signaling type to match that of
the telephony connection you are making. The
keywords are as follows:
•
loop-start—(default) Uses a closed circuit to
indicate off-hook status; used for residential
loops.
•
ground-start—Uses ground and current
detectors; preferred for PBXs and trunks.
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Command
E&M
Router(config-voiceport)# signal {wink-start |
immediate-start | delay-dial}
Purpose
The keywords are as follows:
•
wink-start—(default) Indicates that the
calling side seizes the line, then waits for a
short off-hook wink from the called side
before proceeding.
•
immediate-start—Indicates that the calling
side seizes the line and immediately proceeds;
used for E&M tie trunk interfaces.
•
delay-dial—Indicates that the calling side
seizes the line and waits, then checks to
determine whether the called side is on-hook
before proceeding; if not, it waits until the
called side is on-hook before sending digits.
Used for E&M tie trunk interfaces.
Note
Step 3
Router(config-voiceport)# cptone locale
Configuring the signal keyword for one
voice port on a Cisco 2600 or 3600 series
router VIC changes the signal value for
both ports on the VIC.
Selects the two-letter locale for the voice call
progress tones and other locale-specific
parameters to be used on this voice port.
Cisco routers comply with the ISO 3166 locale
name standards. To see valid choices, enter a
question mark (?) following the cptone command.
The default is us.
Step 4
Router(config-voiceport)# dial-type {dtmf | pulse}
(FXO only) Specifies the dialing method for
outgoing calls.
Step 5
Router(config-voiceport)# operation {2-wire | 4-wire}
(E&M only) Specifies the number of wires used
for voice transmission at this interface (the audio
path only, not the signaling path).
Step 6
Router(config-voiceport)# type {1 | 2 | 3 | 5}
The default is 2-wire.
(E&M only) Specifies the type of E&M interface
to which this voice port is connecting. See Table 5
on page 40 for an explanation of E&M types.
The default is 1.
Step 7
Cisco 1750 Router and 2600 and 3600 Series Routers
Router(config-voiceport)# ring frequency {25 | 50}
Cisco MC3810 Multiservice Concentrator
Router(config-voiceport)# ring frequency {20 | 30}
(FXS only) Selects the ring frequency, in hertz,
used on the FXS interface. This number must
match the connected telephony equipment and
may be country-dependent. If not set properly, the
attached telephony device may not ring or it may
buzz.
The keyword default is 25 on the Cisco 1750
router, 2600 and 3600 series routers; and 20 on the
Cisco MC3810 multiservice concentrator.
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Step 8
Command
Purpose
Router(config-voiceport)# ring number number
(FXO only) Specifies the maximum number of
rings to be detected before an incoming call is
answered by the router.
The default is 1.
Step 9
Router(config-voiceport)# ring cadence {[pattern01 |
pattern02 | pattern03 | pattern04 | pattern05 |
pattern06 | pattern07 | pattern08 | pattern09 |
pattern10 | pattern11 | pattern12] | [define pulse
interval]}
(FXS only) Specifies an existing pattern for ring,
or it defines a new one. Each pattern specifies a
ring-pulse time and a ring-interval time. The
keywords and arguments are as follows:
•
pattern01 through pattern12 name pre-set
ring cadence patterns. Enter ring cadence ? to
see ring pattern explanations.
•
define pulse interval specifies a user-defined
pattern: pulse is a number (one or two digits,
from 1 to 50) specifying ring pulse (on) time
in hundreds of milliseconds, and interval is a
number (one or two digits from 1 to 50)
specifying ring interval (off) time in hundreds
of milliseconds.
The default is the pattern specified by the cptone
locale that has been configured.
Step 10
Router(config-voiceport)# description string
Attaches a text string to the configuration that
describes the connection for this voice port. This
description appears in various displays and is
useful for tracking the purpose or use of the voice
port. The string argument is a character string
from 1 to 255 characters in length.
The default is that there is no text string
(describing the voice port) attached to the
configuration.
Step 11
Router(config-voiceport)# no shutdown
Activates the voice port. If a voice port is not being
used, shut the voice port down with the shutdown
command.
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Analog Voice Ports Configuration Task List
Configuring Analog Telephone Connections on Cisco 803 and 804 Routers
Multiple devices (analog telephone, fax machine, or modem) can be connected to a Cisco 803 or 804
telephone port. The number of devices that can be connected depends on the ringer equivalent number
(REN) of each device that is to be connected. (The REN can usually be found on the bottom of a device.)
The REN of the router telephone port is 5, so if the REN of each device to be connected is 1, a maximum
of five devices can be connected to that particular telephone port.
These routers support touch-tone analog telephones only; they do not support rotary telephones.
To configure standard features for analog telephone connections on Cisco 803 and 804 routers, use the
following commands in global configuration mode:
Step 1
Command
Purpose
Router(config)# pots country country
Specifies the country to use for country-specific
default settings for physical characteristics. Enter
pots country ? for a list of supported countries and
the codes to enter.
A default country is not defined.
Step 2
Router(config)# pots line-type {type1 | type2 |
type3}
(Optional) Specifies the impedance of telephones,
fax machines, or modems connected to a Cisco 800
series router. The keywords are as follows:
•
type1—Specifies the resistance used for the
POTS connection, typically 600 ohms.
•
type2—Specifies the resistance used for the
POTS connection, typically 900 ohms.
•
type3—Specifies the resistance used for the
POTS connection, typically 300/400 ohms.
The default depends on the country chosen in the
pots country command.
Step 3
Router(config)# pots dialing-method {overlap |
enblock}
(Optional) Specifies how the router collects and
sends digits dialed on connected telephones, fax
machines, or modems. The keywords are as follows:
•
overlap—Tells the router to send each digit
dialed in a separate message.
•
enblock—Tells the router to collect all digits
dialed and to send the digits in one message.
The default depends on the country chosen in the
pots country command.
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Step 4
Command
Purpose
Router(config)# pots disconnect-supervision {osi |
reversal}
(Optional) Specifies how the router notifies the
connected telephones, fax machines, or modems
when the calling party has disconnect. The keywords
are as follows:
•
osi—(open switching interval) Specifies the
duration for which DC voltage applied between
tip and ring conductors of a telephone port is
removed.
•
reversal—Specifies the polarity reversal of the
tip and ring conductors of a telephone port.
The default depends on the country chosen in the
pots country command.
Step 5
Router(config)# pots encoding {alaw | ulaw}
(Optional) Specifies the pulse code modulation
(PCM) encoding scheme for telephones, fax
machines, or modems connected to a Cisco 800
series router. The keywords are as follows:
•
alaw—Specifies the ITU-T PCM encoding
scheme used to represent analog voice samples
as digital values.
•
ulaw—Specifies the North American PCM
encoding scheme used to represent analog voice
samples as digital values.
The default depends on the country chosen in the
pots country command.
Step 6
Step 7
Router(config)# pots tone-source {local | remote}
Router(config)# pots ringing-freq {20Hz | 25Hz |
50Hz}
(Optional) Specifies the source of dial, ringback,
and busy tones for telephones, fax machines, or
modems connected to a Cisco 800 series router. The
keywords are as follows:
•
local—(default) Specifies that the router
supplies the tones.
•
remote—Specifies that the telephone switch
supplies the tones.
(Optional) Specifies the frequency at which
telephones, fax machines, or modems connected to a
Cisco 800 series router ring. The keywords are as
follows:
•
20Hz—Indicates that connected devices ring at
20 Hz.
•
25Hz—Indicates that connected devices ring at
25 Hz.
•
50Hz—Indicates that connected devices ring at
50 Hz.
The default depends on the country chosen in the
pots country command.
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Step 8
Command
Purpose
Router(config)# pots disconnect-time interval
(Optional) Specifies the interval at which the
disconnect method is applied if connected
telephones, fax machines, or modems fail to detect
that a calling party has disconnected. The interval
argument is the number of milliseconds of the
interval and ranges from 50 to 2000.
The default depends on the country chosen in the
pots country command.
Step 9
Router(config)# pots silence-time seconds
(Optional) Specifies the interval of silence after a
calling party disconnects. The seconds argument is
the number of seconds of the interval and ranges
from 0 to 10.
The default depends on the country chosen in the
pots country command.
Step 10
Router(config)# pots distinctive-ring-guard-time
milliseconds
(Optional) Specifies the delay after which a
telephone port can be rung after a previous call is
disconnected. The milliseconds argument is the
number of milliseconds of the delay and ranges from
0 to 1000.
The default depends on the country chosen in the
pots country command.
Verifying Analog Telephone Connections on Cisco 803 and 804 Routers
After configuring analog telephone connections, perform the following steps to verify proper operation:
Step 1
Pick up the handset of an attached telephony device and check for a dial tone.
Step 2
Review the configuration using the show pots status command, which displays settings of physical
characteristics and other information on telephone interfaces.
Router# show pots status
POTS Global Configuration:
Country: United States
Dialing Method: Overlap, Tone Source: Remote, CallerId Support: YES
Line Type: 600 ohm, PCM Encoding: u-law, Disc Type: OSI,
Ringing Frequency: 20Hz, Distinctive Ring Guard timer: 0 msec
Disconnect timer: 1000 msec, Disconnect Silence timer: 5 sec
TX Gain: 6dB, RX Loss: -6dB,
Filter Mask: 6F
Adaptive Cntrl Mask: 0
POTS PORT: 1
Hook Switch Finite State Machine:
State: On Hook, Event: 0
Hook Switch Register: 10, Suspend Poll: 0
CODEC Finite State Machine
State: Idle, Event: 0
Connection: None, Call Type: Two Party, Direction: Rx only
Line Type: 600 ohm, PCM Encoding: u-law, Disc Type: OSI,
Ringing Frequency: 20Hz, Distinctive Ring Guard timer: 0 msec
Disconnect timer: 1000 msec, Disconnect Silence timer: 5 sec
TX Gain: 6dB, RX Loss: -6dB,
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Filter Mask: 6F
Adaptive Cntrl Mask: 0
CODEC Registers:
SPI Addr: 2, DSLAC Revision: 4
SLIC Cmd: 0D, TX TS: 00, RX TS: 00
Op Fn: 6F, Op Fn2: 00, Op Cond: 00
AISN: 6D, ELT: B5, EPG: 32 52 00 00
SLIC Pin Direction: 1F
CODEC Coefficients:
GX: A0 00
GR: 3A A1
Z: EA 23 2A 35 A5 9F C2 AD 3A AE 22 46 C2 F0
B: 29 FA 8F 2A CB A9 23 92 2B 49 F5 37 1D 01
X: AB 40 3B 9F A8 7E 22 97 36 A6 2A AE
R: 01 11 01 90 01 90 01 90 01 90 01 90
GZ: 60
ADAPT B: 91 B2 8F 62 31
CSM Finite State Machine:
Call 0 - State: idle, Call Id: 0x0
Active: no
Call 1 - State: idle, Call Id: 0x0
Active: no
Call 2 - State: idle, Call Id: 0x0
Active: no
POTS PORT: 2
Hook Switch Finite State Machine:
State: On Hook, Event: 0
Hook Switch Register: 20, Suspend Poll: 0
CODEC Finite State Machine:
State: Idle, Event: 0
Connection: None, Call Type: Two Party, Direction: Rx only
Line Type: 600 ohm, PCM Encoding: u-law, Disc Type: OSI,
Ringing Frequency: 20Hz, Distinctive Ring Guard timer: 0 mse
Disconnect timer: 1000msec,Disconnect Silence timer: 5 sec
TX Gain: 6dB, RX Loss: -6dB,
Filter Mask: 6F
Adaptive Cntrl Mask: 0
CODEC Registers:
SPI Addr: 3, DSLAC Revision: 4
SLIC Cmd: 0D, TX TS: 00, RX TS: 00
Op Fn: 6F, Op Fn2: 00, Op Cond: 00
AISN: 6D, ELT: B5, EPG: 32 52 00 00
SLIC Pin Direction: 1F
CODEC Coefficients:
GX: A0 00
GR: 3A A1
Z: EA 23 2A 35 A5 9F C2 AD 3A AE 22 46 C2 F0
B: 29 FA 8F 2A CB A9 23 92 2B 49 F5 37 1D 01
X: AB 40 3B 9F A8 7E 22 97 36 A6 2A AE
R: 01 11 01 90 01 90 01 90 01 90 01 90
GZ: 60
ADAPT B: 91 B2 8F 62 31
CSM Finite State Machine:
Call 0 - State: idle, Call Id: 0x0
Active: no
Call 1 - State: idle, Call Id: 0x0
Active: no
Call 2 - State: idle, Call Id: 0x0
Active: no
Time Slot Control: 0
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Configuring Digital Voice Ports
Troubleshooting Tip for Cisco 803 and 804 Routers
Check to ensure that all cables are securely connected.
Configuring Digital Voice Ports
The digital voice port commands discussed in this section configure channelized T1 or E1 connections;
for information on ISDN connections, see “Configuring ISDN Interfaces for Voice” in this configuration
guide.
The T1 or E1 lines that connect a telephony network to the digital voice ports on a router or access server
contain channels for voice calls; a T1 line contains 24 full-duplex channels or timeslots, and an E1 line
contains 30. The signal on each channel is transmitted at 64 kbps, a standard known as digital signal 0
(DS0); the channels are known as DS0 channels. The ds0-group command creates a logical voice port
(a DS0 group) from some or all of the DS0 channels, which allows you to address those channels easily,
as a group, in voice-port configuration commands.
Digital voice ports are found at the intersection of a packet voice network and a digital, circuit-switched
telephone network. The digital voice port interfaces that connect the router or access server to T1 or E1 lines
pass voice data and signaling between the packet network and the circuit-switched network.
Signaling is the exchange of information about calls and connections between two ends of a
communication path. For instance, signaling communicates to the call’s end points whether a line is idle
or busy, whether a device is on-hook or off-hook, and whether a connection is being attempted. An end
point can be a CO switch, a PBX, a telephony device such as a telephone or fax machine, or a
voice-equipped router acting as a gateway. There are two aspects to consider about signaling on digital
lines: one aspect is the actual information about line and device states that is transmitted, and the second
aspect is the method used to transmit the information on the digital lines.
The actual information about line and device states is communicated over digital lines using signaling
methods that emulate the methods used in analog circuit-switched networks: FXS, FXO, and E&M.
The method used to transmit the information describes the way that the emulated analog signaling is
transmitted over digital lines, which may be common-channel signaling (CCS) or channel-associated
signaling (CAS). CCS sends signaling information down a dedicated channel and CAS takes place
within the voice channel itself. This chapter describes CAS signaling, which is sometimes called
robbed-bit signaling because user bandwidth is robbed by the network for signaling. A bit is taken from
every sixth frame of voice data to communicate on- or off-hook status, wink, ground start, dialed digits,
and other information about the call.
In addition to setting up and tearing down calls, CAS provides the receipt and capture of dialed number
identification (DNIS) and automatic number identification (ANI) information, which are used to support
authentication and other functions. The main disadvantage of CAS signaling is its use of user bandwidth
to perform these signaling functions.
For signaling to pass between the packet network and the circuit-switched network, both networks must
use the same type of signaling. The voice ports on Cisco routers and access servers can be configured to
match the signaling of most COs and PBXs, as explained in this chapter.
This section discusses the following topics:
•
Prerequisites for Configuring Digital Voice Ports, page 55
•
Preparing Information to Configure Digital Voice Ports, page 56
•
Platform-Specific Digital Voice Hardware, page 58
•
Configuring Basic Parameters on Digital T1/E1 Voice Ports, page 61
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Configuring Digital Voice Ports
Prerequisites for Configuring Digital Voice Ports
Digital T1 or E1 packet voice capability requires specific service, software, and hardware:
•
Obtain T1 or E1 service from the service provider or from your PBX.
•
Create your company’s dial plan.
•
Establish a working telephony network based on your company’s dial plan.
•
Establish a connection to the network LAN or WAN.
•
Set up a working IP and Frame Relay or ATM network. For more information about configuring IP,
refer to the Cisco IOS IP Configuration Guide, Release 12.2.
•
Install appropriate voice processing and voice interface hardware on the router. See the
“Platform-Specific Digital Voice Hardware” section on page 58.
•
(Cisco 2600 and 3600 series routers) For digital T1 packet voice trunk network modules, install
Cisco IOS Release 12.0(5)XK, 12.0(7)T, 12.2(1), or a later release. The minimum DRAM memory
requirements are as follows:
– 32 MB, with one or two T1 lines
– 48 MB, with three or four T1 lines
– 64 MB, with five to ten T1 lines
– 128 MB, with more than ten T1 lines
The memory required for high-volume applications may be greater than that listed. Support for
digital T1 packet voice trunk network modules is included in Plus feature sets. The IP Plus feature
set requires 8 MB of Flash memory; other Plus feature sets require 16 MB.
•
(Cisco 2600 and 3600 series routers) For digital E1 packet voice trunk network modules, install
Cisco IOS Release 12.1(2)T, 12.2(1), or a later release. The minimum DRAM memory requirements
are:
– 48 MB, with one or two E1s
– 64 MB, with three to eight E1s
– 128 MB, with 9 to 12 E1s
For high-volume applications, the memory required may be greater than these minimum values.
Support for digital E1 packet voice trunk network modules is included in Plus feature sets. The IP
Plus feature set requires 16 MB of Flash memory.
•
(Cisco MC3810 concentrators) HCMs require Cisco IOS Release 12.0(7)XK or 12.1(2)T, 12.2(1),
or a later release.
•
(Cisco 7200 and 7500 series routers) For digital T1/E1 voice port adapters, install Cisco IOS Release
12.0(5)XE, 12.0(7)T, 12.2(1), or a later release. The minimum DRAM memory requirement to
support T1/E1 high-capacity digital voice port adapters is 64 MB.
The memory required for high-volume applications may be greater than that listed. Support for T1/E1
high-capacity digital voice port adapters is included in Plus feature sets. The IP Plus feature set requires
16 MB of Flash memory.
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Preparing Information to Configure Digital Voice Ports
Gather the following information about the telephony network connection that the voice port will be
making:
•
Line interface: T1 or E1
•
Signaling interface: FXO, FXS, or E&M. If the interfaces are Primary Rate Interface (PRI) or BRI,
see the “Configuring ISDN Interfaces for Voice” chapter in this configuration guide and Cisco IOS
Terminal Services Configuration Guide.
•
Line coding: AMI or B8ZS for T1, and AMI or HDB3 for E1
•
Framing format: SF (D4) or ESF for T1, and CRC4 or no-CRC4 for E1
•
Number of channels
Table 8 describes voice-port hardware configurations for various platforms. After the controllers have
been configured, the show voice port summary command can also be used to determine available voice
port numbers. If the show voice port command and a specific port number is entered, the default
voice-port configuration for that port displays.
Table 8
Digital Voice Slot/Port Designations
Router Platform
Voice Hardware
Slot Number
Port Number
Cisco 2600 series
Digital T1/E1 Packet Voice
Trunk Network Module
(NM-HDV with VWIC-1MFT
or VWIC-2MFT)
slot is the router
location of the voice
module.
port is the VWIC
location in the
network module.
1
0 to 1
slot is the router
location of the voice
module.
port is the VWIC
location in the
network module.
One network module can be
installed in a Cisco 2600 series
router.
Cisco 3600 series
Digital T1/E1 Packet Voice
Trunk Network Module
(NM-HDV with VWIC-1MFT
or VWIC-2MFT)
3620: 0 to 1
One network module can be
3640: 0 to 3
installed in a Cisco 3620
3660: 0 to 5
router. A Cisco 3640 router
can support three modules, and
as many as six can be installed
in a Cisco 3660 router.
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Table 8
Digital Voice Slot/Port Designations (continued)
Router Platform
Cisco MC3810
Voice Hardware
•
Digital voice module
(DVM)
•
Voice compression
module (VCM3 or
VCM6)
Slot Number
Port Number
1
—
or
•
High-compression
module (HCM2 or
HCM6)
VCM3 and VCM6 do not
support codec complexity
options.
Cisco AS5300
Cisco AS5800
Cisco 7200 series
One Octal T1/E1 feature card —
(eight ports) or one Quad
T1/E1 feature card (four ports)
and one or two VFCs for voice
and fax features.
controller is :
Up to four 12-port T1/E1 trunk shelf is 1
cards and up to eight VFCs
slot is 0 to 5
0 to 11
Octal: 0 to 7
Quad: 0 to 3
Interface port: 0 to 1
Two-port T1/E1 enhanced Port adapter slot:
digital voice port adapters from 1 to 4, or from 1
to 6
• PA-VXC (high-capacity)
•
•
PA-VXB (moderate
capacity)
Port adapter slot 0 is reserved
for the Fast Ethernet port on
the I/O controller (if present).
Cisco 7500 series
PA-VXB and PA-VXC on a
VIP2 or VIP4 in Cisco 7500
series routers
If the VIP is inserted in
interface processor slot 3 and
port adapter slot 0, then the
addresses of the PA-VXB or
PA-VXC are 3/0/0 or 3/0/1
(interface processor slot 3,
port adapter slot 0, and
interfaces 0 and 1).
Port adapter slot:
Interface processor
slot: 0 to 12 (depends always 0 or 1
on the number of slots
Interface port: 0 or 1
in the router)
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The following is show voice port summary sample output for a Cisco MC3810 multiservice
concentrator:
Router# show voice port summary
IN
PORT
======
0:17
0:18
0:19
0:20
0:21
0:22
0:23
OUT
CH SIG-TYPE
== ==========
18 fxo-ls
19 fxo-ls
20 fxo-ls
21 fxo-ls
22 fxo-ls
23 fxo-ls
24 e&m-imd
ADMIN
=====
down
up
up
up
up
up
up
OPER
====
down
dorm
dorm
dorm
dorm
dorm
dorm
STATUS
========
idle
idle
idle
idle
idle
idle
idle
STATUS
========
on-hook
on-hook
on-hook
on-hook
on-hook
on-hook
idle
EC
==
y
y
y
y
y
y
y
Platform-Specific Digital Voice Hardware
This section briefly describes digital voice hardware on the following platforms:
Note
•
Cisco 2600 series and Cisco 3600 series routers
•
Cisco MC3810 multiservice concentrator
•
Cisco AS5300 universal access server
•
Cisco AS5800 universal access server
•
Cisco 7200 series and Cisco 7500 series routers
For current information about supported hardware, see the release notes for the platform and
Cisco IOS release you are using.
Cisco 2600 Series and Cisco 3600 Series Routers
Digital voice hardware on Cisco 2600 series and Cisco 3600 series modular access routers includes the
high-density voice (HDV) network module and the multiflex trunk (MFT) voice/WAN interface card
(VWIC). When an HDV is used in conjunction with an MFT and packet voice DSP modules (PVDMs),
the HDV module is also called a digital packet voice trunk network module. The digital T1 or E1 packet
voice trunk network module supports T1 or E1 applications, including fractional use. The T1 version
integrates a fully managed data service unit/channel service unit (DSU/CSU), and the E1 version
includes a fully managed DSU. The digital T1 or E1 packet voice trunk network module provides
per-channel T1 or E1 data rates of 64 or 56 kbps for WAN services (Frame Relay or leased line).
Digital T1 or E1 packet voice trunk network modules for Cisco 2600 and 3600 series routers allow
enterprises or service providers, using the voice-equipped routers as customer premise equipment (CPE),
to deploy digital voice and fax relay. These network modules receive constant bit-rate telephony
information over T1 or E1 interfaces and convert that information to a compressed format so that it can
be sent over a packet network. The digital T1 or E1 packet voice trunk network modules can connect
either to a PBX (or similar telephony device) or to a CO to provide PSTN connectivity. One digital T1
or E1 packet voice trunk network module can be installed in a Cisco 2600 series router or in a Cisco 3620
router. A Cisco 3640 router can support three network modules, and a Cisco 3660 router can support up
to six network modules.
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The MFT VWICs that are used in the packet voice trunk network modules are available in one- and
two-port configurations for T1 and for E1, and in two-port configurations with drop-and-insert capability
for T1 and E1. MFTs support the following kinds of traffic:
•
Data. As WICs for T1 or E1 applications, including fractional data line use, the T1 version includes
a fully managed DSU/CSU, and the E1 version includes a fully managed DSU.
•
Packet voice. As VWICs included with the digital T1 or E1 packet voice trunk network module to
provide connections to PBXs and COs, the MFTs enable packet voice applications.
•
Multiplexed voice and data. Some two-port T1 or E1 VWICs can provide drop-and-insert
multiplexing services with integrated DSU/CSUs. For example, when used with a digital T1 packet
voice trunk network module, drop-and-insert allows 64-kbps DS0 channels to be taken from one T1
and digitally cross-connected to 64-kbps DS0 channels on another T1. Drop and insert, sometimes
called TDM cross-connect, uses circuit switching rather than the DSPs that VoIP technology
employs. (Drop-and-insert is described in the “Configuring Trunk Connections and Trunk
Conditioning Features” chapter in this configuration guide.)
The digital T1 or E1 packet voice trunk network module contains five 72-pin Single In-line Memory
Module (SIMM) sockets or banks, numbered 0 through 4, for PVDMs. Each socket can be filled with a
single 72-pin PVDM, and there must be at least one packet voice data module (PVDM-12) in the network
module to process voice calls. Each PVDM holds three digital signal processors (DSPs), so with five
PVDM slots populated, a total of 15 DSPs are provided. High-complexity codecs support two
simultaneous calls on each DSP, and medium-complexity codecs support four calls on each DSP. A
digital T1 or E1 packet voice trunk network module can support the following numbers of channels:
•
When the digital T1 or E1 packet voice trunk network module is configured for high-complexity
codec mode, up to six voice or fax calls can be completed per PVDM-12, using the following codecs:
G.711, G.726, G.729, G729 Annex A (E1), G.729 Annex B, G.723.1, G723.1 Annex A (T1), G.728,
and fax relay.
•
When the digital T1 or E1 packet voice trunk network module is configured for medium-complexity
codec mode, up to 12 voice or fax calls can be completed per PVDM-12, using the following codecs:
G.711, G.726, G.729 Annex A, G.729 Annex B with Annex A, and fax relay.
For more information, refer to the following publications:
•
Cisco 2600 Series Hardware Installation Guide
•
Cisco 3600 Series Hardware Installation Guide
•
Cisco Network Module Hardware Installation Guide
•
Cisco IOS Release 12.0(7)T online document Configuring 1- and 2-Port T1/E1 Multiflex Voice/WAN
Interface Cards on Cisco 2600 and 3600 Series Routers
Cisco MC3810 Multiservice Concentrator
To support a T1 or E1 digital voice interface, the Cisco MC3810 multiservice concentrator must be
equipped with a digital voice interface card (DVM). The DVM interfaces with a digital PBX, channel
bank, or video codec. It supports up to 24 channels of compressed digital voice at 8 kbps, or it can
cross-connect channelized data from user equipment directly onto the router’s trunk port for connection
to a carrier network.
The DVM is available with a balanced interface using an RJ-48 connector or with an unbalanced
interface using Bayonet-Neill-Concelman (BNC) connectors.
Optional HCMs can replace standard VCMs to operate according to the voice compression coding
algorithm (codec) specified when the Cisco MC3810 multiservice concentrator is configured. The
HCM2 provides 4 voice channels at high codec complexity and 8 channels at medium complexity. The
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HCM6 provides 12 voice channels at high complexity and 24 channels at medium complexity. You can
install one or two HCMs in a Cisco MC3810, but an HCM can not be combined with a VCM in the same
chassis.
For more information, refer to the following publications:
•
Cisco MC3810 Multiservice Concentrator Hardware Installation Guide
•
Overview of the Cisco MC3810 Series
•
Configuring Cisco MC3810 Series Concentrators to Use High-Performance Compression Modules
Cisco AS5300 Universal Access Server
The Cisco AS5300 Universal Access Server includes three expansion slots. One slot is for either an Octal
T1/E1/PRI feature card (eight ports) or a Quad T1/E1/PRI feature card (four ports), and the other two
can be used for voice/fax or modem feature cards. Because a single voice/fax feature card (VFC) can
support up to 48 (T1) or 60 (E1) voice calls, the Cisco AS5300/Voice Gateway system can support a total
of 96 or 120 simultaneous voice calls. The use of VFCs requires Cisco IOS release 12.0.2XH or later.
Cisco AS5300 VFCs are coprocessor cards, each with a powerful reduced instruction set computing
(RISC) engine and dedicated, high-performance DSPs to ensure predictable, real-time voice processing.
The design couples this coprocessor with direct access to the Cisco AS5300 routing engine for
streamlined packet forwarding.
For more information, refer to the following publications:
•
Cisco AS5300 Chassis Installation Guide
•
Cisco AS5300 Module Installation Guide
Cisco AS5800 Universal Access Server
The Cisco AS5800 Universal Access Server consists of two primary system components: the Cisco 5814
dial shelf (DS), which holds channelized trunk cards and connects to the PSTN, and the Cisco 7206
router shelf (RS), which holds port adapters and connects to the IP backbone.
The dial shelf acts as the access concentrator by accepting and consolidating all types of remote traffic,
including voice, dial-in analog and digital ISDN data, and industry-standard WAN and remote
connection types. The dial shelf also contains controller cards voice feature cards, modem feature cards,
trunk cards, and dial shelf interconnect cards.
One or two dial shelf controllers (DSCs) provide clock and power control to the dial shelf cards. Each
DSC contains a block of logic that is referred to as the common logic and system clocks. This block of
logic can use a variety of sources to generate the system timing, including an E1 or T1/T3 input signal
from the BNC connector on the DSC's front panel. The configuration commands for the master clock
specify the various clock sources and a priority for each source (see the “Clock Sources on Digital T1/E1
Voice Ports” section on page 66).
The Cisco AS5800 voice feature card is a multi-DSP coprocessing board and software package that adds
VoIP capabilities to the Cisco AS5800 platform. The Cisco AS5800 voice feature card, when used with
other cards such as LAN/WAN and modem cards, provides a gateway for up to 192 packetized voice/fax
calls and 360 data calls per card. A Cisco AS5800 can support up to 1,344 voice calls in split-dial-shelf
configuration with two 7206VXR router shelves.
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For more information, refer to the following publications:
•
•
Cisco AS5800 Universal Access Server Operation, Administration, Maintenance, and Provisioning
Guide
Cisco AS5800 Access Server Hardware Installation Guide
Cisco 7200 and Cisco 7500 Series Routers
Cisco 7200 and Cisco 7500 series routers support multimedia routing and bridging with a wide variety
of protocols and media types. The Cisco 7000 family versatile interface processor (VIP) is based on a
RISC engine optimized for I/O functions. To this engine are attached one or two port adapters or
daughter boards, which provide the media-specific interfaces to the network. The network interfaces
provide connections between the routers’ peripheral component interconnect (PCI) buses and external
networks. Port adapters can be placed in any available port adapter slot, in any desired combination.
T1/E1 high-capacity digital voice port adapters for Cisco 7200 and Cisco 7500 series routers allow
enterprises or service providers, using the equipped routers as customer premise equipment, to deploy
digital voice and fax relay. These port adapters receive constant bit-rate telephony information over
T1/E1 interfaces and can convert that information to a compressed format for transmission as voice over
IP (VoIP). Two types of digital voice port adapters are supported on Cisco 7200 and Cisco 7500 series
routers: two-port high-capacity (up to 48 or 120 channels of compressed voice, depending on codec
choice), and two-port moderate capacity (up to 24 or 48 channels of compressed voice). These
single-width port adapters incorporate two universal ports configurable for either T1 or E1 connection,
for use with high-performance digital signal processors (DSPs). Integrated CSU/DSUs, echo
cancellation, and DS0 drop-and-insert functionality eliminate the need for external line termination
devices and multiplexers.
For more information, refer to the following publications:
Note
•
Cisco 7200 VXR Installation and Configuration Guide
•
Cisco 7500 Series Installation and Configuration Guide
•
Two-Port T1/E1 Moderate-Capacity and High-Capacity Digital Voice Port Adapter Installation and
Configuration
For current information about supported hardware, see the release notes for the platform and
Cisco IOS release being used.
Configuring Basic Parameters on Digital T1/E1 Voice Ports
This section describes commands for basic digital voice port configuration. Make sure you have all the
data recommended in the “Preparing Information to Configure Digital Voice Ports” section on page 56
before starting this procedure.
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The basic steps for configuring digital voice ports are described in the next three sections. They are
grouped by the configuration mode from which they are executed, as follows:
•
Configuring Codec Complexity for Digital T1/E1 Voice Ports, page 62
Codec complexity refers to the amount of processing power assigned to codec processing on a voice
port. On most router platforms that support codec complexity, codec complexity is selected in voice
card configuration mode, although it is selected in DSP interface mode on the Cisco 7200 and
7500 series. The value configured for codec complexity establishes the choice of codecs that are
available on the dial peers. See the Configuring Dial Plans, Dial Peers, and Digit Manipulation
chapter in this configuration guide for more information about configuring dial peers.
•
Configuring Controller Settings for Digital T1/E1 Voice Ports, page 65
Specific line characteristics must be configured to match those of the PSTN line that is being
connected to the voice port. These are typically configured in controller configuration mode.
•
Configuring Basic Voice Port Parameters for Digital T1/E1 Voice Ports, page 76
Voice port configuration mode allows many of the basic voice call attributes to be configured to
match those of the PSTN or PBX connection being made on this voice port.
In addition to the basic voice port parameters, there are additional commands that allow for the finetuning of the voice port configurations or for configuration of optional features. In most cases, the
default values for these commands are sufficient for establishing voice port configurations. If it is
necessary to change some of these parameters to improve voice quality or to match parameters in
proprietary PBXs to which you are connecting, use the commands in the “Fine-Tuning Analog and
Digital Voice Ports” section on page 79.
After voice port configuration, make sure the ports are operational by following the steps described in
these sections:
•
Verifying Analog and Digital Voice-Port Configurations, page 97
•
Troubleshooting Analog and Digital Voice Port Configurations, page 108
For more information on voice port commands, refer to the Cisco IOS Voice, Video, and Fax Command
Reference.
Configuring Codec Complexity for Digital T1/E1 Voice Ports
On the Cisco 2600, 3600, 7200, and 7500 routers, codec complexity can be configured separately for
each T1/E1 digital packet voice trunk network module or port adapter. On a Cisco MC3810 multiservice
concentrator, only a single codec complexity setting is used, even when two HCMs are installed. The
value specified in this task affects the choice of codecs available when the codec dial-peer configuration
command is configured.
For details on the number of calls that can be handled simultaneously using each of the codec standards,
refer to the entries for codec and codec complexity in the Cisco IOS Voice, Video, and Fax Command
Reference and to platform-specific product literature.
For more information on codec complexity, see the “Configuring Codec Complexity for Analog Voice
Ports on the Cisco MC3810 with High-Performance Compression Modules” section on page 45.
Two configuration task tables are shown below: one for the Cisco 2600 and 3600 series routers and the
Cisco MC3810 concentrator, which use voice card configuration mode, and the second for the
Cisco 7200 and 7500 series routers, which use DSP interface configuration mode.
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Cisco 2600 and 3600 Series and Cisco MC3810
This procedure applies to voice ports on digital packet voice trunk network modules on
Cisco 2600 series and Cisco 3600 series routers, and to voice ports on HCMs on Cisco MC3810
multiservice concentrators.
Note
On Cisco 2600 and 3600 series routers with digital T1/E1 packet voice trunk network modules, codec
complexity cannot be configured if DS0 groups are configured. Use the no ds0-group command to
remove DS0 groups before configuring codec complexity.
Note
On the Cisco MC3810 multiservice concentrator with high compression modules, check the DSP
voice channel activity with the show voice dsp command. If any DSP voice channels are in the busy
state, you cannot change the codec complexity. When all of the DSP channels are in the idle state,
you can make changes to the codec complexity selection.
To configure codec complexity, use the following commands beginning in privileged EXEC mode:
Step 1
Command
Purpose
Router# show voice dsp
Checks the DSP voice channel activity. If any DSP voice
channels are in the busy state, codec complexity cannot be
changed.
When all of the DSP channels are in the idle state, continue to
Step 2.
Step 2
Router# configure terminal
Enters global configuration mode.
Step 3
Router(config)# voice-card slot
Enters voice card configuration mode for the card or cards in
the slot specified.
For the Cisco 2600 and 3600 series routers, the slot argument
ranges from 0 to 5. For the Cisco MC3810 multiservice
concentrator, slot must be 0.
Step 4
Router(config-voicecard)# codec complexity
{high | med}
Specifies codec complexity based on the codec standard being
used. This setting restricts the codecs available in dial peer
configuration. All voice cards in a router must use the same
codec complexity setting. The keywords are as follows:
•
high—(Optional) Specifies up to six voice or fax calls
completed per PVDM-12, using the following codecs:
G.711, G.726, G.729, G.729 Annex B, G.723.1, G.723.1
Annex A, G.728, and fax relay.
•
med—(Optional) Supports up to 12 voice or fax calls
completed per PVDM-12, using the following codecs:
G.711, G.726, G.729 Annex A, G.729 Annex B with
Annex A, and fax relay. The default is med.
Note
On the Cisco MC3810 multiservice concentrator, this
command is valid only with one or more HCMs
installed, and voice card 0 must be specified. If two
HCMs are installed, this command configures both
HCMs at once.
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Cisco AS5300 Universal Access Server
Codec support on the Cisco AS5300 universal access server is determined by the capability list on the
voice feature card, which defines the set of codecs that can be negotiated for a voice call. The capability
list is created and populated when VCWare is unbundled and DSPWare is added to VFC Flash memory.
The capability list does not indicate codec preference; it simply reports the codecs that are available. The
session application decides which codec to use. Codec support is configured on dial peers rather than on
voice ports; see the “Configuring Dial Plans, Dial Peers, and Digit Manipulation” chapter in this
configuration guide.
Cisco AS5800 Universal Access Server
Selection of codec support on Cisco AS5800 access servers is made during dial peer configuration. See
the “Configuring Dial Plans, Dial Peers, and Digit Manipulation” chapter in this configuration guide.
Cisco 7200 Series and Cisco 7500 Series Routers
On Cisco 7200 series and Cisco 7500 series routers, codec complexity is configured on the DSP
interface.
Note
Check the DSP voice channel activity using the show interfaces dspfarm command. If any DSP
voice channels are in the busy state, codec complexity cannot be changed. When all of the DSP
channels are in the idle state, changes can be made to the codec complexity selection.
To configure the DSP interface, use the following commands beginning in privileged EXEC mode:
Step 1
Command
Purpose
Router# show interfaces dspfarm
Displays the DSP voice channel activity. If any
DSP voice channels are in the busy state, codec
complexity cannot be changed.
When all of the DSP channels are in the idle state,
continue to Step 2.
Step 2
Router# configure terminal
Enters global configuration mode.
Step 3
Cisco 7200 series
Enters DSP interface configuration mode. The
arguments are as follows:
Router(config)# dspint dspfarm slot/port
Cisco 7500 series
•
slot/port—Specifies the slot and port numbers
of the interface.
•
adapter/port—Specifies the adapter and port
numbers of the interface.
Router(config)# dspint dspfarm slot/port-adapter/port
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Step 4
Step 5
Command
Purpose
Router(config-dspfarm)# codec {high | med}
Specifies the codec complexity based on the codec
standard being used. The keyword specified for
codec affects the choice of codecs available when
the codec dial-peer configuration command is
used. The keywords are as follows:
Router(config-dspfarm)# description
•
high—Supports two voice channels encoded
in any of the following formats: G.711, G.726,
G.729, G.729 Annex B, G.723.1, G.723.1
Annex A, G.728, and fax relay.
•
med—(default) Supports up to four calls
using the following codecs: G.711, G.726,
G.729 Annex A, G.729 Annex B with Annex
A, and fax relay.
Enters a string to include descriptive text about
this DSP interface connection. This information is
displayed in the output for show commands and
does not affect the operation of the interface in any
way.
Configuring Controller Settings for Digital T1/E1 Voice Ports
The purpose of configuring controllers for digital T1/E1 voice ports is to match the configuration of the
router to the line characteristics of the telephony network connection being made so that voice and
signaling can be transferred between them and so that logical voice ports, or DS0 groups, may be
established.
Figure 16 shows how a ds0-group command gathers some of the DS0 time slots from a T1 line into a
group that becomes a single logical voice port, which can later be addressed as a single entity in voice
port configurations. Other DS0 groups for voice can be created from the remaining time slots shown in
the figure, or the time slots can be used for data or serial pass-through.
Note that all the controller commands in Figure 16 other than ds0-group apply to all the time slots in
the T1.
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Figure 16
T1 Controller Configuration on Cisco 2600 or 3600 Series Routers
Network module slot 1
VWIC slot 0
T1
V
V
Creates DS0 group, or
logical voice port, 1/0:1
by grouping 12
time slots together
controller t1 1/0
framing esf
clock source line
linecode b8zs
ds0-group 1 timeslots 1-12 type e&m-wink-start
37760
Configures T1
controller 1/0
Voice port controller configuration includes setting the parameters described in the following sections:
•
Framing Formats on Digital T1/E1 Voice Ports
•
Clock Sources on Digital T1/E1 Voice Ports
•
Line Coding on Digital T1/E1 Voice Ports
•
DS0 Groups on Digital T1/E1 Voice Ports
Another controller command that might be needed, cablelength, is discussed in the Cisco IOS Interface
Command Reference, Release 12.2.
Framing Formats on Digital T1/E1 Voice Ports
The framing format parameter describes the way that bits are robbed from specific frames to be used for
signaling purposes. The controller must be configured to use the same framing format as the line from
the PBX or CO that connects to the voice port you are configuring.
Digital T1 lines use super frame (SF) or extended super frame (ESF) framing formats. SF provides
two-state, continuous supervision signaling, in which bit values of 0 are used to represent on-hook and
bit values of 1 are used to represent off-hook. ESF robs four bits instead of two, yet has little impact on
voice quality. ESF is required for 64-kbps operation on DS0 and is recommended for Primary Rate
Interface (PRI) configurations.
E1 lines can be configured for cyclic redundancy check (CRC4) or no cyclic redundancy check, with an
optional argument for E1 lines in Australia.
Clock Sources on Digital T1/E1 Voice Ports
Digital T1/E1 interfaces use timers called clocks to ensure that voice packets are delivered and
assembled properly. All interfaces handling the same packets must be configured to use the same source
of timing so that packets are not lost or delivered late. The timing source that is configured can be
external (from the line) or internal to the router’s digital interface.
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If the timing source is internal, timing derives from the onboard phase-lock loop (PLL) chip in the digital
voice interface. If the timing source is line (external), then timing derives from the PBX or PSTN CO to
which the voice port is connected. It is generally preferable to derive timing from the PSTN because their
clocks are maintained at an extremely accurate level. This is the default setting for the clocks. When two
or more controllers are configured, one should be designated as the primary clock source; it will drive
the other controllers.
The line keyword specifies that the clock source is derived from the active line rather than from the
free-running internal clock. The following rules apply to clock sourcing on the controller ports:
•
When both ports are set to line clocking with no primary specification, port 0 is the default primary
clock source and port 1 is the default secondary clock source.
•
When both ports are set to line and one port is set as the primary clock source, the other port is by
default the backup or secondary source and is loop-timed.
•
If one port is set to clock source line or clock source line primary and the other is set to clock source
internal, the internal port recovers clock from the clock source line port if the clock source line port
is up. If it is down, then the internal port generates its own clock.
•
If both ports are set to clock source internal, there is only one clock source: internal.
This section describes the five basic timing scenarios that can occur when a digital voice port is
connected to a PBX or CO. In all the examples that follow, the PSTN (or CO) and the PBX are
interchangeable for purposes of providing or receiving clocking.
•
Single Voice Port Providing Clocking—In this scenario, the digital voice hardware is the clock
source for the connected device, as shown in Figure 17. The PLL generates the clock internally and
drives the clocking on the line. Generally, this method is useful only when connecting to a PBX, key
system, or channel bank. A Cisco VoIP gateway rarely provides clocking to the CO because CO
clocking is much more reliable. The following configuration sets up this clocking method for a
digital E1 voice port:
controller E1 1/0
framing crc4
linecoding hdb3
clock source internal
ds0-group timeslots 1-15 type e&m-wink-start
Single Voice Port Providing Clocking
E1 0
Clock
Single Voice Port Receiving Internal Clocking—In this scenario, the digital voice hardware receives
clocking from the connected device (CO telephony switch or PBX) (see Figure 18). The PLL
clocking is driven by the clock reference on the receive (Rx) side of the digital line connection.
Figure 18
Single E1 Port Receiving Clocking from the Line
E1 0
Clock
PSTN
26920
•
PBX
26919
Figure 17
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The following configuration sets up this clocking method:
controller T1 1/0
framing esf
linecoding ami
clock source line
ds0-group timeslots 1-12 type e&m-wink-start
•
Dual Voice Ports Receiving Clocking from the Line—In this scenario, the digital voice port has two
reference clocks, one from the PBX and another from the CO, as shown in Figure 19. Because the
PLL can derive clocking from only one source, this case is more complex than the two preceding
examples.
Before looking at the details, consider the following as they pertain to the clocking method:
– Looped-time clocking: The voice port takes the clock received on its Rx (receive) pair and
regenerates it on its Tx (transmit) pair. While the port receives clocking, the port is not driving
the PLL on the card but is “spoofing” (that is, fooling) the port so that the connected device has
a viable clock and does not see slips (that is, loss of data bits). PBXs are not designed to accept
slips on a T1 or E1 line, and such slips cause a PBX to drop the link into failure mode. While
in looped-time mode, the router often sees slips, but because these are controlled slips, they
usually do not force failures of the router’s voice port.
– Slips: These messages indicate that the voice port is receiving clock information that is out of
phase (out of synchronization). Because the router has only a single PLL, it can experience
controlled slips while it receives clocking from two different time sources. The router can
usually handle controlled slips because its single-PLL architecture anticipates them.
Note
Physical layer issues, such as bad cabling or faulty clocking references, can cause slips.
Eliminate these slips by addressing the physical layer or clock reference problems.
In the dual voice ports receiving clocking from the line scenario, the PLL derives clocking from the
CO and puts the voice port connected to the PBX into looped-time mode. This is usually the best
method because the CO provides an excellent clock source (and the PLL usually requires that the
CO provide that source) and a PBX usually must receive clocking from the other voice port.
Figure 19
Dual E1 Ports Receiving Clocking from the Line
Clock
PSTN
E1 1
Clock
PBX
26921
E1 0
Looped time
The following configuration sets up this clocking method:
controller E1 1/0 << description - connected to the CO
framing crc4
linecoding hdb3
clock source line primary
ds0-group timeslots 1-15 type e&m-wink-start
!
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controller E1 1/1 << description - connected to the PBX
framing crc4
linecoding hdb3
clock source line
ds0-group timeslots 1-15 type e&m-wink-start
The clock source line primary command tells the router to use this voice port to drive the PLL. All
other voice ports configured as clock source line are then put into an implicit loop-timed mode. If
the primary voice port fails or goes down, the other voice port instead receives the clock that drives
the PLL. In this configuration, port 1/1 might see controlled slips, but these should not force it down.
This method prevents the PBX from seeing slips.
Note
When terminating two T1/E1 lines on a two-port interface card, such as the VWIC-2MFT, if
both controllers are set for line clocking but the lines are not within clocking tolerance of one
another, one of the controllers is likely to experience slips. To prevent slips, ensure that the
two T1/E1 lines are within clocking tolerance of one another, even if the lines are from
different providers.
•
Dual Voice Ports (One Receives Clocking and One Provides Clocking)—In this scenario, the digital
voice hardware receives clocking for the PLL from E1 0 and uses this clock as a reference to clock
E1 1 (see Figure 20). If controller E1 0 fails, the PLL internally generates the clock reference to
drive E1 1.
Figure 20
Dual E1 ports—One Receiving and One Providing Clocking
Clock
PSTN
E1 1
Clock
PBX
26922
E1 0
The following configuration sets up this clocking method:
controller E1 1/0
framing crc4
linecoding hdb3
clock source line
ds0-group timeslots 1-15 type e&m-wink-start
!
controller E1 1/1
framing crc4
linecoding hdb3
clock source internal
ds0-group timeslots 1-15 type e&m-wink-start
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•
Dual Voice Ports (Router Provides Both Clocks)—In this scenario, the router generates the clock for
the PLL and, therefore, for both voice ports (see Figure 21).
Figure 21
Dual E1 Ports—both Clocks from the Router
Clock
PSTN
E1 1/1
Clock
PBX
26923
E1 1/0
The following configuration sets up this clocking method:
controller E1 1/0
framing crc4
linecoding hdb3
clock source internal
ds0-group timeslots 1-15 type e&m-wink-start
!
controller E1 1/1
framing esf
linecoding b8zs
clock source internal
ds0-group timeslots 1-15 type e&m-wink-start
Line Coding on Digital T1/E1 Voice Ports
Digital T1/E1 interfaces require that line encoding be configured to match that of the PBX or CO that is
being connected to the voice port. Line encoding defines the type of framing used on the line.
T1 line encoding methods include alternate mark inversion (AMI) and binary 8 zero substitution (B8ZS).
AMI is used on older T1 circuits and references signal transitions with a binary 1, or “mark.” B8ZS, a
more reliable method, is more popular and is recommended for PRI configurations as well. B8ZS
encodes a sequence of eight zeros in a unique binary sequence to detect line-coding violations.
Supported E1 line encoding methods are AMI and high-density bipolar 3 (HDB3), which is a form of
zero-suppression line coding.
DS0 Groups on Digital T1/E1 Voice Ports
For digital voice ports, a single command, ds0-group, performs the following functions:
•
Defines the T1/E1 channels for compressed voice calls.
•
Automatically creates a logical voice port.
The numbering for the logical voice port created as a result of this command is
controller:ds0-group-no, where controller is defined as the platform-specific address for a particular
controller. On a Cisco 3640 router, for example, ds0-group 1 timeslots 1-24 type e&m-wink
automatically creates the voice port 1/0:1 when issued in the configuration mode for controller 1/0.
On a Cisco MC3810 universal concentrator, when you are in the configuration mode for controller
0, the command ds0-group 1 timeslots 1-24 type e&m-wink creates logical voice port 0:1.
To map individual DS0s, define additional DS0 groups under the T1/E1 controller, specifying
different time slots. Defining additional DS0 groups also creates individual DS0 voice ports.
•
Defines the emulated analog signaling method that the router uses to connect to the PBX or PSTN.
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Most digital T1/E1 connections used for switch-to-switch (or switch-to-router) trunks are E&M
connections, but FXS and FXO connections are also supported. These are normally used to provide
emulated-OPX (Off-Premises eXtension) from a PBX to remote stations. FXO ports connect to FXS
ports. The FXO or FXS connection between the router and switch (CO or PBX) must use matching
signaling, or calls cannot connect properly. Either ground start or loop start signaling is appropriate
for these connections. Ground start provides better disconnect supervision to detect when a remote
user has hung up the telephone, but ground start is not available on all PBXs.
Digital ground start differs from digital E&M because the A and B bits do not track each other as
they do in digital E&M signaling (that is, A is not necessarily equal to B). When the CO delivers a
call, it seizes a channel (goes off-hook) by setting the A bit to 0. The CO equipment also simulates
ringing by toggling the B bit. The terminating equipment goes off-hook when it is ready to answer
the call. Digits are usually not delivered for incoming calls.
E&M connections can use one of three different signaling types to acknowledge on-hook and
off-hook states: wink start, immediate start, and delay start. E&M wink start is usually preferred,
but not all COs and PBXs can handle wink start signaling. The E&M connection between the router
and switch (CO or PBX) must match the CO or PBX E&M signaling type, or calls cannot be
connected properly.
E&M signaling is normally used for trunks. It is normally the only way that a CO switch can provide
two-way dialing with Direct Inward Dialing (DID). In all the E&M protocols, off-hook is indicated
by A=B=1 and on-hook is indicated by A=B=0 (robbed-bit signaling). If dial pulse dialing is used,
the A and B bits are pulsed to indicate the addressing digits. The are several further important
subclasses of E&M robbed-bit signaling:
– E&M Wink Start—Feature Group B
In the original wink start handshaking protocol, the terminating side responds to an off-hook
from the originating side with a short wink (transition from on-hook to off-hook and back
again). This wink tells the originating side that the terminating side is ready to receive
addressing digits. After receiving addressing digits, the terminating side then goes off-hook for
the duration of the call. The originating endpoint maintains off-hook for the duration of the call.
– E&M Wink Start—Feature Group D
In Feature Group D wink start with wink acknowledge handshaking protocol, the terminating
side responds to an off-hook from the originating side with a short wink (transition from
on-hook to off-hook and back again) just as in the original wink start. This wink tells the
originating side that the terminating side is ready to receive addressing digits. After receiving
addressing digits, the terminating side provides another wink (called an acknowledgment wink)
that tells the originating side that the terminating side has received the dialed digits. The
terminating side then goes off-hook to indicate connection. This last indication can be due to
the ultimate called endpoint’s having answered. The originating endpoint maintains an off-hook
condition for the duration of the call.
– E&M Immediate Start
In the immediate-start protocol, the originating side does not wait for a wink before sending
addressing information. After receiving addressing digits, the terminating side then goes
off-hook for the duration of the call. The originating endpoint maintains off-hook for the
duration of the call.
Note
Feature Group D is supported on Cisco AS5300 platforms, and on Cisco 2600, 3600, and 7200 series
with digital T1 packet voice trunk network modules. Feature Group D is not supported on E1 or
analog voice ports.
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To configure controller settings for digital T1/E1 voice ports, use the following commands beginning in
global configuration mode:
Step 1
Command
Purpose
Cisco 7200 and 7500 series
Defines the card as T1 or E1 and stipulates the
location.
Router(config)# card type {t1 | e1} slot
The keywords and arguments are as follows:
Step 2
•
t1 | e1—Defines the type of card.
•
slot—A value from 0 to 5.
Cisco 2600 and 3600 series, Cisco MC3810, and Cisco 7200 series
Enters controller configuration mode.
Router(config)# controller {t1 | e1} slot/port
The keywords and arguments are as follows:
Cisco AS5300
•
t1 | e1—The type of controller.
Router(config)# controller {t1 | e1} number
•
slot/port—The backplane slot number and
port number for the interface being
configured.
•
number—The network processor module
number; the range is from 0 to 2.
•
shelf/slot/port—Indicates the controller ports;
the range for port is from 0 to 11.
Cisco AS5800
Router(config)# controller {t1 | e1} shelf/slot/port
Cisco 7500 series
Router(config)# controller {t1 | e1}
slot/port-adapter/slot
Step 3
T1
Selects frame type for T1 or E1 line.
Router(config-controller)# framing {sf | esf}
The keywords and arguments are as follows:
E1
T1 lines
Router(config-controller)# framing {crc4 | no-crc4}
[australia]
•
sf—super frame
•
esf—extended super frame
E1 lines
•
crc4—Provides 4 bits of error protection.
•
no-crc4—Disables crc4.
•
australia—(Optional) Specifies the E1 frame
type used in Australia.
The default for T1 is sf.
The default for E1 is crc4.
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Step 4
Command
Purpose
Router(config-controller)# clock source {line [primary
| secondary] | internal}
Configures the clock source.
The keywords and arguments are as follows:
•
line—Specifies that the PLL on this port
derives clocking from the external source to
which the port is connected (generally the
CO).
•
primary—(Optional) Specifies that the PLL
on this port derives clocking from the external
source and puts the other port (generally
connected to the PBX) into looped-time
mode. Both ports are configured with line, but
only the port connected to the external source
is configured with primary.
•
secondary—(Optional) Indicates a backup
external source for clocking if the primary
clocking shuts down. Configure the clock
source line secondary command on the
controller that has the next-best-known
clocking.
•
internal—(Optional) Specifies that the clock
is generated from the voice port’s internal
PLL.
For more information about clock sources, see the
“Clock Sources on Digital T1/E1 Voice Ports”
section on page 66.
The default is line.
Step 5
T1 lines
Specifies the line encoding to use.
Router(config-controller)# linecode {ami | b8zs}
The keywords are as follows:
E1 lines
•
ami—Specifies the alternate mark inversion
(AMI) line code type. (T1 and E1)
•
b8zs—Specifies the binary 8 zero substitution
(B8ZS) line code type. (T1 only)
•
hdb3—Specifies the high-density bipolar 3
(HDB3) line code type. (E1 only)
Router(config-controller)# linecode {ami | hdb3}
The default for T1 is ami.
The default for E1 is hdb3.
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Step 6
Command
Purpose
Cisco 2600 and 3600 Series Routers and Cisco MC3810 Multiservice
Concentrators—T1
Defines the T1 channels for use by compressed
voice calls and the signaling method that the
router uses to connect to the PBX or CO.
Router(config-controller)# ds0-group ds0-group-no
timeslots timeslot-list type {e&m-delay-dial | e&m-fgd
| e&m-immediate-start e&m-wink-start | ext-sig |
fgd-eana | fxo-ground-start | fxo-loop-start |
fxs-ground-start | fxs-loop-start}
Note
Cisco 2600 and 3600 Series Routers and Cisco MC3810 Multservice
Concentrators—E1
Router(config-controller)# ds0-group ds0-group-no
timeslots timeslot-list type {e&m-delay-dial |
e&m-immediate-start | e&m-melcas-delay |
e&m-melcas-immed | e&m-melcas-wink | e&m-wink-start |
ext-sig | fgd-eana | fxo-ground-start | fxo-loop-start
| fxo-melcas | fxs-ground-start | fxs-loop-start |
fxs-melcas | r2-analog | r2-digital | r2-pulse}
This step shows the basic syntax and
signaling types available with the
ds0-group command. For the complete
syntax, see the Cisco IOS Voice, Video,
and Fax Command Reference,
Release 12.2.
The keywords and arguments are as follows:
•
ds0-group-no—Identifies the DS0 group
(number from 0 to 23, for T1, or from 0 to 30,
for E1).
•
timeslots timeslot-list—Specifies the single
time slot number, single range of numbers, or
multiple ranges of numbers separated by
commas. For T1/E1, allowable values are
from 1 to 24. Examples are as follows:
Cisco AS5300 Universal Access Servers—T1
Router(config-controller)# ds0-group ds0-group-no
timeslots timeslot-list [service {data | fax | voice}]
[type {e&m-fgb | e&m-fgd | e&m-immediate-start |
fxs-ground-start | fxs-loop-start | fgd-eana | fgd-os
| r1-itu | sas-ground-start | sas-loop-start | none}]
– 2, 3-5
– 1, 7, 9
Cisco AS5300 Universal Access Servers—E1
Router(config-controller)# ds0-group ds0-group-no
timeslots timeslot-list type {none | p7 | r2-analog |
r2-digital | r2-lsv181-digital | r2-pulse}
Cisco AS5800 Universal Access Servers—T1
Router(config-controller)# ds0-group ds0-group-no
timeslots timeslot-list type {e&m-fgb | e&m-fgd |
e&m-immediate-start | fxs-ground-start |
fxs-loop-start | fgd-eana | r1-itu | r1-modified |
r1-turkey | sas-ground-start | sas-loop-start | none}
Cisco AS5800 Universal Access Servers E1 Voice Ports
Router(config-controller)# ds0-group ds0-group-no
timeslots timeslot-list type {e&m-fgb | e&m-fgd |
e&m-immediate-start | fxs-ground-start |
fxs-loop-start | p7 | r2-analog | r2-digital |
r2-pulse | sas-ground-start | sas-loop-start | none}
– 1-12
•
service—Indicates the type of calls to be
handled by this DS0 group—data, fax, or
voice).
•
type—Refers to the signaling type of the
telephony connection being made. Types
include the following:
– e&m-delay-dial—Specifies the
originating endpoint that sends an
off-hook signal and waits for the off-hook
signal followed by an on-hook signal
from the destination.
– e&m-fgb—E & M Type II Feature
Group B.
– e&m-fgd—E & M Type II Feature
Cisco 7200 and 7500 Series Series Routers T1 and E1 Voice Ports
Router(config-controller)# ds0-group ds0-group-no
timeslots timeslot-list type {e&m-delay |
e&m-immediate | e&m-wink | fxs-ground-start |
fxs-loop-start | fxo-ground-start | fxo-loop-start}
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Command
Purpose
– e&m-immediate-start—E & M
Immediate Start.
– e&m-melcas-delay—E&M Mercury
Exchange Limited Channel Associated
Signaling (MELCAS) delay start
signaling support.
– e&m-melcas-immed—E&M MELCAS
immediate start signaling support.
– e&m-melcas-wink—E&M MELCAS
wink start signaling support.
– e&m-wink-start—The originating
endpoint sends an off-hook signal and
waits for a
– ext-sig—For the specified channel,
automatically generates the off-hook
signal and stays in the off-hook state.
– fgd-eana—Feature Group D Exchange
Access North American.
– fgd-os—Feature Group D Operator
Services.
– fxo-melcas—MELCAS Foreign
Exchange Office signaling support.
– fxs-melcas—MELCAS Foreign
Exchange Station signaling support.
– fxs-ground-start—FXS Ground Start.
– fxs-loop-start—FXS Loop Start.
– none—Null Signaling for External Call
Control.
– p7—Specifies the p7 switch type.
– r1-itu—R1 ITU
– sas-ground-start—SAS Ground Start.
– sas-loop-start—SAS Loop Start.
The r1 and r2 keywords refer to line signaling,
based on international signaling standards.
The r1 itu keywords are based on signaling
standards in countries besides the United States.
An “ITU variant” means that there are multiple R1
standards in a particular country but that Cisco
supports the ITU variant.
Step 7
Router(config-controller)# no shutdown
Activates the controller.
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Configuring Basic Voice Port Parameters for Digital T1/E1 Voice Ports
For FXO and FXS connections the default voice-port parameter values are often adequate. However, for
E&M connections, it is important to match the characteristics of your PBX, so voice port parameters may
need to be reconfigured from their defaults.
Each voice port that you address in digital voice port configuration is one of the logical voice ports that
you created with the ds0-group command.
Companding (from compression and expansion), used in Step 4 of the following table, is the part of the
PCM process in which analog signal values are logically rounded to discrete scale-step values on a
nonlinear scale. The decimal step number is then coded in its binary equivalent prior to transmission.
The process is reversed at the receiving terminal using the same nonlinear scale.
Note
The commands, keywords, and arguments that you are able to use may differ slightly from those
presented here, based on your platform, Cisco IOS release, and configuration. When in doubt, use
Cisco IOS command help (command ?) to determine the syntax choices that are available.
To configure basic parameters for digital T1/E1 voice ports, use the following commands beginning in
global configuration mode.
Step 1
Command
Purpose
Cisco 2600 and 3600 Series Routers
Enters voice-port configuration mode. The
arguments are defined as the following
Router(config)# voice-port slot/port:ds0-group-no
•
slot—Specifies the router location where the
network module (Cisco 2600, 3600, and
MC3810) or voice port adapter (Cisco
AS5300, AS5800, 7200, and 7500) is
installed. This is the same number as the
controller for the T1/E1 voice port.
•
port—Indicates the voice interface card
location.
•
ds0-group-no—Specifies the logical voice
port that was created with the ds0-group
controller command.
Router(config)# voice-port
slot/port-adapter:ds0-group-no
•
controller—Indicates the controller for the
T1/E1 voice port.
Cisco 7500 Series Routers
•
shelf—Specifies the dial shelf, which is
always 0.
•
port-adapter—Indicates the port adapter for
the voice port.
Cisco MC3810 Multiseries Concentrators
Router(config)# voice-port slot:ds0-group-no
Cisco AS5300 Universal Access Server
Router(config)# voice-port controller:ds0-group-no
Cisco AS5800 Universal Access Server
Router(config)# voice-port
shelf/slot/port:ds0-group-no
Cisco 7200 Series Routers
Router(config)# voice-port
slot/port-adapter/slot:ds0-group-no
Step 2
Router(config-voiceport)# type {1 | 2 | 3 | 5}
(E&M only) Specifies the type of E&M interface
to which this voice port is connected. See Table 5
for an explanation of E&M types.
The default is 1.
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Step 3
Command
Purpose
Router(config-voiceport)# cptone locale
Selects a two-letter locale keyword for the voice
call progress tones and other locale-specific
parameters to be used on this voice port. Voice call
progress tones include dial tone, busy tone, and
ringback tone, which vary with geographical
region.
Other parameters include ring cadence and
compand type. Cisco routers comply with the
ISO3166 locale name standards; to see valid
choices, enter a question mark (?) following the
cptone command.
The default is us.
Step 4
Router(config-voiceport)# compand-type {u-law | a-law}
(Cisco 2600 and 3600 series routers and
Cisco MC3810 multiservice concentrators only)
Specifies the companding standard used. This
command is used in cases when the DSP is not
used, such as local cross-connects, and overwrites
the compand-type value set by the cptone
command. The keywords are as follows:
•
a-law—Specifies the ITU-T PCM a-law
companding standard used primarily in
Europe. The default for E1 is a-law.
•
u-law—Specifies the ITU-T PCM mu-law
companding standard used in North America
and Japan. The default for T1 is u-law.
Note
Step 5
Cisco 2600 series and 3600 series
Router(config-voiceport)# ring frequency {25 | 50}
Cisco MC3810
Router(config-voiceport)# ring frequency {20 | 30}
If you have a Cisco MC3810 multiservice
concentrator or Cisco 3660 router, the
compand-type a-law command must be
configured on the analog ports only. The
Cisco 2660, 3620, and 3640 routers do not
require the compand-type a-law
command configured, however, if you
request a list of commands, the
compand-type a-law command will
display.
(FXS only) Selects the ring frequency, in hertz,
used on the FXS interface. This number must
match the connected telephony equipment, and
can be country-dependent. If not set properly, the
attached telephony device may not ring or it may
buzz.
The default is 25 on the Cisco 2600 and 3600
series routers and 20 on the Cisco MC3810
multiservice concentrators.
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Step 3
Command
Purpose
Router(config-voiceport)# cptone locale
Selects a two-letter locale keyword for the voice
call progress tones and other locale-specific
parameters to be used on this voice port. Voice call
progress tones include dial tone, busy tone, and
ringback tone, which vary with geographical
region.
Other parameters include ring cadence and
compand type. Cisco routers comply with the
ISO3166 locale name standards; to see valid
choices, enter a question mark (?) following the
cptone command.
The default is us.
Step 4
Router(config-voiceport)# compand-type {u-law | a-law}
(Cisco 2600 and 3600 series routers and
Cisco MC3810 multiservice concentrators only)
Specifies the companding standard used. This
command is used in cases when the DSP is not
used, such as local cross-connects, and overwrites
the compand-type value set by the cptone
command. The keywords are as follows:
•
a-law—Specifies the ITU-T PCM a-law
companding standard used primarily in
Europe. The default for E1 is a-law.
•
u-law—Specifies the ITU-T PCM mu-law
companding standard used in North America
and Japan. The default for T1 is u-law.
Note
Step 5
Cisco 2600 series and 3600 series
Router(config-voiceport)# ring frequency {25 | 50}
Cisco MC3810
Router(config-voiceport)# ring frequency {20 | 30}
If you have a Cisco MC3810 multiservice
concentrator or Cisco 3660 router, the
compand-type a-law command must be
configured on the analog ports only. The
Cisco 2660, 3620, and 3640 routers do not
require the compand-type a-law
command configured, however, if you
request a list of commands, the
compand-type a-law command will
display.
(FXS only) Selects the ring frequency, in hertz,
used on the FXS interface. This number must
match the connected telephony equipment, and
can be country-dependent. If not set properly, the
attached telephony device may not ring or it may
buzz.
The default is 25 on the Cisco 2600 and 3600
series routers and 20 on the Cisco MC3810
multiservice concentrators.
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Step 6
Command
Purpose
Router(config-voiceport)# ring number number
(FXO only) Specifies the maximum number of
rings to be detected before an incoming call is
answered by the router.
The default is 1.
Step 7
Router(config-voiceport)# ring cadence {[pattern01 |
pattern02 | pattern03 | pattern04 | pattern05 |
pattern06 | pattern07 | pattern08 | pattern09 |
pattern10 | pattern11 | pattern12] [define pulse
interval]}
(FXS only) Specifies an existing pattern for ring,
or defines a new one. Each pattern specifies a
ring-pulse time and a ring-interval time. The
keywords and arguments are as follows:
•
pattern01 through pattern12—Specifies
preset ring cadence patterns. Enter ring
cadence ? to see ring pattern explanations.
•
define pulse interval—Specifies a
user-defined pattern as follows:
– pulse is a number (1 or 2 digits from 1 to
50) specifying ring pulse (on) time in
hundreds of milliseconds.
– interval is a number (1 or 2 digits from 1
to 50) specifying ring interval (off) time
in hundreds of milliseconds.
The default is the pattern specified by the
configured cptone locale command.
Step 8
Router(config-voiceport)# description string
Attaches a text string to the configuration that
describes the connection for this voice port. This
description appears in various displays and is
useful for tracking the purpose or use of the voice
port. The string argument is a character string
from 1 to 255 characters in length.
The default is that no description is attached to the
configuration.
Step 9
Router(config-voiceport)# no shutdown
Activates the voice port.
Fine-Tuning Analog and Digital Voice Ports
Normally, default parameter values for voice ports are sufficient for most networks. Depending on the
specifics of your particular network, however, you may need to adjust certain parameters that are
configured on voice ports. Collectively, these commands are referred to as voice port tuning commands.
Note
The commands, keywords, and arguments that you are able to use may differ slightly from those
presented here, based on your platform, Cisco IOS release, and configuration. When in doubt, use
Cisco IOS command help (command ?) to determine the syntax choices that are available.
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The voice port tuning commands are grouped into these categories and explained in the following
sections:
•
Auto Cut-Through Command, page 80
•
Bit Modification Commands for Digital Voice Ports, page 80
•
Calling Number Outbound Commands, page 82
•
Disconnect Supervision Commands, page 83
•
FXO Supervisory Disconnect Tone Commands, page 85
•
Timeouts Commands, page 87
•
Timing Commands, page 89
•
DTMF Timer Inter-Digit Command for Cisco AS5300 Access Servers, page 90
•
Voice Quality Tuning Commands, page 92
Full descriptions of the commands in this section can be found in the Cisco IOS Voice, Video, and Fax
Command Reference, Release 12.2.
Auto Cut-Through Command
The auto-cut-through command allows you to connect to PBXs that do not provide an M-lead response.
To configure auto-cut-through, use the following command in voice-port configuration mode:
Command
Purpose
Router(config-voiceport)# auto-cut-through
(E&M only) Enables call completion on a router when a PBX
does not provide an M-lead response.
Bit Modification Commands for Digital Voice Ports
The bit modification commands for digital voice ports modify sent or received bit patterns. Different
versions of E&M use different ABCD signaling bits to represent idle and seize. For example, North
American CAS E&M represents idle as 0XXX and seize as 1XXX, where X indicates that the state of
the BCD bits is ignored. In MELCAS E&M, idle is 1101 and seize is 0101. The commands in this section
are provided to modify bit patterns to match particular E&M schemes.
To manipulate bit patterns for digital voice ports, use the following commands as necessary, in voice-port
configuration mode:
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Step 1
Command
Purpose
Router(config-voiceport)# condition {tx-a-bit |
tx-b-bit | tx-c-bit | tx-d-bit} {rx-a-bit | rx-b-bit |
rx-c-bit | rx-d-bit} {on | off | invert}
Manipulates sent or received bit patterns to match
expected patterns on a connected device. Repeat
the command for each transmit and/or receive bit
to be modified, but be careful not to destroy the
information content of the bit pattern.
The default is that the signaling format is not
manipulated (for all transmit or receive A, B, C,
and D bits).
The keywords are as follows:
•
on—Sets the bit to 1 permanently.
•
off—Sets the bit to 0 permanently.
•
invert—Changes the state to the opposite of
the original transmit or receive state.
Note
Step 2
Router(config-voiceport)# define {tx-bits | rx-bits}
{seize | idle} {0000 | 0001 | 0010 | 0011 | 0100 |
0101 | 0110 | 0111 | 1000 | 1001 | 1010 | 1011 | 1100
| 1101 | 1110 | 1111}
The show voice port command reports at
the protocol level, and the show
controller command reports at the driver
level. The driver is not notified of any bit
manipulation using the condition
command. As a result, the show
controller command output does not
account for the bit conditioning.
(Digital E1 E&M voice ports on Cisco 2600 and
3600 series routers and Cisco MC3810
multiservice concentrators only) Defines specific
transmit or receive signaling bits to match the bit
patterns required by a connected device for North
American E&M and E&M MELCAS voice
signaling, if patterns different from the preset
defaults are required.
Also specifies which bits a voice port monitors
and which bits it ignores, if patterns that are
different from the defaults are required.
See the define command for the default signaling
patterns as defined in American National
Standards Institute (ANSI) and code excited linear
prediction compression (CEPT) standards. The
keywords are as follows:
•
tx-bits—Indicates the pattern applies to
transmit signaling bits.
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Command
Step 3
Purpose
Router(config-voiceport)# ignore {rx-a-bit | rx-b-bit
| rx-c-bit | rx-d-bit}
•
rx-bits—Indicates the pattern applies to
receive signaling bits
•
seize—Indicates that the pattern represents
line seizure.
•
idle—Indicates that the pattern represents an
idle condition.
•
0000...1111—Represents the bit pattern to
use.
(Digital E1 E&M voice ports on Cisco 2600 and
3600 series routers and Cisco MC3810
multiservice concentrators only) Configures the
voice port to ignore the specified receive bit for
North American E&M or E&M MELCAS, if
patterns different from the defaults are required.
See the command reference for the default
signaling patterns as defined in ANSI and CEPT
standards.
Calling Number Outbound Commands
On the Cisco AS5300 universal access server platform, if T1 CAS is configured with the Feature
Group-D (FGD)—Exchange Access North American (FGD-EANA) signaling, the automatic number
identification (ANI) can be sent for outgoing calls by using the calling-number outbound command.
FGD-EANA is a FGD signaling protocol of type EANA, which provides certain call services, such as
emergency (USA 911) calls. ANI is an SS7 (Signaling System 7) feature in which a series of digits,
analog or digital, are included in the call to identify the telephone number of the calling device. In other
words, ANI identifies the number of the calling party. ANI digits are used for billing purposes by Internet
service providers (ISPs), among other things. The commands in this section can be issued in voice-port
or dial-peer mode, because the syntax is the same.
To configure your digital T1/E1 packet voice trunk network module to generate outbound ANI digits on
a Cisco AS5300 universal access server, use the following commands in voice-port configuration mode:
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Step 1
Command
Purpose
Router(config-voiceport)# calling-number outbound
range string1 string2
(Cisco AS5300 universal access server only)
Specifies ANI to be sent out when the T1-CAS
fgd-eana command is configured as signaling
type. The string1 and string2 arguments are valid
E.164 telephone number strings. Both strings must
be of the same length and cannot be more than 32
digits long.
Only the last four digits are used for specifying the
range (string1 to string2) and for generating the
sequence of ANI by rotating through the range
until string2 is reached and then starting from
string1 again. If strings are less than four digits in
length, then entire strings are used.
Step 2
Router(config-voiceport)# calling-number outbound
sequence [string1] [string2] [string3] [string4]
[string5]
(Cisco AS5300 universal access server only)
Specifies ANI to be sent out when the T1-CAS
fgd-eana command is configured as signaling
type. This option configures a sequence of discrete
strings (string1...string5) to be passed out as ANI
for successive calls using the dial peer or voice
port. Limit is five (5) strings. All strings must be
valid E.164 numbers, up to 32 digits in length.
Step 3
Router(config-voiceport)# calling-number outbound null
(Cisco AS5300 universal access server only)
Suppresses ANI. No ANI is passed when this
voice port is selected.
Disconnect Supervision Commands
PBX and PSTN switches use several different methods to indicate that a call should be disconnected
because one or both parties have hung up. The commands in this section are used to configure the router
to recognize the type of signaling in use by the PBX or PSTN switch connected to the voice port. These
methods include the following:
•
Battery reversal disconnect
•
Battery denial disconnect
•
Supervisory tone disconnect (STD)
Battery reversal occurs when the connected switch changes the polarity of the line in order to indicate
changes in call state (such as off-hook or, in this case, call disconnect). This is the signaling looked for
when the battery reversal command is enabled on the voice port, which is the default configuration.
Battery denial (sometimes called power denial) occurs when the connected switch provides a short
(approximately 600 ms) interruption of line power to indicate a change in call state. This is the signaling
looked for when the supervisory disconnect command is enabled on the voice port, which is the default
configuration.
Supervisory tone disconnect occurs when the connected switch provides a special tone to indicate a
change in call state. Some PBXs and PSTN CO switches provide a 600-millisecond interruption of line
power as a supervisory disconnect, and others provide supervisory tone disconnect (STD). This is the
signal that the router is looking for when the no supervisory disconnect command is configured on the
voice port.
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Note
In some circumstances, you can use the FXO Disconnect Supervision feature to enable analog FXO
ports to monitor call progress tones for disconnect supervision that are returned from a PBX or from
the PSTN. For more information, see the “FXO Supervisory Disconnect Tone Commands” section
on page 85.
To change parameters related to disconnect supervision, use the following commands as appropriate, in
voice-port configuration mode:
Step 1
Command
Purpose
Router(config-voiceport)# no battery-reversal
(Analog only) Enables battery reversal. The
default is that battery reversal is enabled.
•
For FXO ports—Use the no battery-reversal
command to configure a loop-start voice port
not to disconnect when it detects a second
battery reversal. The default is to disconnect
when a second battery reversal is detected.
This functionality is supported on
Cisco MC3810 analog voice ports; on
Cisco 1750, Cisco 2600 series, and
Cisco 3600 series routers, only analog voice
ports on VIC-2FXO cards are able to detect
battery reversal.
– Also use the no battery-reversal
command when a connected FXO port
does not support battery reversal
detection.
•
For FXS ports—Use the no battery-reversal
command to configure the voice port not to
reverse battery when it connects calls. The
default is to reverse battery when a call is
connected, then return to normal when the call
is over, providing positive disconnect.
See also the disconnect-ack command (Step 7).
Step 2
Router(config-voiceport)# no supervisory disconnect
(FXO only) Enables the PBX or PSTN switch to
provide STD. By default the supervisory
disconnect command is enabled.
Step 3
Router(config-voiceport)# disconnect-ack
(FXS only) Configures the voice port to return an
acknowledgment upon receipt of a disconnect
signal. The FXS port removes line power if the
equipment on the FXS loop-start trunk
disconnects first. This is the default.
The no disconnect-ack command prevents the
FXS port from responding to the on-hook
disconnect with a removal of line power.
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FXO Supervisory Disconnect Tone Commands
If the FXO supervisory disconnect tone is configured and a detectable tone from the PSTN or PBX is
detected by the digital signal processor (DSP), the analog FXO port goes on-hook. This feature prevents
an analog FXO port from remaining in an off-hook state after an incoming call is ended. FXO
supervisory disconnect tone enables interoperability with PSTN and PBX systems whether or not they
transmit supervisory tones.
Note
This feature applies only to analog FXO ports with loop-start signaling on the Cisco 2600 and 3600
series routers and on Cisco MC3810 multiservice concentrators with high-performance compression
modules (HCMs).
To configure a voice port to detect incoming tones, you need to know the parameters of the tones
expected from the PBX or PSTN. Then create a voice class that defines the tone detection parameters,
and, finally, apply the voice class to the applicable analog FXO voice ports. This procedure configures
the voice port to go on-hook when it detects the specified tones. The parameters of the tones need to be
precisely specified to prevent unwanted disconnects due to detection of nonsupervisory tones or noise.
A supervisory disconnect tone is normally a dual tone consisting of two frequencies; however, tones of
only one frequency can also be detected. Use caution if you configure voice ports to detect nondual
tones, because unwanted disconnects can result from detection of random tone frequencies. You can
configure a voice port to detect a tone with one on/off time cycle, or you can configure it to detect tones
in a cadence pattern with up to four on/off time cycles.
Note
In the following procedure, the following commands were not supported until Cisco IOS Release
12.2(2)T: freq-max-deviation, freq-max-power, freq-min-power, freq-power-twist, and
freq-max-delay.
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To create a voice class that defines the specific tone or tones to be detected and then apply the voice class
to the voice port, use the following commands beginning in global configuration mode:
Step 1
Command
Purpose
Router(config)# voice class dualtone tag
Creates a voice class for defining one tone
detection pattern. The range for the tag number is
from 1 to 10000. The tag number must be unique
on the router.
For more information about configuring voice
classes, see the “Configuring Dial Plans, Dial
Peers, and Digit Manipulation” chapter in this
configuration guide.
Step 2
Router(config-voice-class)# freq-pair tone-id
frequency-1 frequency-2
Specifies the two frequencies, in Hz, for a tone to
be detected (or one frequency if a nondual tone is
to be detected). If the tone to be detected contains
only one frequency, enter 0 for frequency-2. The
arguments are as follows:
•
tone-id—Ranges from 1 to 16. There is no
default.
•
frequency-1 and frequency-2—Ranges from
300 to 3600, or you can enter 0 for
frequency-2. There is no default.
Note
Repeat this command for each additional
tone to be specified.
Step 3
Router(config-voice-class)# freq-max-deviation
frequency
Specifies the maximum frequency deviation that
will be detected, in Hz. The frequency argument
ranges from 10 to 125. The default is 10.
Step 4
Router(config-voice-class)# freq-max-power dBmO
Specifies the maximum tone power that will be
detected, in dBmO. The dBmO argument ranges
from 0 to 20. The default is 10.
Step 5
Router(config-voice-class)# freq-min-power dBmO
Specifies the minimum tone power that will be
detected, in dBmO. The dBmO argument ranges
from 10 to 35. The default is 30.
Step 6
Router(config-voice-class)# freq-power-twist dBmO
Specifies the power difference allowed between
the two frequencies, in dBmO. The dBmO
argument ranges from 0 to 15. The default is 6.
Step 7
Router(config-voice-class)# freq-max-delay time
Specifies the timing difference allowed between
the two frequencies, in 10-millisecond increments.
The time argument ranges from 10 to 100 (100 ms
to 1 s). The default is 20 (200 ms).
Step 8
Router(config-voice-class)# cadence-min-on-time time
Specifies the minimum tone on time that will be
detected, in 10-millisecond increments. The time
argument ranges from 0 to 100 (0 ms to 1 s).
Step 9
Router(config-voice-class)# cadence-max-off-time time
Specifies the maximum tone off time that will be
detected, in 10-millisecond increments. The time
argument ranges from 0 to 5000 (0 ms to 50 s).
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Step 10
Command
Purpose
Router(config-voice-class)# cadence-list cadence-id
cycle-1-on-time cycle-1-off-time cycle-2-on-time
cycle-2-off-time cycle-3-on-time cycle-3-off-time
cycle-4-on-time cycle-4-off-time
(Optional) Specifies a tone cadence pattern to be
detected. Specify an on time and off time for each
cycle of the cadence pattern.
The arguments are as follows:
•
cadence-id—Ranges from 1 to 10. There is no
default.
•
cycle-N-on-time and
cycle-N-off-time—Range from 0 to 1000 (0
ms to 10 s). The default is 0.
Step 11
Router(config-voice-class)# cadence-variation time
(Optional) Specifies the maximum time that the
tone onset can vary from the specified onset time
and still be detected, in 10-millisecond
increments. The time argument ranges from 0 to
200 (0 ms to 2 s). The default is 0.
Step 12
Router(config-voice-class)# exit
Exits voice class configuration mode.
Step 13
Cisco 2600 and 3600 Series Routers
Enters voice-port configuration mode.
Router(config)# voice-port slot/subunit/port
The arguments are as follows:
Cisco MC3810 Multiservice Concentrators
•
slot—Specifies the slot number where the
voice network module is installed (Cisco 2600
and Cisco 3600 series routers) or the router
slot number where the analog voice module is
installed (Cisco MC3810 multiservice
concentrators).
•
subunit—Specifies the voice interface card
(VIC) where the voice port is located.
•
port—Identifies the analog voice-port
number.
Router(config)# voice-port slot/port
Step 14
Router(config-voiceport)# supervisory disconnect
dualtone {mid-call | pre-connect} voice-class tag
Assigns an FXO supervisory disconnect tone
voice class to the voice port.
The keywords are as follows:
Step 15
Router(config-voiceport)# supervisory disconnect
anytone
•
mid-call—Specifies tone detection during the
entire call.
•
pre-connect—Specifies tone detection only
during call set-up.
Configures the voice port to disconnect on receipt
of any tone.
Timeouts Commands
To change timeouts parameters, use the following commands as appropriate, in voice-port configuration
mode:
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Command
Purpose
Step 1
Router(config-voiceport)# timeouts call-disconnect
seconds
Configures the call disconnect timeout value in
seconds. Valid entries range from 0 to 120. The
default is 60.
Step 2
Router(config-voiceport)# timeouts initial seconds
Sets the number of seconds that the system waits
between the caller input of the initial digit and the
subsequent digit of the dialed string. If the wait
time expires before the destination is identified, a
tone sounds and the call ends. The seconds
argument is the initial timeout duration. A valid
entry is an integer from 0 to 120. The default is 10.
Step 3
Router(config-voiceport)# timeouts interdigit seconds
Configures the number of seconds that the system
waits after the caller has input the initial digit or a
subsequent digit of the dialed string. If the timeout
ends before the destination is identified, a tone
sounds and the call ends. This value is important
when using variable-length dial peer destination
patterns (dial plans). The seconds argument is the
interdigit timeout wait time in seconds. A valid
entry is an integer from 0 to 120. The default is 10.
Step 4
Router(config-voiceport)# timeouts ringing {seconds |
infinity}
Specifies the duration that the voice port allows
ringing to continue if a call is not answered.
The keyword and argument are as follows:
•
infinity—Indicates ringing should continue
until the caller goes on hook.
•
seconds—Specifies the number of seconds to
allow ringing without answer. The range is
from 5 to 60000.
The default is 180.
Step 5
Router(config-voiceport)# timeouts wait-release
{seconds | infinity}
Specifies the duration that a voice port stays in the
call-failure state while the Cisco device sends a
busy tone, reorder tone, or an out-of-service tone
to the port.
The keyword and argument are as follows:
•
infinity—Indicates the voice port should not
be released as long as the call-failure state
remains.
•
seconds—Specifies the number of seconds to
allow before the call is released. The range is
from 3 to 3600.
The default is 30.
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Timing Commands
To change timing parameters, use the following commands as appropriate, in voice-port configuration
mode:
Command
Purpose
Step 1
Router(config-voiceport)# timing clear-wait
milliseconds
(E&M only) Specifies the minimum amount of
time between the inactive seizure signal and
clearing of the call. Valid entries for the
milliseconds argument are from 200 to
2000 milliseconds. The default is 400.
Step 2
Router(config-voiceport)# timing delay-duration
milliseconds
(E&M only) Specifies the delay signal duration
for delay-dial signaling in milliseconds. Valid
entries are from 100 to 5000. The default is 2000.
Step 3
Router(config-voiceport)# timing delay-start
milliseconds
(E&M only) Specifies minimum delay time, in
milliseconds, from outgoing seizure to outdial
address. Valid entries are from 20 to 2000.
The default is 300 for the Cisco 3600 series
routers, and 150 for the Cisco MC3810
multiservice concentrators.
Step 4
Router(config-voiceport)# timing delay-with-integrity
milliseconds
(Cisco MC3810 multiservice concentrators E&M
ports only) Specifies duration of the wink pulse
for the delay dial in milliseconds. Valid entries are
from 0 to 5000. The default is 0.
Step 5
Router(config-voiceport)# timing dial-pulse min-delay
milliseconds
Specifies time, in milliseconds, between the
generation of wink-like pulses when the type is
pulse. Valid entries are from 0 to 5000.
The default is 300 for the Cisco 3600 series
routers, and 140 for the Cisco MC3810
multiservice concentrators.
Step 6
Router(config-voiceport)# timing dialout-delay
milliseconds
(Cisco MC3810 multiservice concentrators only)
Specifies dialout delay, in milliseconds, for the
sending digit or cut-through on an FXO trunk or
an E&M immediate trunk. Valid entries are from
100 to 5000. The default is 300.
Step 7
Router(config-voiceport)# timing digit milliseconds
Specifies the DTMF digit signal duration in
milliseconds. Valid entries are from 50 to 100. The
default is 100.
Step 8
Router(config-voiceport)# timing guard-out
milliseconds
(FXO ports only) Specifies the duration in
milliseconds of the guard-out period that prevents
this port from seizing a remote FXS port before
the remote port detects a disconnect signal. The
range is from 300 to 3000. The default is 2000.
Step 9
Router(config-voiceport)# timing hookflash-out
milliseconds
Specifies the duration, in milliseconds, of the
hookflash. Valid entries are from 50 to 500. The
default is 300.
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Command
Purpose
Step 10
Router(config-voiceport)# timing interdigit
milliseconds
Specifies the DTMF interdigit duration, in
milliseconds. Valid entries are from 50 to 500. The
default is 100.
Step 11
Router(config-voiceport)# timing percentbreak percent
(Cisco MC3810 multiservice concentrators FXO
and E&M ports only) Specifies the percentage of
the break period for the dialing pulses, if different
from the default. The range is from 20 to 80. The
default is 50.
Step 12
Router(config-voiceport)# timing pulse
pulses-per-second
(FXO and E&M only) Specifies the pulse dialing
rate in pulses per second. Valid entries are from 10
to 20. The default is 20.
Step 13
Router(config-voiceport)# timing pulse-digit
milliseconds
(FXO only) Configures the pulse digit signal
duration. The range of the pulse digit signal
duration is from 10 to 20. The default is 20.
Step 14
Router(config-voiceport)# timing pulse-interdigit
(FXO and E&M only) Specifies pulse dialing
interdigit timing in milliseconds. Valid entries are
from 100 to 1000. The default is 500.
Step 15
Router(config-voiceport)# timing wink-duration
milliseconds
(E&M only) Specifies maximum wink-signal
duration, in milliseconds, for a wink-start signal.
Valid entries are from 100 to 400. The default is
200.
Step 16
Router(config-voiceport)# timing wink-wait
milliseconds
(E&M only) Specifies maximum wink-wait
duration, in milliseconds, for a wink-start signal.
Valid entries are from 100 to 5000. The default is
200.
DTMF Timer Inter-Digit Command for Cisco AS5300 Access Servers
To configure the DTMF timer for Cisco AS5300 access servers, use the following commands beginning
in global configuration mode:
Command
Purpose
Step 1
Router(config)# controller T1 number
Configures a T1 controller and enters controller
configuration mode.
Step 2
Router(config)# ds0-group channel-number timeslots
range type signaling-type dtmf dnis
Configures channelized T1 timeslots, which enables a
Cisco AS5300 modem to answer and send an analog
call.
Step 3
Router(config)# cas-custom channel
Customizes E1 R2 signaling parameters for a
particular E1 channel group on a channelized E1 line.
Step 4
Router(conf-ctrl-cas)# dtmf-timer-inter-digit
milliseconds
Configures the DTMF inter-digit timer for a DS0
group.
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Configuring Voice Ports
Configuring Digital Voice Ports
Verifying DTMF Timer Inter-Digit Command
To verify the DTMF timer, use the following command in EXEC mode:
Command
Purpose
Router# show running-config
Displays the configuration information currently
running on the terminal.
Voice Activity Detection Commands Related to Voice-Port Configuration Mode
In normal voice conversations, only one person speaks at a time. Today’s circuit-switched telephone
networks dedicate a bidirectional, 64 kbps channel for the duration of each conversation, regardless of
whether anyone is speaking at the moment. This means that, in a normal voice conversation, at least
50 percent of the bandwidth is wasted when one or both parties are silent. This figure can actually be
much higher when normal pauses and breaks in conversation are taken into account.
Packet-switched voice networks, on the other hand, can use this “wasted” bandwidth for other purposes
when voice activity detection (VAD) is configured. VAD works by detecting the magnitude of speech in
decibels and deciding when to cut off the voice from being framed. VAD has some technological
problems, however, which include the following:
•
General difficulties determining when speech ends
•
Clipped speech when VAD is slow to detect that speech is beginning again
•
Automatic disabling of VAD when conversations take place in noisy surroundings
VAD is configured on dial peers; by default it is enabled. For more information, see the “Configuring
Dial Plans, Dial Peers, and Digit Manipulation” chapter in this configuration guide. Two parameters
associated with VAD, music threshold and comfort noise, are configured on voice ports.
If VAD is enabled, use the following commands to adjust parameter values associated with VAD,
beginning in voice-port configuration mode:
Command
Purpose
Step 1
Router(config-voiceport)# music-threshold number
Specifies the minimal decibel level of music
played when calls are put on hold. The decibel
level affects how voice activity detection (VAD)
treats the music data. Valid entries range from –70
to –30. When used with VAD, if the level is set too
high, the remote end hears no music; if it is set too
low, there is unnecessary voice traffic. The default
is –38.
Step 2
Router(config-voiceport)# comfort-noise
This parameter creates subtle background noise to
fill silent gaps during calls when VAD is enabled
on voice dial peers. If comfort noise is not
generated, the resulting silence can fool the caller
into thinking the call is disconnected instead of
being merely idle. The default is that comfort
noise is enabled.
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Configuring Voice Ports
Configuring Digital Voice Ports
Voice Quality Tuning Commands
The commands in this section configure parameters to improve voice quality. Common voice quality
issues include the following:
•
Delay in Voice Networks
•
Jitter Adjustment
•
Echo Adjustment
•
Voice Level Adjustment
Delay in Voice Networks
Delay is the time it takes for voice packets to travel between two endpoints. Excessive delay can cause
quality problems with real-time traffic such as voice. However, because of the speed of network links
and the processing power of intermediate devices, some delay is expected.
When listening to speech, the human ear normally accepts up to about 150 ms of delay without noticing
delays. The ITU G.114 standard recommends no more than 150 ms of one-way delay for a normal voice
conversation. Once the delay exceeds 150 ms, a conversation is more like a “walkie-talkie” conversation
in which one person must wait for the other to stop speaking before beginning to talk.
You can measure delay fairly easily by using ping tests at various times of the day with different network
traffic loads. If network delay is excessive, it must be reduced for adequate voice quality.
Several different types of delay combine to make up the total end-to-end delay associated with voice
calls:
•
Propagation delay—Amount of time it takes the data to physically travel over the media.
•
Handling delay—Amount of time it takes to process data by adding headers, taking samples,
forming packets, etc.
•
Queuing delay—Amount of time lost due to congestion.
•
Variable delay or jitter—Amount of time that causes the conversation to break and become
unintelligible. Jitter is described in detail below.
Propagation, handling, and queuing delay are not addressed by voice-port commands and fall outside the
scope of this chapter.
Jitter Adjustment
Delay can cause unnatural starting and stopping of conversations, but variable-length delays (also known
as jitter) can cause a conversation to break and become unintelligible. Jitter is not usually a problem with
PSTN calls because the bandwidth of calls is fixed and each call has a dedicated circuit for the duration
of the call. However, in VoIP networks, data traffic might be bursty, and jitter from the packet network
can become an issue. Especially during times of network congestion, packets from the same conversation
can arrive at different interpacket intervals, disrupting the steady, even delivery needed for voice calls.
Cisco voice gateways have built-in jitter buffering to compensate for a certain amount of jitter; the
playout-delay command can be used to adjust the jitter buffer.
Normally, the defaults in effect are sufficient for most networks. However, a small playout delay from
the jitter buffer can cause lost packets and choppy audio, and a large playout delay can cause
unacceptably high overall end-to-end delay.
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Configuring Digital Voice Ports
Note
Prior to Cisco IOS Release 12.1(5)T, playout delay was configured in voice-port configuration mode.
For Cisco IOS Release 12.1(5)T and later releases, in most cases playout delay should be configured
in dial-peer configuration mode on the VoIP dial peer that is on the receiving end of the voice traffic
that is to be buffered. This dial peer senses network conditions and relays them to the DSPs, which
adjust the jitter buffer as necessary. When multiple applications are configured on the gateway,
playout delay should be configured in dial-peer configuration mode. When there are numerous dial
peers to configure, it might be simpler to configure playout delay on a voice port. If there are
conflicting playout delay configurations on a voice port and also on a dial peer, the dial peer
configuration takes precedence.
To configure the playout delay jitter buffer, use the following commands beginning in dial-peer or
voice-port configuration mode:
Step 1
Command
Purpose
Router(config-voiceport)# playout-delay mode {adaptive
| fixed}
Determines the mode in which the jitter buffer will
operate for calls on this voice port.
The keywords are as follows:
•
adaptive—Adjusts the jitter buffer size and
amount of playout delay during a call based
on current network conditions.
•
fixed—Defines the jitter buffer size as fixed
so that the playout delay does not adjust
during a call. A constant playout delay is
added.
The default is adaptive.
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Configuring Digital Voice Ports
Step 2
Command
Purpose
Router(config-voiceport)# playout-delay {nominal value
| maximum value | minimum {default | low | high}}
Tunes the playout buffer to accommodate packet
jitter caused by switches in the WAN.
The keywords and arguments are as follows:
•
nominal—Defines the amount of playout
delay applied at the beginning of a call by the
jitter buffer in the gateway. In fixed mode, this
is also the maximum size of the jitter buffer
throughout the call.
•
value—Specifies the range that depends on
type of DSP and configured codec
complexity. For medium codec complexity,
the range is from 0 to 150 ms. For high codec
complexity and DSPs that do not support
codec complexity, the range is from
0 to 250 ms.
•
maximum (adaptive mode only)—Specifies
the jitter buffer's upper limit (80ms), or the
highest value to which the adaptive delay is
set.
•
minimum (adaptive mode only)—Specifies
the jitter buffer's lower limit (10 ms), or the
lowest value to which the adaptive delay is
set.
•
default—Specifies 40 ms.
Echo Adjustment
Echo is the sound of your own voice reverberating in the telephone receiver while you are talking. When
timed properly, echo is not a problem in the conversation; however, if the echo interval exceeds
approximately 25 milliseconds, it is distracting. Echo is controlled by echo cancellers.
In the traditional telephony network, echo is generally caused by an impedance mismatch when the
four-wire network is converted to the two-wire local loop. In voice packet-based networks, echo
cancellers are built into the low-bit rate codecs and are operated on each DSP.
By design, echo cancellers are limited by the total amount of time they wait for the reflected speech to
be received, which is known as an echo trail. The echo trail is normally 32 milliseconds. In Cisco
System’s voice implementations, echo cancellers are enabled using the echo-cancel enable command,
and echo trails are configured using the echo-cancel coverage command.
To configure parameters related to the echo canceller, use the following commands beginning in
voice-port configuration mode:
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Configuring Digital Voice Ports
Step 1
Command
Purpose
Router(config-voiceport)# echo-cancel enable
Enables the cancellation of voice that is sent and
received on the same interface. Echo cancellation
coverage must also be configured. The default is
that echo cancellation is enabled.
Note
Not valid for four-wire E&M interfaces.
Use no echo-cancel enable to disable the
feature.
Step 2
Router(config-voiceport)# echo-cancel coverage {8 | 16
| 24 | 32}
Adjusts the echo canceller by the specified number
of milliseconds. The default is 16.
Step 3
Router(config-voiceport)# non-linear
Enables nonlinear processing (residual echo
suppression) in the echo canceler, which shuts off
any signal if no near-end speech is detected. Echo
cancelling must be enabled for this feature. The
default is that nonlinear processing is enabled.
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Configuring Voice Ports
Configuring Digital Voice Ports
Voice Level Adjustment
As much as possible, it is desirable to achieve a uniform input decibel level to the packet voice network
in order to limit or eliminate any voice distortion due to incorrect input and output decibel levels.
Adjustments to levels may be required by the type of equipment connected to the network or by local
country-specific conditions.
Incorrect input or output levels can cause echo, as can an impedance mismatch. Too much input gain can
cause clipped or fuzzy voice quality. If the output level is too high at the remote router’s voice port, the
local caller will hear echo. If the local router’s voice port input decibel level is too high, the remote side
will hear clipping. If the local router’s voice port input decibel level is too low, or the remote router’s
output level is too low, the remote side voice can be distorted at a very low volume and DTMF may be
missed.
Use the input gain and output attenuation commands to adjust voice levels, and the impedance
command to set the impedance value to match that of the voice circuit to which the voice port connects.
To change parameters related to voice levels, use the following commands as appropriate, in voice-port
configuration mode:
Command
Purpose
Step 1
Router(config-voiceport)# input gain value
Specifies, in decibels, the amount of gain to be
inserted at the receiver side of the interface,
increasing or decreasing the signal. After an input
gain setting is changed, the voice call must be
disconnected and reestablished before the changes
take effect. The value argument is any integer
from –6 to 14. The default is 0.
Step 2
Router(config-voiceport)# output attenuation value
Specifies the amount of attenuation in decibels at
the transmit side of the interface, decreasing the
signal. A system-wide loss plan can be
implemented using the input gain and output
attenuation commands.
The default value for this command assumes that a
standard transmission loss plan is in effect,
meaning that normally there must be –6 dB
attenuation between phones.
The value argument is any integer from –6 to 14.
The default is 0.
Step 3
Router(config-voiceport)# impedance {600c | 600r |
900c | complex1 | complex2}
Specifies the terminating impedance of a voice
port interface, which needs to match the
specifications from the specific telephony system
to which it is connected.
•
600c—Specifies 600 ohms complex.
•
600r—Specifies 600 ohms real.
•
900c—Specifies 900 ohms complex.
•
complex1—Specifies Complex 1.
•
complex2—Specifies Complex 2.
The default is 600r.
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Verifying Analog and Digital Voice-Port Configurations
Command
Purpose
Step 4
Router(config-voiceport)# loss-plan {plan1 | plan2 |
plan5 | plan6 | plan7 | plan8 | plan9}
(Cisco MC3810 multiservice concentrators FXO
or FXS analog voice ports only) Specifies the
analog-to-digital gain offset loss plan. For
definitions of each plan, see the Cisco IOS Voice,
Video, and Fax Command Reference. The default
is the plan1 keyword.
Step 5
Router(config-voiceport)# idle-voltage {high | low}
(Cisco MC3810 multiservice concentrators analog
FXS ports only) Specifies the talk-battery
(tip-to-ring) voltage condition when the port is
idle.
The keywords are as follows:
•
high—Specifies that the voltage is high
(–48V).
•
low—Specifies that the voltage is low (–24V)
and is the default.
Verifying Analog and Digital Voice-Port Configurations
After configuring the voice ports on your router, perform the following steps to verify proper operation:
Step 1
Pick up the handset of an attached telephony device and check for a dial tone.
Step 2
If you have dial tone, check for DTMF detection. If the dial tone stops when you dial a digit, then the
voice port is most likely configured properly.
Step 3
To identify port numbers of voice interfaces installed in your router, use the show voice port summary
command. For examples of the output, see the “show voice port summary Command Examples” section
on page 98.
Step 4
To verify voice-port parameter settings, use the show voice port command with the appropriate syntax
from Table 9. For sample output, see the “show voice port Command Examples” section on page 99.
Table 9
Show Voice Port Command Syntax
Platform
Voice Port Type
Command Syntax
Cisco 1750
Analog
show voice port [slot/port | summary]
Cisco 2600 series
Cisco 3600 series
Analog
show voice port [slot/port | summary]
Digital
show voice port [slot/port:ds0-group-no | summary]
Cisco MC3810
Analog
show voice port [slot/port | summary]
Digital
show voice port [slot:ds0-group-no | summary]
Cisco AS5300
Digital
show voice-port controller:ds0-group-no
Cisco AS5800
Digital
show voice-port {shelf/slot/port:ds0-group-no}
Cisco 7200 series
Digital
show voice port {slot/port-adapter:ds0-group-no}
Cisco 7500 series
Digital
show voice port {slot/port-adapter/slot:ds0-group-no}
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Verifying Analog and Digital Voice-Port Configurations
Step 5
For digital T1/E1 connections, to verify the codec complexity configuration, use the show
running-config command to display the current voice-card setting. If medium complexity is specified,
the codec complexity setting is not displayed. If high complexity is specified, the setting codec
complexity high is displayed. The following example shows an excerpt from the command output when
high complexity has been specified:
Router# show running-config
.
.
.
hostname router-alpha
voice-card 0
codec complexity high
.
.
.
Step 6
For digital T1/E1 connections, to verify that the controller is up and that no alarms have been reported,
and to display information about clock sources and other controller settings, use the show controller
command. For output examples, see the “show controller Command Examples” section on page 103.
Router# show controller {t1 | e1} controller-number
Step 7
To display voice-channel configuration information for all DSP channels, use the show voice dsp
command. For output examples, see the “show voice dsp Command Examples” section on page 104.
Router# show voice dsp
Step 8
To verify the call status for all voice ports, use the show voice call summary command. For output
examples, see the “show voice call summary Command Examples” section on page 105.
Router# show voice call summary
Step 9
To display the contents of the active call table, which shows all of the calls currently connected through
the router or concentrator, use the show call active voice command. For output examples, see the “show
call active voice Command Example” section on page 105.
Router# show call active voice
Step 10
To display the contents of the call history table, use the show call history voice command. To limit the
display to the last calls connected through this router, use the keyword last and define the number of
calls to be displayed with the argument number. To limit the display to a shortened version of the call
history table, use the brief keyword. For output examples, see the “show call history voice Command
Example” section on page 106.
Router# show call history voice [last | number | brief]
show voice port summary Command Examples
In the following sections, output examples of the following types are shown:
•
Cisco 3640 Router Analog Voice Port
•
Cisco MC3810 Multiservice Concentrator Digital Voice Port
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Configuring Voice Ports
Verifying Analog and Digital Voice-Port Configurations
Cisco 3640 Router Analog Voice Port
The following output is from a Cisco 3640 router:
Router# show voice port summary
PORT
======
2/0/0
2/0/1
2/1/0
2/1/1
CH
==
-----
SIG-TYPE
==========
e&m-wnk
e&m-wnk
fxs-ls
fxs-ls
ADMIN
=====
up
up
up
up
OPER
====
dorm
dorm
dorm
dorm
IN
STATUS
========
idle
idle
on-hook
on-hook
OUT
STATUS
========
idle
idle
idle
idle
EC
==
y
y
y
y
Cisco MC3810 Multiservice Concentrator Digital Voice Port
The following output is from a Cisco MC3810 multiservice concentrator:
Router# show voice port summary
PORT
======
0:17
0:18
0:19
0:20
0:21
0:22
0:23
1/1
1/2
1/3
1/4
1/5
1/6
CH
==
18
19
20
21
22
23
24
-------
SIG-TYPE
==========
fxo-ls
fxo-ls
fxo-ls
fxo-ls
fxo-ls
fxo-ls
e&m-imd
fxs-ls
fxs-ls
e&m-imd
e&m-imd
fxo-ls
fxo-ls
ADMIN
=====
down
up
up
up
up
up
up
up
up
up
up
up
up
OPER
====
down
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
dorm
IN
STATUS
========
idle
idle
idle
idle
idle
idle
idle
on-hook
on-hook
idle
idle
idle
idle
OUT
STATUS
========
on-hook
on-hook
on-hook
on-hook
on-hook
on-hook
idle
idle
idle
idle
idle
on-hook
on-hook
EC
==
y
y
y
y
y
y
y
y
y
y
y
y
y
show voice port Command Examples
In the following sections, output examples of the following types are shown:
•
Cisco 3600 Series Router Analog E&M Voice Port, page 99
•
Cisco 3600 Series Router Analog FXS Voice Port, page 100
•
Cisco 3600 Series Router Digital E&M Voice Port, page 101
•
Cisco AS5300 Universal Access Server T1 CAS Voice Port, page 101
•
Cisco 7200 Series Router Digital E&M Voice Port, page 102
Cisco 3600 Series Router Analog E&M Voice Port
The following output is from a Cisco 3600 series router analog E&M voice port:
Router# show voice port 1/0/0
E&M Slot is 1, Sub-unit is 0, Port is 0
Type of VoicePort is E&M
Operation State is unknown
Administrative State is unknown
The Interface Down Failure Cause is 0
Alias is NULL
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Noise Regeneration is disabled
Non Linear Processing is disabled
Music On Hold Threshold is Set to 0 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is disabled
Echo Cancel Coverage is set to 16ms
Connection Mode is Normal
Connection Number is
Initial Time Out is set to 0 s
Interdigit Time Out is set to 0 s
Analog Info Follows:
Region Tone is set for northamerica
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Voice card specific Info Follows:
Signal Type is wink-start
Operation Type is 2-wire
Impedance is set to 600r Ohm
E&M Type is unknown
Dial Type is dtmf
In Seizure is inactive
Out Seizure is inactive
Digit Duration Timing is set to 0 ms
InterDigit Duration Timing is set to 0 ms
Pulse Rate Timing is set to 0 pulses/second
InterDigit Pulse Duration Timing is set to 0 ms
Clear Wait Duration Timing is set to 0 ms
Wink Wait Duration Timing is set to 0 ms
Wink Duration Timing is set to 0 ms
Delay Start Timing is set to 0 ms
Delay Duration Timing is set to 0 ms
Cisco 3600 Series Router Analog FXS Voice Port
The following output is from a Cisco 3600 series router analog FXS voice port:
Router# show voice port 1/2
Voice port 1/2 Slot is 1, Port is 2
Type of VoicePort is FXS
Operation State is UP
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Coder Type is g729ar8
Companding Type is u-law
Voice Activity Detection is disabled
Ringing Time Out is 180 s
Wait Release Time Out is 30 s
Nominal Playout Delay is 80 milliseconds
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Verifying Analog and Digital Voice-Port Configurations
Maximum Playout Delay is 160 milliseconds
Analog Info Follows:
Region Tone is set for northamerica
Currently processing Voice
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Analog interface A-D gain offset = -3 dB
Analog interface D-A gain offset = -3 dB
Voice card specific Info Follows:
Signal Type is loopStart
Ring Frequency is 20 Hz
Hook Status is On Hook
Ring Active Status is inactive
Ring Ground Status is inactive
Tip Ground Status is active
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Ring Cadence are [20 40] * 100 msec
InterDigit Pulse Duration Timing is set to 500 ms
Cisco 3600 Series Router Digital E&M Voice Port
The following output is from a Cisco 3600 series router digital E&M voice port:
Router# show voice port 1/0:1
receEive and transMit Slot is 1, Sub-unit is 0, Port is 1
Type of VoicePort is E&M
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Region Tone is set for US
Cisco AS5300 Universal Access Server T1 CAS Voice Port
The following output is from a Cisco AS5300 universal access server T1 CAS voice port:
Router# show voice port
DS0 Group 1:0 - 1:0
Type of VoicePort is CAS
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
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Verifying Analog and Digital Voice-Port Configurations
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Playout-delay Mode is set to default
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 200 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call-Disconnect Time Out is set to 60 s
Ringing Time Out is set to 180 s
Companding Type is u-law
Region Tone is set for US
Wait Release Time Out is 30 s
Station name None, Station number None
Voice card specific Info Follows:
DS0 channel specific status info:
IN
PORT
CH SIG-TYPE
OPER STATUS
OUT
STATUS
TIP
RING
Cisco 7200 Series Router Digital E&M Voice Port
The following output is from a Cisco 7200 series router digital E&M voice port:
Router# show voice port 1/0:1
receEive and transMit Slot is 1, Sub-unit is 0, Port is 1
Type of VoicePort is E&M
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Region Tone is set for US
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Verifying Analog and Digital Voice-Port Configurations
show controller Command Examples
In the following sections, output examples of the following types are shown:
•
Cisco 3600 Series Router T1 Controller, page 103
•
Cisco MC3810 Multiservice Concentrator E1 Controller, page 103
•
Cisco AS5800 Universal Access Server T1 Controller, page 103
Cisco 3600 Series Router T1 Controller
The following output is from a Cisco 3600 series router with a T1 controller:
Router# show controller T1 1/1/0
T1 1/0/0 is up.
Applique type is Channelized T1
Cablelength is long gain36 0db
No alarms detected.
alarm-trigger is not set
Framing is ESF, Line Code is B8ZS, Clock Source is Line.
Data in current interval (180 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
Cisco MC3810 Multiservice Concentrator E1 Controller
The following output is from a Cisco MC3810 multiservice concentrator with an E1 controller:
Router# show controller E1 1/0
E1 1/0 is up.
Applique type is Channelized E1
Cablelength is short 133
Description: E1 WIC card Alpha
No alarms detected.
Framing is CRC4, Line Code is HDB3, Clock Source is Line Primary.
Data in current interval (1 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
Cisco AS5800 Universal Access Server T1 Controller
The following output is from a Cisco AS5800 universal access server with a T1 controller:
Router# show controller t1 2
T1 2 is up.
No alarms detected.
Version info of slot 0:
HW: 2, Firmware: 16, PLD Rev: 0
Manufacture Cookie Info:
EEPROM Type 0x0001, EEPROM Version 0x01, Board ID 0x42,
Board Hardware Version 1.0, Item Number 73-2217-4,
Board Revision A0, Serial Number 06467665,
PLD/ISP Version 0.0, Manufacture Date 14-Nov-1997.
Framing is ESF, Line Code is B8ZS, Clock Source is Internal.
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Configuring Voice Ports
Verifying Analog and Digital Voice-Port Configurations
Data in current interval (269 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
show voice dsp Command Examples
The following output is from a Cisco 3640 router when a digital voice port is configured:
Router# show voice dsp
TYPE DSP CH CODEC
VERS STATE STATE
RST AI PORT
TS ABORT
TX/RX-PAK-CNT
==== === == ======== ==== ===== ======= === == ======= == ===== ===============
C549 010 00 g729r8
3.3 busy idle
0 0 1/015
1
0
67400/85384
01 g729r8
.8 busy idle
0 0 1/015
7
0
67566/83623
02 g729r8
busy idle
0 0 1/015 13
0
65675/81851
03 g729r8
busy idle
0 0 1/015 20
0
65530/83610
C549 011 00 g729r8
3.3 busy idle
0 0 1/015
2
0
66820/84799
01 g729r8
.8 busy idle
0 0 1/015
8
0
59028/66946
02 g729r8
busy idle
0 0 1/015 14
0
65591/81084
03 g729r8
busy idle
0 0 1/015 21
0
66336/82739
C549 012 00 g729r8
3.3 busy idle
0 0 1/015
3
0
59036/65245
01 g729r8
.8 busy idle
0 0 1/015
9
0
65826/81950
02 g729r8
busy idle
0 0 1/015 15
0
65606/80733
03 g729r8
busy idle
0 0 1/015 22
0
65577/83532
C549 013 00 g729r8
3.3 busy idle
0 0 1/015
4
0
67655/82974
01 g729r8
.8 busy idle
0 0 1/015 10
0
65647/82088
02 g729r8
busy idle
0 0 1/015 17
0
66366/80894
03 g729r8
busy idle
0 0 1/015 23
0
66339/82628
C549 014 00 g729r8
3.3 busy idle
0 0 1/015
5
0
68439/84677
01 g729r8
.8 busy idle
0 0 1/015 11
0
65664/81737
02 g729r8
busy idle
0 0 1/015 18
0
65607/81820
03 g729r8
busy idle
0 0 1/015 24
0
65589/83889
C549 015 00 g729r8
3.3 busy idle
0 0 1/015
6
0
66889/83331
01 g729r8
.8 busy idle
0 0 1/015 12
0
65690/81700
02 g729r8
busy idle
0 0 1/015 19
0
66422/82099
03 g729r8
busy idle
0 0 1/015 25
0
65566/83852
Router# show voice dsp
TYPE DSP CH CODEC
VERS STATE STATE
RST AI PORT
TS ABORT
TX/RX-PAK-CNT
==== === == ======== ==== ===== ======= === == ======= == ===== ===============
C549 007 00 {medium} 3.3 IDLE idle
0 0 1/0:1
4
0
0/0
.13
C549 008 00 {medium} 3.3 IDLE idle
0 0 1/0:1
5
0
0/0
.13
C549 009 00 {medium} 3.3 IDLE idle
0 0 1/0:1
6
0
0/0
.13
C549 010 00 {medium} 3.3 IDLE idle
0 0 1/0:1
7
0
0/0
.13
C549 011 00 {medium} 3.3 IDLE idle
0 0 1/0:1
8
0
0/0
.13
C549 012 00 {medium} 3.3 IDLE idle
0 0 1/0:1
9
0
0/0
.13
C542 001 01 g711ulaw 3.3 IDLE idle
0 0 2/0/0
0
512/519
.13
C542 002 01 g711ulaw 3.3 IDLE idle
0 0 2/0/1
0
505/502
.13
C542 003 01 g711alaw 3.3 IDLE idle
0 0 2/1/0
0
28756/28966
.13
C542 004 01 g711ulaw 3.3 IDLE idle
0 0 2/1/1
0
834/838
.13
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Configuring Voice Ports
Verifying Analog and Digital Voice-Port Configurations
show voice call summary Command Examples
In the following sections, output examples of the following types are shown:
•
Cisco MC3810 Multiservice Concentrator Analog Voice Port
•
Cisco 3600 Series Router Digital Voice Port
Cisco MC3810 Multiservice Concentrator Analog Voice Port
The following output is from a Cisco MC3810 multiservice concentrator:
Router# show voice call summary
PORT
=========
1/1
1/2
1/3
1/4
1/5
1/6
CODEC
VAD VTSP STATE
VPM STATE
======== === ===================== ========================
g729r8
y S_CONNECT
FXSLS_CONNECT
- FXSLS_ONHOOK
- EM_ONHOOK
- EM_ONHOOK
- FXOLS_ONHOOK
- FXOLS_ONHOOK
Cisco 3600 Series Router Digital Voice Port
The following output is from a Cisco 3600 series router:
Router# show voice call summary
PORT
CODEC
========= ========
1/015.1 g729r8
1/015.2 g729r8
1/015.3 g729r8
1/015.4 g729r8
1/015.5 g729r8
1/015.6 g729r8
1/015.7 g729r8
1/015.8 g729r8
1/015.9 g729r8
1/015.10 g729r8
1/015.11 g729r8
1/015.12 g729r8
VAD VTSP STATE
VPM STATE
=== ===================== ========================
y S_CONNECT
S_TSP_CONNECT
y S_CONNECT
S_TSP_CONNECT
y S_CONNECT
S_TSP_CONNECT
y S_CONNECT
S_TSP_CONNECT
y S_CONNECT
S_TSP_CONNECT
y S_CONNECT
S_TSP_CONNECT
y S_CONNECT
S_TSP_CONNECT
y S_CONNECT
S_TSP_CONNECT
y S_CONNECT
S_TSP_CONNECT
y S_CONNECT
S_TSP_CONNECT
y S_CONNECT
S_TSP_CONNECT
y S_CONNECT
S_TSP_CONNECT
show call active voice Command Example
The following output is from a Cisco 7200 series router:
Router# show call active voice
GENERIC:
SetupTime=94523746 ms
Index=448
PeerAddress=##73072
PeerSubAddress=
PeerId=70000
PeerIfIndex=37
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Configuring Voice Ports
Verifying Analog and Digital Voice-Port Configurations
LogicalIfIndex=0
ConnectTime=94524043
DisconectTime=94546241
CallOrigin=1
ChargedUnits=0
InfoType=2
TransmitPackets=6251
TransmitBytes=125020
ReceivePackets=3300
ReceiveBytes=66000
VOIP:
ConnectionId[0x142E62FB 0x5C6705AF 0x0 0x385722B0]
RemoteIPAddress=171.68.235.18
RemoteUDPPort=16580
RoundTripDelay=29 ms
SelectedQoS=best-effort
tx_DtmfRelay=inband-voice
SessionProtocol=cisco
SessionTarget=ipv4:171.68.235.18
OnTimeRvPlayout=63690
GapFillWithSilence=0 ms
GapFillWithPrediction=180 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=70 ms
LoWaterPlayoutDelay=30 ms
ReceiveDelay=40 ms
LostPackets=0 ms
EarlyPackets=1 ms
LatePackets=18 ms
VAD = disabled
CoderTypeRate=g729r8
CodecBytes=20
cvVoIPCallHistoryIcpif=0
SignalingType=cas
show call history voice Command Example
The following output is from a Cisco 7200 series router:
Router# show call history voice
GENERIC:
SetupTime=94893250 ms
Index=450
PeerAddress=##52258
PeerSubAddress=
PeerId=50000
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Verifying Analog and Digital Voice-Port Configurations
PeerIfIndex=35
LogicalIfIndex=0
DisconnectCause=10
DisconnectText=normal call clearing.
ConnectTime=94893780
DisconectTime=95015500
CallOrigin=1
ChargedUnits=0
InfoType=2
TransmitPackets=32258
TransmitBytes=645160
ReceivePackets=20061
ReceiveBytes=401220
VOIP:
ConnectionId[0x142E62FB 0x5C6705B3 0x0 0x388F851C]
RemoteIPAddress=171.68.235.18
RemoteUDPPort=16552
RoundTripDelay=23 ms
SelectedQoS=best-effort
tx_DtmfRelay=inband-voice
SessionProtocol=cisco
SessionTarget=ipv4:171.68.235.18
OnTimeRvPlayout=398000
GapFillWithSilence=0 ms
GapFillWithPrediction=1440 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=97 ms
LoWaterPlayoutDelay=30 ms
ReceiveDelay=49 ms
LostPackets=1 ms
EarlyPackets=1 ms
LatePackets=132 ms
VAD = disabled
CoderTypeRate=g729r8
CodecBytes=20
cvVoIPCallHistoryIcpif=0
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Configuring Voice Ports
Troubleshooting Analog and Digital Voice Port Configurations
Troubleshooting Analog and Digital Voice Port Configurations
The following sections will assist in analyzing and troubleshooting voice port problems:
•
Troubleshooting Chart, page 108
•
Voice Port Testing Commands, page 110
Troubleshooting Chart
Table 10 lists some problems you might encounter after configuring voice ports and has some suggested
remedies.
Table 10
Troubleshooting Voice Port Configurations
Problem
Suggested Action
No connectivity
Ping the associated IP address to confirm connectivity. If you
cannot successfully ping your destination, refer to the Cisco IOS
IP Configuration Guide.
No connectivity
Enter the show controller t1 or show controller e1 command
with the controller number for the voice port you are
troubleshooting. This will tell you:
•
If the controller is up. If it is not, use the no shutdown
command to make it active.
•
Whether alarms have been reported.
•
What parameter values have been set for the controller
(framing, clock source, line code, cable length). If these
values do not match those of the telephony connection you
are making, reconfigure the controller.
See the “show controller Command Examples” section on
page 103 for output.
No connectivity
Enter the show voice port command with the voice port number
that you are troubleshooting, which will tell you:
•
If the voice port is up. If it is not, use the no shutdown
command to make it active.
•
What parameter values have been set for the voice port,
including default values (these do not appear in the output for
the show running-config command). If these values do not
match those of the telephony connection you are making,
reconfigure the voice port.
See the “show voice port Command Examples” section on
page 99 for sample output.
Telephony device buzzes or does
not ring
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Use the show voice port command to confirm that ring frequency
is configured correctly. It must match the connected telephony
equipment and may be country-dependent.
Configuring Voice Ports
Troubleshooting Analog and Digital Voice Port Configurations
Table 10
Troubleshooting Voice Port Configurations (continued)
Problem
Suggested Action
Distorted speech
Use the show voice port command to confirm the cptone
keyword setting (also called region tone) is US.
Setting a wrong cptone could result in faulty voice reproduction
during analog-to-digital or digital-to-analog conversions.
Music on hold is not heard
Reduce the music-threshold level.
Background noise is not heard
Enable the comfort-noise command.
Long pauses occur in
conversation; like speaking on a
walkie-talkie
Overall delay is probably excessive; the standard for adequate
voice quality is 150 ms one-way transit delay. Measure delay by
using ping tests at various times of the day with different network
traffic loads. If delay must be reduced, areas to examine include
propagation delay of signals between the sending and receiving
endpoints, voice encoding delay, and the voice packetization time
for various VoIP codecs.
Jerky or choppy speech
Variable delay, or jitter, is being introduced by congestion in the
packet network. Two possible remedies are to:
•
Reduce the amount of congestion in your packet network.
Pings between VoIP endpoints will give an idea of the
round-trip delay of a link, which should never exceed 300 ms.
Network queuing and dropped packets should also be
examined.
•
Increase the size of the jitter buffer with the playout-delay
command. (See the “Jitter Adjustment” section on page 92.)
Clipped or fuzzy speech
Reduce input gain. (See the “Voice Level Adjustment” section on
page 96.)
Clipped speech
Reduce the input level at the listener’s router. (See the “Voice
Level Adjustment” section on page 96.)
Volume too low or missed DTMF Increase speaker’s output level or listener’s input level. (See the
“Voice Level Adjustment” section on page 96.)
Echo interval is greater than 25 ms Configure the echo-cancel enable command and increase the
(sounds like a separate voice)
value for the echo-cancel coverage keyword. (See the “Echo
Adjustment” section on page 94.)
Too much echo
Reduce the output level at the speaker’s voice port. (See the
“Voice Level Adjustment” section on page 96.)
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Configuring Voice Ports
Troubleshooting Analog and Digital Voice Port Configurations
Voice Port Testing Commands
These commands allow you to force voice ports into specific states for testing. They require the use of
Cisco IOS Release 12.0(7)XK or 12.1(2)T or a later release, and they apply only to Cisco 2600 and 3600
series routers, and to Cisco MC3810 multiservice concentrators. The following types of voice-port tests
are covered:
•
Detector-Related Function Tests, page 110
•
Loopback Function Tests, page 112
•
Tone Injection Tests, page 113
•
Relay-Related Function Tests, page 114
•
Fax/Voice Mode Tests, page 114
Detector-Related Function Tests
Using the test voice port detector command, you are able to force a particular detector into an on or off
state, perform tests on the detector, and then return the detector to its original state.
To configure this feature, use the following commands in privileged EXEC mode:
Step 1
Command
Purpose
Cisco 2600 and 3600 Series Routers Analog Voice Ports
Identifies the voice port you want to test. Enter a
keyword for the detector under test and specify
whether to force it to the on or off state.
Router# test voice port slot/subunit/port detector
{m-lead | battery-reversal | loop-current | ring |
tip-ground | ring-ground | ring-trip} {on | off}
Note
Cisco 2600 and 3600 Series Routers Digital Voice Ports
Router# test voice port slot/port:ds0-group detector
{m-lead | battery-reversal | loop-current | ring |
tip-ground | ring-ground | ring-trip} {on | off}
Cisco MC3810 Multiservice Concentrators Analog Voice Ports
Router# test voice port slot/port detector {m-lead |
battery-reversal | loop-current | ring | tip-ground |
ring-ground | ring-trip} {on | off}
Cisco IOS Voice, Video, and Fax Configuration Guide
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For each signaling type (E&M, FXO,
FXS), only the applicable keywords are
displayed. The disable keyword is
displayed only when a detector is in the
forced state.
Configuring Voice Ports
Troubleshooting Analog and Digital Voice Port Configurations
Command
Cisco MC3810 Multiservice Concentrators Digital Voice Ports
Purpose
Router# test voice port slot:ds0-group detector
{m-lead | battery-reversal | loop-current | ring |
tip-ground | ring-ground | ring-trip} {on | off}
Step 2
Cisco 2600 and 3600 Series Routers Analog Voice Ports
Router# test voice port slot/subunit/port detector
{m-lead | battery-reversal | loop-current | ring |
tip-ground | ring-ground | ring-trip} disable
Cisco 2600 and 3600 Series Routers Digital Voice Ports
Router# test voice port slot/port:ds0-group detector
{m-lead | battery-reversal | loop-current | ring |
tip-ground | ring-ground | ring-trip} disable
Identifies the voice port on which you want to end
the test. Enter a keyword for the detector under
test and the keyword disable to end the forced
state.
Note
For each signaling type (E&M, FXO,
FXS), only the applicable keywords are
displayed. The disable keyword is
displayed only when a detector is in the
forced state.
Cisco MC3810 Multiservice Concentrators Analog Voice Ports
Router# test voice port slot/port detector {m-lead |
battery-reversal | loop-current | ring | tip-ground |
ring-ground | ring-trip} disable
Cisco MC3810 Multiservice Concentrators Digital Voice Ports
Router# test voice port slot:ds0-group detector
{m-lead | battery-reversal | loop-current | ring |
tip-ground | ring-ground | ring-trip} disable
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Configuring Voice Ports
Troubleshooting Analog and Digital Voice Port Configurations
Loopback Function Tests
To establish loopbacks on a voice port, use the following commands in privileged EXEC mode:
Step 1
Command
Purpose
Cisco 2600 and 3600 Series Routers Analog Voice Ports
Identifies the voice port you want to test and enters
a keyword for the loopback direction.
Router# test voice port slot/subunit/port loopback
{local | network}
Cisco 2600 and 3600 Series Routers Digital Voice Ports
Note
A call must be established on the voice
port under test.
Router# test voice port slot/port:ds0-group loopback
{local | network}
Cisco MC3810 Multiservice Concentrators Analog Voice Ports
Router# test voice port slot/port detector loopback
{local | network}
Cisco MC3810 Multiservice Concentrators Digital Voice Ports
Router# test voice port slot:ds0-group loopback {local
| network}
Step 2
Cisco 2600 and 3600 Series Routers Analog Voice Ports
Router# test voice port slot/subunit/port loopback
disable
Cisco 2600 and 3600 Series Routers Digital Voice Ports
Router# test voice port slot/port:ds0-group loopback
disable
Cisco MC3810 Multiservice Concentrators Analog Voice Ports
Router# test voice port slot/port detector loopback
disable
Cisco MC3810 Multiservice Concentrators Digital Voice Ports
Router# test voice port slot:ds0-group loopback
disable
Cisco IOS Voice, Video, and Fax Configuration Guide
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Identifies the voice port on which you want to end
the test and enters the keyword disable to end the
loopback.
Configuring Voice Ports
Troubleshooting Analog and Digital Voice Port Configurations
Tone Injection Tests
To inject a test tone into a voice port, use the following commands in privileged EXEC mode:
Step 1
Command
Purpose
Cisco 2600 and 3600 Series Routers Analog Voice Ports
Identifies the voice port you want to test and enter
keywords for the direction to send the test tone and
for the frequency of the test tone.
Router# test voice port slot/subunit/port inject-tone
{local | network} {1000hz | 2000hz | 200hz | 3000hz |
300hz | 3200hz | 3400hz | 500hz | quiet}
Note
Cisco 2600 and 3600 Series Routers Digital Voice Ports
A call must be established on the voice
port under test.
Router# test voice port slot/port:ds0-group
inject-tone {local | network} {1000hz | 2000hz | 200hz
| 3000hz | 300hz | 3200hz | 3400hz | 500hz | quiet}
Cisco MC3810 Multiservice Concentrators Analog Voice Ports
Router# test voice port slot/port detector inject-tone
{local | network} {1000hz | 2000hz | 200hz | 3000hz |
300hz | 3200hz | 3400hz | 500hz | quiet}
Cisco MC3810 Multiservice Concentrators Digital Voice Ports
Router# test voice port slot:ds0-group inject-tone
{local | network} {1000hz | 2000hz | 200hz | 3000hz |
300hz | 3200hz | 3400hz | 500hz | quiet}
Step 2
Cisco 2600 and 3600 Series Routers Analog Voice Ports
Identifies the voice port on which you want to end
the test and enter the keyword disable to end the
test tone.
Router# test voice port slot/subunit/port inject-tone
disable
Note
Cisco 2600 and 3600 Series Routers Digital Voice Ports
The disable keyword is available only if a
test condition is already activated.
Router# test voice port slot/port:ds0-group
inject-tone disable
Cisco MC3810 Multiservice Concentrators Analog Voice Ports
Router# test voice port slot/port detector inject-tone
disable
Cisco MC3810 Multiservice Concentrators Digital Voice Ports
Router# test voice port slot:ds0-group inject-tone
disable
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Configuring Voice Ports
Troubleshooting Analog and Digital Voice Port Configurations
Relay-Related Function Tests
To test relay-related functions on a voice port, use the following commands in privileged EXEC mode:
Step 1
Command
Purpose
Cisco 2600 and 3600 Series Routers Analog Voice Ports
Identifies the voice port you want to test. Enter a
keyword for the relay under test and specify
whether to force it to the on or off state.
Router# test voice port slot/subunit/port relay
{e-lead | loop | ring-ground | battery-reversal |
power-denial | ring | tip-ground} {on | off}
Note
Cisco 2600 and 3600 Series Routers Digital Voice Ports
Router# test voice port slot/port:ds0-group relay
{e-lead | loop | ring-ground | battery-reversal |
power-denial | ring | tip-ground} {on | off}
For each signaling type (E&M, FXO,
FXS), only the applicable keywords are
displayed. The disable keyword is
displayed only when a relay is in the
forced state.
Cisco MC3810 Multiservice Concentrators Analog Voice Ports
Router# test voice port slot/port detector relay
{e-lead | loop | ring-ground | battery-reversal |
power-denial | ring | tip-ground} {on | off}
Cisco MC3810 Multiservice Concentrators Digital Voice Ports
Router# test voice port slot:ds0-group relay {e-lead |
loop | ring-ground | battery-reversal | power-denial |
ring | tip-ground} {on | off}
Step 2
Cisco 2600 and 3600 Series Routers Analog Voice Ports
Router# test voice port slot/subunit/port relay
{e-lead | loop | ring-ground | battery-reversal |
power-denial | ring | tip-ground} disable
Identifies the voice port on which you want to end
the test. Enter a keyword for the relay under test,
and the keyword disable to end the forced state.
Note
Cisco 2600 and 3600 Series Routers Digital Voice Ports
Router# test voice port slot/port:ds0-group relay
{e-lead | loop | ring-ground | battery-reversal |
power-denial | ring | tip-ground} disable
For each signaling type (E&M, FXO,
FXS), only the applicable keywords are
displayed. The disable keyword is
displayed only when a relay is in the
forced state.
Cisco MC3810 Multiservice Concentrators Analog Voice Ports
Router# test voice port slot/port detector relay
{e-lead | loop | ring-ground | battery-reversal |
power-denial | ring | tip-ground} disable
Cisco MC3810 Multiservice Concentrators Digital Voice Ports
Router# test voice port slot:ds0-group relay {e-lead |
loop | ring-ground | battery-reversal | power-denial |
ring | tip-ground} disable
Fax/Voice Mode Tests
The test voice port switch fax command forces a voice port into fax mode for testing. After you enter
this command, you can use the show voice call or show voice call summary command to check whether
the voice port is able to operate in fax mode. If no fax data is detected by the voice port, the voice port
remains in fax mode for 30 seconds and then reverts automatically to voice mode.
The disable keyword ends the forced mode switch; however, the fax mode ends automatically after
30 seconds. The disable keyword is available only while the voice port is in fax mode.
Cisco IOS Voice, Video, and Fax Configuration Guide
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Configuring Voice Ports
Troubleshooting Analog and Digital Voice Port Configurations
To force a voice port into fax mode and return it to voice mode, use the following commands in
privileged EXEC mode:
Step 1
Command
Cisco 2600 and 3600 Series Routers Analog Voice Ports
Purpose
Identifies the voice port you want to test. Enter the
keyword fax to force the voice port into fax mode.
Router# test voice port slot/subunit/port switch fax
Cisco 2600 and 3600 Series Routers Digital Voice Ports
Router# test voice port slot/port:ds0-group switch fax
Cisco MC3810 Multiservice Concentrators Analog Voice Ports
Router# test voice port slot/port detector switch fax
Cisco MC3810 Multiservice Concentrators Digital Voice Ports
Router# test voice port slot:ds0-group switch fax
Step 2
Cisco 2600 and 3600 Series Routers Analog Voice Ports
Router# test voice port slot/subunit/port switch
disable
Identifies the voice port on which you want to end
the test. Enter the keyword disable to return the
voice port to voice mode.
Cisco 2600 and 3600 Series Routers Digital Voice Ports
Router# test voice port slot/port:ds0-group switch
disable
Cisco MC3810 Multiservice Concentrators Analog Voice Ports
Router# test voice port slot/port detector switch
disable
Cisco MC3810 Multiservice Concentrators Digital Voice Ports
Router# test voice port slot:ds0-group switch disable
Cisco IOS Voice, Video, and Fax Configuration Guide
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Configuring Voice Ports
Troubleshooting Analog and Digital Voice Port Configurations
Cisco IOS Voice, Video, and Fax Configuration Guide
VC-116
Configuring Dial Plans, Dial Peers, and Digit
Manipulation
This chapter describes how to implement dial plans by configuring dial peers and using dial peer
matching and digit manipulation features. This chapter contains the following sections:
•
Dial Plan Overview, page 117
•
Configuring Dial Peers, page 124
•
Dial Peer Overview, page 137
•
Configuring Dial Peer Matching Features, page 141
•
Configuring Digit Manipulation, page 151
For a complete description of the commands used in this chapter, refer to the Cisco IOS Voice, Video,
and Fax Command Reference. To locate documentation of other commands that appear in this chapter,
use the command reference master index or search online.
To identify the hardware platform or software image information associated with a feature in this
chapter, use the Feature Navigator on Cisco.com to search for information about the feature or refer to
the software release notes for a specific release. For more information, see the “Identifying Supported
Platforms” section in the “Using Cisco IOS Software” chapter.
Dial Plan Overview
A dial plan essentially describes the number and pattern of digits that a user dials to reach a particular
telephone number. Access codes, area codes, specialized codes, and combinations of the number of
digits dialed are all part of a dial plan. For instance, the North American Public Switched Telephone
Network (PSTN) uses a 10-digit dial plan that includes a 3-digit area code and a 7-digit telephone
number. Most PBXs support variable length dial plans that use 3 to 11 digits. Dial plans must comply
with the telephone networks to which they connect. Only totally private voice networks that are not
linked to the PSTN or to other PBXs can use any dial plan they choose.
Dial plans on Cisco routers are manually defined using dial peers. Dial peers are similar to static routes;
they define where calls originate and terminate and what path the calls take through the network.
Attributes within the dial peer determine which dialed digits the router collects and forwards to
telephony devices.
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Note
If you are using Media Gateway Control Protocol (MGCP) or Simple Gateway Control Protocol
(SGCP) on your call agent, you do not need to configure static dial peers. See the chapter
“Configuring MGCP and Related Protocols” for more information.
The following sections provide an overview of basic dial peer concepts:
Note
•
Dial Peer Overview, page 118
•
Inbound and Outbound Dial Peers, page 119
•
Destination Pattern, page 120
•
Fixed- and Variable-Length Dial Plans, page 122
•
Session Target, page 123
•
Digit Stripping on Outbound POTS Dial Peers, page 124
The illustrations and sample configurations in this section use VoIP; the same concepts also apply to
Voice over Frame Relay (VoFR) and Voice over ATM (VoATM) networks.
Dial Peer Overview
Configuring dial peers is the key to setting up dial plans and implementing voice over a packet network.
Dial peers are used to identify call source and destination endpoints and to define the characteristics
applied to each call leg in the call connection.
A traditional voice call over the PSTN uses a dedicated 64K circuit end to end. In contrast, a voice call
over the packet network is made up of discrete segments or call legs. A call leg is a logical connection
between two routers or between a router and a telephony device. A voice call comprises four call legs,
two from the perspective of the originating router and two from the perspective of the terminating router,
as shown in Figure 22.
Dial Peer Call Legs
Source
Destination
IP network
V
Call leg 1
(POTS dial peer)
Call leg 2
(VoIP dial peer)
Call leg 3
(VoIP dial peer)
V
35950
Figure 22
Call leg 4
(POTS dial peer)
A dial peer is associated with each call leg. Attributes that are defined in a dial peer and applied to the
call leg include codec, Quality of Service (QoS), voice activity detection (VAD), and fax rate. To
complete a voice call, you must configure a dial peer for each of the four call legs in the call connection.
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Depending on the call leg, a call is routed using one of the two types of dial peers:
•
POTS—Dial peer that defines the characteristics of a traditional telephony network connection.
POTS dial peers map a dialed string to a specific voice port on the local router, normally the voice
port connecting the router to the local PSTN, PBX, or telephone.
•
Voice-network—Dial peer that defines the characteristics of a packet network connection.
Voice-network dial peers map a dialed string to a remote network device, such as the destination
router that is connected to the remote telephony device.
The specific type of voice-network dial peer depends on the packet network technology:
– VoIP (Voice over IP)—Points to the IP address of the destination router that terminates the call.
– VoFR (Voice over Frame Relay)—Points to the data-link connection identifier (DLCI) of the
interface from which the call exits the router.
– VoATM (Voice over ATM)—Points to the ATM virtual circuit for the interface from which the
call exits the router.
– MMoIP (Multimedia Mail over IP)—Points to the e-mail address of the SMTP server. This type
of dial peer is used only for fax traffic. For more information, see the chapter “Configuring Fax
Applications.”
Both POTS and voice-network dial peers are needed to establish voice connections over a packet
network.
Inbound and Outbound Dial Peers
Dial peers are used for both inbound and outbound call legs. It is important to remember that these terms
are defined from the perspective of the router. An inbound call leg originates when an incoming call
comes to the router. An outbound call leg originates when an outgoing call is placed from the router.
Figure 23 illustrates call legs from the perspective of the originating router; Figure 24 illustrates call legs
from the perspective of the terminating router.
Figure 23 and Figure 24 apply to voice calls that are being sent across the packet network. If the
originating and terminating POTS interfaces share the same router or if the call requires hairpinning,
then two POTS call legs are sufficient. See Figure 29 on page 126 for more information.
Figure 23
Call Legs from the Perspective of the Originating Router
Source
Destination
IP network
V
Inbound
POTS call leg
V
Outbound
VoIP call leg
35946
Note
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Figure 24
Call Legs from the Perspective of the Terminating Router
Source
Destination
IP network
V
Inbound
VoIP call leg
Outbound
POTS call leg
36849
V
For inbound calls from a POTS interface that are destined for the packet network, the router matches a
POTS dial peer for the inbound call leg and a voice-network dial peer, such as VoIP or VoFR, for the
outbound leg. For inbound calls from the packet network, the router matches a POTS dial peer to
terminate the call and a voice-network dial peer to apply features such as codec or QoS.
For inbound POTS call legs going to outbound voice-network dial peers, the router forwards all digits
that it collects. On outbound POTS call legs, the router strips off explicitly matching digits and forwards
any excess digits out the designated port. For specific information about how the router handles excess
digits, see the “Two-Stage Dialing” section on page 137.
The following examples show basic configurations for POTS and VoIP dial peers:
dial-peer voice 1 pots
destination-pattern 555....
port 1/0:1
dial-peer voice 2 voip
destination-pattern 555....
session target ipv4:192.168.1.1
The router selects a dial peer for a call leg by matching the string that is defined by using the
answer-address, destination-pattern, or incoming called-number command in the dial peer
configuration. For specific information about how the router matches dial peers, see the “Dial Peer
Overview” section on page 137.
Destination Pattern
The destination pattern associates a dialed string with a specific telephony device. It is configured in a
dial peer by using the destination-pattern command. If the dialed string matches the destination pattern,
the call is routed according to the voice port in POTS dial peers, or the session target in voice-network
dial peers. For outbound voice-network dial peers, the destination pattern may also determine the dialed
digits that the router collects and then forwards to the remote telephony interface, such as a PBX, a
telephone, or the PSTN. You must configure a destination pattern for each POTS and voice-network dial
peer that you define on the router.
The destination pattern can be either a complete telephone number or a partial telephone number with
wildcard digits, represented by a period (.) character. Each “.” represents a wildcard for an individual
digit that the originating router expects to match. For example, if the destination pattern for a dial peer
is defined as “555....”, then any dialed string beginning with 555, plus at least four additional digits,
matches this dial peer.
In addition to the period (.), there are several other symbols that can be used as wildcard characters in
the destination pattern. These symbols provide additional flexibility in implementing dial plans and
decrease the need for multiple dial peers in configuring telephone number ranges.
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Table 11 shows the wildcard characters that are supported in the destination pattern.
Table 11
Wildcard Symbols Used in Destination Patterns
Symbol
Description
.
Indicates a single-digit placeholder. For example, 555.... matches any dialed string
beginning with 555, plus at least four additional digits.
[]
Indicates a range of digits. A consecutive range is indicated with a hyphen (-); for
example, [5-7]. A nonconsecutive range is indicated with a comma (,); for example, [5,8].
Hyphens and commas can be used in combination; for example, [5-7,9].
Note
Note
Only single-digit ranges are supported. For example, [98-102] is invalid.
()
Indicates a pattern; for example, 408(555). It is used in conjunction with the symbol ?, %,
or +.
?
Indicates that the preceding digit occurred zero or one time. Enter ctrl-v before entering
? from your keyboard.
%
Indicates that the preceding digit occurred zero or more times. This functions the same as
the “*” used in regular expression.
+
Indicates that the preceding digit occurred one or more times.
T
Indicates the interdigit timeout. The router pauses to collect additional dialed digits.
The period (.) is the only wildcard character that is supported for dial strings that are configured using
the answer-address or incoming called-number commands.
Table 12 shows some examples of how these wildcard symbols are applied to the destination pattern and
the dial string that results when dial string 4085551234 is matched to an outbound POTS dial peer. The
wildcard symbols follow regular expression rules.
Table 12
Dial Peer Matching Examples Using Wildcard Symbols
Destination Pattern Dial String Translation
String After Stripping1
1
408555.+
408555, followed by one or more wildcard digits. 1234
This pattern implies that the string must contain at
least 7 digits starting with 408555.
2
408555.%
408555, followed by zero or more wildcard digits. 1234
This pattern implies that the string must contain at
least 408555.
3
408555+
40855, followed by 5 repeated one or more times. 1234
4
408555%
40855, followed by 5 repeated zero or more times. 51234
Any explicitly matching digit before the % symbol
is not stripped off.
5
408555?
40855, followed by 5 repeated zero or one time.
51234
Any explicitly matching digit before the ? symbol
is not stripped off.
6
40855[5-7].+
40855, followed by 5, 6, or 7, plus any digit
repeated one or more times.
51234
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Table 12
Dial Peer Matching Examples Using Wildcard Symbols (continued)
Destination Pattern Dial String Translation
String After Stripping1
7
40855[5-7].%
40855, followed by 5, 6, or 7, plus any digit
repeated zero or more times.
51234
8
40855[5-7]+1234
40855, followed by 5, 6, or 7 repeated one or more 51234
times, followed by 1234.
9
408(555)+1234
408, followed by 555, which may repeat one or
more times, followed by 1234.
5551234
1. These examples apply only to one-stage dialing, where DID is enabled on the inbound POTS dial peer. If the router is using
two-stage dialing and collecting digits one at a time as dialed, then the call is routed immediately after a dial peer is matched
and any subsequent dialed digits are lost.
In addition to wildcard characters, the following characters can also be used in the destination pattern:
•
Asterisk (*) and pound sign (#)—These characters on standard touch-tone dial pads can be used
anywhere in the pattern. They can be used as the leading character (for example, *650), except on
the Cisco 3600 series.
•
Dollar sign ($)—Disables variable-length matching. Must be used at the end of the dial string.
The same destination pattern can be shared across multiple dial peers to form hunt groups. For
information on building hunt groups, see the “Hunt Groups and Preferences” section on page 144.
For information on how the terminating router strips off digits after matching a destination pattern, see
the “Digit Stripping on Outbound POTS Dial Peers” section on page 124.
Fixed- and Variable-Length Dial Plans
Fixed-length dialing plans, in which all the dial-peer destination patterns have a fixed length, are
sufficient for most voice networks because the telephone number strings are of known lengths. Some
voice networks, however, require variable-length dial plans, particularly for international calls, which
use telephone numbers of different lengths.
If you enter the timeout T-indicator at the end of the destination pattern in an outbound voice-network
dial peer, the router accepts a fixed-length dial string and then waits for additional dialed digits. The
timeout character must be an uppercase T. The following dial-peer configuration shows how the
T-indicator is set to allow variable-length dial strings:
dial-peer voice 1 voip
destination-pattern 2222T
session target ipv4:10.10.1.1
In the example above, the router accepts the digits 2222, and then waits for an unspecified number of
additional digits. The router can collect up to 31 additional digits, as long as the interdigit timeout has
not expired. When the interdigit timeout expires, the router places the call.
The default value for the interdigit timeout is 10 seconds. Unless the default value is changed, using the
T-indicator adds 10 seconds to each call setup because the call is not attempted until the timer has expired
(unless the # character is used as a terminator). You should therefore reduce the voice-port interdigit
timeout value if you use variable-length dial plans. You can change the interdigit timeout by using the
timeouts inter-digit voice-port command.
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The calling party can immediately terminate the interdigit timeout by entering the # character. If the #
character is entered while the router is waiting for additional digits, the # character is treated as a
terminator; it is not treated as part of the dial string or sent across the network. But if the # character is
entered before the router begins waiting for additional digits (meaning that the # is entered as part of the
fixed-length destination pattern), then the # character is treated as a dialed digit.
For example, if the destination pattern is configured as 2222...T, then the entire dialed string of
2222#9999 is collected, but if the dialed string is 2222#99#99, the #99 at the end of the dialed digits is
not collected because the final # character is treated as a terminator. You can change the termination
character by using the dial-peer terminator command.
Note
In most cases, you must configure the T-indicator only when the router uses two-stage dialing. If
Direct Inward Dialing (DID) is configured in the inbound POTS dial peer, the router uses one-stage
dialing, which means that the full dialed string is used to match outbound dial peers. The only
exception is when the ISDN overlap-receiving command is configured; the ISDN overlap-receiving
feature requires the T-indicator.
Session Target
The session target is the network address of the remote router to which you want to send a call once a
local voice-network dial peer is matched. It is configured in voice-network dial peers by using the
session target command. For outbound dial peers, the destination pattern is the telephone number of the
remote voice device that you want to reach. The session target represents the path to the remote router
that is connected to that voice device. Figure 25 illustrates the relationship between the destination
pattern and the session target, as shown from the perspective of the originating router.
Figure 25
Relationship Between Destination Pattern and Session Target
Source
Destination
VoIP dial peer
session target
VoIP dial peer
destination pattern
579…
389…
Serial port
Voice port
37557
IP network
PBX
The address format of the session target depends on the type of voice-network dial peer:
Note
•
VoIP—IP address, hostname of the Domain Name System (DNS) server that resolves the IP address,
ras for registration, admission, and status (RAS) if an H.323 gatekeeper resolves the IP address, or
settlement if the settlement server resolves the IP address
•
VoFR—Interface type and number and the DLCI
•
VoATM—Interface number, and ATM virtual circuit
•
MMoIP—E-mail address
For inbound dial peers, the session target is ignored.
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Digit Stripping on Outbound POTS Dial Peers
When a terminating router receives a voice call, it selects an outbound POTS dial peer by comparing the
called number (the full E.164 telephone number) in the call information with the number configured as
the destination pattern in the POTS dial peer. The access server or router then strips off the left-justified
digits that match the destination pattern. If you have configured a prefix, the prefix is added to the front
of the remaining digits, creating a dial string, which the router then dials. If all numbers in the destination
pattern are stripped out, the user receives a dial tone.
For example, consider a voice call whose E.164 called number is 1(408) 555-2222. If you configure a
destination-pattern of “1408555” and a prefix of “9,” the router strips off “1408555” from the E.164
telephone number, leaving the extension number of “2222.” It then appends the prefix, “9,” to the front
of the remaining numbers, so that the actual numbers dialed are “9, 2222.” The comma in this example
means that the router will pause for one second between dialing the “9” and dialing the “2” to allow for
a secondary dial tone.
For detailed information about digit stripping and the prefix command, see the “Digit Stripping and
Prefixes” section on page 151.
Configuring Dial Peers
This section describes how to configure dial peers:
Note
•
Configuring Dial Peers for Call Legs, page 125
•
Creating a Dial Peer Configuration Table, page 127
•
Configuring POTS Dial Peers, page 128
•
Configuring Dial Plan Options for POTS Dial Peers, page 130
•
Configuring VoIP Dial Peers, page 131
•
Configuring Dial Plan Options for VoIP Dial Peers, page 133
•
Configuring VoFR Dial Peers, page 135
•
Configuring VoATM Dial Peers, page 135
The example configurations in this section show VoIP dial peers; the same concepts also apply to
VoFR and VoATM dial peers.
Establishing voice communication over a packet network is similar to configuring a static route: you are
establishing a specific voice connection between two defined endpoints. Call legs define the discrete
segments that lie between two points in the call connection. A voice call over the packet network
comprises four call legs, two on the originating router and two on the terminating router; a dial peer is
associated with each of these four call legs.
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Configuring Dial Peers for Call Legs
When a voice call comes into the router, the router must match dial peers to route the call. For inbound
calls from a POTS interface that are being sent over the packet network, the router matches a POTS dial
peer for the inbound call leg and a voice-network dial peer for the outbound call leg. For calls coming
into the router from the packet network, the router matches an outbound POTS dial peer to terminate the
call and an inbound voice-network dial peer for features such as codec, VAD, and QoS.
Figure 26 shows the call legs and associated dial peers necessary to complete a voice call.
Figure 26
Matching Call Legs to Dial Peers
Source
Destination
IP network
Inbound call leg
(POTS dial peer)
Outbound call leg
(VoIP dial peer)
V
Inbound call leg
(VoIP dial peer)
37207
V
Outbound call leg
(POTS dial peer)
The following configurations show an example of a call being made from 4085554000 to 3105551000.
Figure 27 shows the inbound POTS dial peer and the outbound VoIP dial peer that are configured on the
originating router. The POTS dial peer establishes the source of the call (via the calling number or voice
port), and the voice-network dial peer establishes the destination by associating the dialed number with
the network address of the remote router.
Dial Peers from the Perspective of the Originating Router
Source
Destination
Router A
1/0/0
4085554000
Router B
10.1.1.1
V
IP network
35965
Figure 27
1/0/0
10.1.1.2
V
3105551000
dial-peer voice 1 pots
destination-pattern 1408555 . . . .
port 1/0/0
dial-peer voice 2 voip
destination-pattern 1310555 . . . .
session target ipv4:10.1.1.2
In this example, the dial string 14085554000 maps to telephone number 555-4000, with the digit 1 plus
the area code 408 preceding the number. When you configure the destination pattern, set the string to
match the local dialing conventions.
Figure 28 shows the inbound VoIP dial peer and outbound POTS dial peer that are configured on the
terminating router to complete the call. Dial peers are of local significance only.
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Figure 28
Dial Peers from the Perspective of the Terminating Router
Source
Destination
Router A
1/0/0
Router B
10.1.1.1
IP network
V
1/0/0
10.1.1.2
V
4085554000
3105551000
dial-peer voice 2 voip
destination-pattern 1408555 . . . .
session target ipv4:10.1.1.1
35966
dial-peer voice 1 pots
destination-pattern 1310555 . . . .
port 1/0/0
In the previous configuration examples, the last four digits in the VoIP dial peer’s destination pattern
were replaced with wildcards. This means that from Router A, calling any telephone number that begins
with the digits “1310555” will result in a connection to Router B. This implies that Router B services all
numbers beginning with those digits. From Router B, calling any telephone number that begins with the
digits “1408555” will result in a connection to Router A. This implies that Router A services all numbers
beginning with those digits.
Note
It is not always necessary to configure the inbound dial peers. If the router is unable to match a
configured dial peer for the inbound call leg, it uses an internally defined default POTS or
voice-network dial peer to match inbound voice calls. In the example shown in Figure 28, dial peer
2 is only required when making a call from Router B to Router A.
The only exception to the previous example occurs when both POTS dial peers share the same router, as
shown in Figure 29. In this circumstance, you do not need to configure a voice-network dial peer.
Figure 29
Communication Between Dial Peers Sharing the Same Router
dial-peer voice 3 pots
destination-pattern 4001
port 1/0/0
Source
4001
1/0/0
1/1/0
Destination
4000
dial-peer voice 1 pots
destination-pattern 4000
port 1/1/0
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This type of configuration is similar to the configuration used for hairpinning, which occurs when a
voice call destined for the packet network is instead routed back over the PSTN because the packet
network is unavailable. For more information about the hairpinning feature, see the “Hunt Groups and
Preferences” section on page 144.
Creating a Dial Peer Configuration Table
Before you can configure dial peers, you must obtain specific information about your network. One way
to identify this information is to create a dial peer configuration table. This table should contain all the
telephone numbers and access codes for each router that is carrying telephone traffic in the network.
Because most installations require integrating equipment into an existing voice network, the telephone
dial plans are usually preset.
Figure 30 shows an example of a network in which Router A, with an IP address of 10.1.1.1, connects a
small sales branch office to the main office through Router B, with an IP address of 10.1.1.2.
The example in Figure 30 shows a VoIP configuration. The same concepts also apply to VoFR and
VoATM applications. The only change is in the format of the session target.
Figure 30
Sample VoIP Network
729 555-1001
729 555-1002
408 115-1001
729 555-1000
408 116-1002
0:D
1:D
0:D
Router A
V
729 555-1003
WAN
10.1.1.1
IP network
WAN
10.1.1.2
V
Router B
36850
Note
408 117-1003
There are three telephone numbers in the sales branch office that need dial peers configured for them.
Router B is the primary gateway to the main office; as such, it needs to be connected to the company’s
PBX. There are four devices that need dial peers configured for them in the main office, all of which are
connected to the PBX.
Table 13 shows the peer configuration table for the example in Figure 30.
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Table 13
Dial Peer
Dial Peer Configuration Table for Sample Voice over IP Network
Extension
Prefix
Destination Pattern Type
Voice Port
Session Target
1
51001
5
1408115....
POTS
0:D
—
2
61002
6
1408116....
POTS
0:D
—
3
71003
7
1408117....
POTS
0:D
—
10
—
—
1729555....
VoIP
—
10.1.1.2
1
1000,
1001,
1002,
1003
—
1729555....
POTS
0:D
—
10
—
—
1408.......
VoIP
—
10.1.1.1
Router A
Router B
Configuring POTS Dial Peers
To configure a POTS dial peer, you must do the following:
•
Identify the dial peer by assigning it a unique tag number
•
Define its destination telephone number or range of telephone numbers
•
Associate it with a voice port through which calls are established
Under most circumstances, the default values for the remaining dial peer configuration commands are
sufficient to establish connections.
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To configure a POTS dial peer, use the following commands beginning in global configuration mode:
Step 1
Command
Purpose
Router(config)# dial-peer voice number pots
Enters dial-peer configuration mode and defines a
local dial peer that connects to a POTS interface.
The number argument is one or more digits
identifying the dial peer. Valid entries are from 1 to
2147483647.
The pots keyword indicates a dial peer using basic
telephone service.
Step 2
Router(config-dial-peer)# destination-pattern
string [T]
Matches dialed digits to a telephony device.
The string argument is a series of digits that specify
the E.164 or private dialing plan telephone number.
Valid entries are the numbers 0 through 9 and the
letters A through D.
You can also enter the following special characters:
•
The asterisk (*) or pound sign (#) on standard
touch-tone dial pads can be used anywhere in the
pattern.
•
The period (.) acts as a wildcard character.
For a list of additional wildcard characters, see
Table 11 on page 121.
When the timer (T) character is included at the end of
the destination pattern, the router collects dialed
digits until the interdigit timer expires (10 seconds,
by default) or until you dial the termination character
(the default is #). The timer character must be a
capital T.
Step 3
Router(config-dial-peer)# port location
Maps the dial peer to a specific logical interface.
The port command syntax is platform-specific. For
more information about the syntax of this command,
see the chapter “Configuring Voice Ports” in this
document.
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Configuring Dial Plan Options for POTS Dial Peers
When you configure a dial plan, you have different options, depending on how the dial plan is designed.
To configure optional dial plan features for POTS dial peers, use one or more of the following commands
in dial-peer configuration mode:
Command
Purpose
Router(config-dial-peer)# answer-address string
(Optional) Selects the inbound dial peer based on the
calling number.
Router(config-dial-peer)# incoming called-number string
(Optional) Selects the inbound dial peer based on the
called number to identify voice and modem calls.
Router(config-dial-peer)# direct-inward-dial string
(Optional) Enables the Direct Inward Dialing (DID) call
treatment for the incoming called number. For more
information, see the “DID for POTS Dial Peers” section
on page 142.
Router(config-dial-peer)# forward-digits {num-digit |
all | extra}
(Optional) Configures the digit-forwarding method used
by the dial peer. The valid range for the number of digits
forwarded (num-digit) is 0 through 32. For more
information, see the “Forward Digits” section on
page 154.
Router(config-dial-peer)# max-conn number
(Optional) Specifies the maximum number of allowed
connections to and from the POTS dial peer. The valid
range is 1 through 2147483647.
Router(config-dial-peer)# numbering-type {abbreviated |
international | national | network | reserved |
subscriber | unknown}
(Optional) Specifies the numbering type to match, as
defined by the ITU Q.931 specification. For more
information, see the “Numbering Type Matching” section
on page 147.
Router(config-dial-peer)# preference value
(Optional) Configures a preference for the POTS dial peer.
The valid range is 0 through 10, where the lower the
number, the higher the preference. For more information,
see the “Hunt Groups and Preferences” section on
page 144.
Router(config-dial-peer)# prefix string
(Optional) Includes a prefix that the system adds
automatically to the front of the dial string before passing
it to the telephony interface.
Valid entries for the string argument are 0 through 9 and a
comma (,). Use a comma to include a one-second pause
between digits to allow for a secondary dial tone.
For more information, see the “Digit Stripping and
Prefixes” section on page 151.
Router(config-dial-peer)# translate-outgoing {called |
calling} name-tag
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(Optional) Specifies the translation rule set to apply to the
calling number or called number. For more information,
see the “Digit Translation Rules for VoIP” section on
page 157.
Configuring Dial Plans, Dial Peers, and Digit Manipulation
Configuring Dial Peers
Configuring VoIP Dial Peers
VoIP dial peers enable the router to make outbound calls to a particular telephony device. To configure
a VoIP dial peer, you must do the following:
•
Identify the dial peer by assigning it a unique tag number
•
Define its destination telephone number
•
Define its destination IP address
As with POTS dial peers, under most circumstances the default values for the remaining dial peer
configuration commands are adequate to establish connections.
To configure a VoIP peer, use the following commands beginning in global configuration mode:
Step 1
Command
Purpose
Router(config)# dial-peer voice number voip
Enters dial-peer configuration mode and defines a
remote VoIP dial peer.
The number argument is one or more digits
identifying the dial peer. Valid entries are from 1 to
2147483647.
The voip keyword indicates a dial peer using voice
encapsulation on the IP network.
Step 2
Router(config-dial-peer)# destination-pattern
string [T]
Configures the dial peer’s destination pattern so that
the system can reconcile dialed digits with a
telephone number.
The string argument is a series of digits that specify
the E.164 or private dialing plan telephone number.
Valid entries are the numbers 0 through 9 and the
letters A through D. You can also enter the following
special characters:
•
The asterisk (*) or pound sign (#) on standard
touch-tone dial pads can be used anywhere in the
pattern.
•
The period (.) acts as a wildcard character.
For a list of additional wildcard characters, see
Table 11 on page 121.
When the timer (T) character is included at the end of
the destination pattern, the router collects dialed
digits until the interdigit timer expires (10 seconds,
by default) or until you dial the termination character
(the default is #). The timer character must be a
capital T.
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Step 3
Command
Purpose
Router(config-dial-peer)# session target
{ipv4:destination-address | dns:[$s$. | $d$. | $e$.
| $u$.] host-name}
Defines the IP address of the router that is connected
to the remote telephony device.
The ipv4:destination-address keyword and argument
indicate the IP address of the remote router.
The dns:host-name keyword and argument indicate
that the domain name server will resolve the name of
the IP address. Valid entries for this parameter are
characters representing the name of the host device.
Wildcards are also available for defining domain
names with the keyword by using source, destination,
and dialed information in the host name.
Step 4
Router(config-dialpeer)# codec {g711alaw | g711ulaw
| g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16
| g726r24 | g726r32 | g728 | g729br8 | g729r8
[pre-ietf]} [bytes]
Defines the codec for the dial peer.
The optional bytes parameter sets the number of voice
data bytes per frame. Acceptable values are from
10 to 240 in increments of 10 (for example, 10, 20,
30, and so on). Any other value is rounded down (for
example, from 236 to 230).
The same codec value must be configured in both
VoIP dial peers on either side of the connection.
If you specify g729r8, then IETF bit-ordering is used.
For interoperability with a Cisco 2600 series,
Cisco 3600 series, or Cisco AS5300 running a release
earlier than Cisco IOS Release 12.0(5)T or
12.0(4)XH, you must specify the additional keyword
pre-ietf after g729r8.
The codec command syntax is platform- and
release-specific. For more information about the
syntax of this command, refer to the Cisco IOS Voice,
Video, and Fax Command Reference.
If you have used the codec complexity voice-card interface configuration command, the codec
command sets the codec options that are available. If you do not set codec complexity, g729r8 with IETF
bit-ordering is used. For more information about the codec complexity command, see the “Configuring
Voice Ports” chapter.
Configuring Codec Selection Order
You can create a voice class in which you define a selection order for codecs, and then apply the voice
class to VoIP dial peers. The voice class codec global configuration command allows you to define the
voice class containing the codec selection order. Then you use the voice-class codec dial-peer
configuration command to apply the class to individual dial peers.
To configure codec selection order, perform the tasks described in the following sections:
•
Creating a Voice Class to Define Codec Selection Order
•
Applying Codec Selection Order to a VoIP Dial Peer
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Creating a Voice Class to Define Codec Selection Order
To create a voice class to define the order of preference for selecting a codec when the router negotiates
with a destination router, use the following commands beginning in global configuration mode:
Command
Purpose
Step 1
router(config)# voice class codec tag
Creates a voice class for a codec preference list. The
range for the tag number is from 1 through 10000.
The tag number must be unique on the router.
Step 2
router(config-voice-class)# codec preference
priority codec [ bytes payload-size]
Configures the order of preference for selecting a
codec. Repeat this command to specify the preferred
selection order for additional codecs, if required.
Applying Codec Selection Order to a VoIP Dial Peer
To apply voice-class codec attributes to a VoIP dial peer, use the following commands beginning in
global configuration mode:
Step 1
Command
Purpose
router(config)# dial-peer voice tag voip
Defines a VoIP dial peer and enters dial-peer
configuration mode.
The tag is a number that identifies the dial peer and
must be unique on the router.
Step 2
router(config-dialpeer)# voice-class codec tag
Assigns to the dial peer the voice class that you
created in the “Creating a Voice Class to Define
Codec Selection Order” section.
The voice-class command in dial-peer configuration
mode is entered with a hyphen. The voice class
command in global configuration mode is entered
without the hyphen.
Note
You cannot assign voice-class codec attributes to POTS dial peers.
Configuring Dial Plan Options for VoIP Dial Peers
When you configure a dial plan, you have different options, depending on how the dial plan is designed.
To configure optional dial plan features, use the following commands in dial-peer configuration mode:
Command
Purpose
Step 1
Router(config-dial-peer)# answer-address string
(Optional) Selects the inbound dial peer based on the
calling number.
Step 2
Router(config-dial-peer)# incoming called-number
string
(Optional) Selects the inbound dial peer based on the
called number to identify voice and modem calls.
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Step 3
Router(config-dial-peer)# dtmf-relay [cisco-rtp]
[h245-signal] [h245-alphanumeric]
(Optional) Configures the tone that sounds in response to
a keypress on a touch-tone telephone. Dual tone
multifrequency (DTMF) tones are compressed at one end
of a call and decompressed at the other end.
If a low-bandwidth codec such as G.729 or G.723 is used,
the tones can sound distorted. The dtmf-relay command
transports DTMF tones generated after call establishment
out-of-band by using a method that sends with greater
fidelity than is possible in-band for most low-bandwidth
codecs. Without DTMF Relay, calls established with
low-bandwidth codecs can have trouble accessing
automated telephone menu systems such as voice mail and
interactive voice response (IVR) systems.
A signaling method is supplied only if the remote end
supports it. Options are Cisco proprietary (cisco-rtp),
standard H.323 (h245-alphanumeric), and H.323
standard with signal duration (h245-signal).
Step 4
Router(config-dial-peer)# fax rate {2400 | 4800 |
7200 | 9600 | 12000 | 14400 | disable | voice}
(Optional) Specifies the transmission speed of a fax to be
sent to this dial peer. The disable keyword turns off fax
transmission capability. The voice keyword, which is the
default, specifies the highest possible transmission speed
supported by the voice rate.
Step 5
Router(config-dial-peer)# numbering-type
{abbreviated | international | national | network
| reserved | subscriber | unknown}
(Optional) Specifies the numbering type to match, as
defined by the ITU Q.931 specification. For more
information, see the “Numbering Type Matching” section
on page 147.
Step 6
Router(config-dial-peer)# playout-delay mode
{adaptive | fixed}
(Optional) Specifies the type of jitter buffer playout delay
to use.
Step 7
Router(config-dial-peer)# playout-delay {maximum
value | nominal value | minimum {default | low |
high}}
(Optional) Specifies the amount of time that a packet is
held in the jitter buffer before it is played out on the audio
path. For detailed information, see the chapter “Quality of
Service” in this document.
Step 8
Router(config-dial-peer)# preference value
(Optional) Configures a preference for the VoIP dial peer.
The value is a number from 0 through 10, where the lower
the number, the higher the preference. For more
information, see the “Hunt Groups and Preferences”
section on page 144.
Step 9
Router(config-dial-peer)# tech-prefix number
(Optional) Specifies that a particular technology prefix be
prepended to the destination pattern of this dial peer.
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Step 10 Router(config-dial-peer)# translate-outgoing
{called | calling} name-tag
Step 11 Router(config-dial-peer)# vad
(Optional) Specifies the translation rule set to apply to the
calling number or called number. For more information,
see the “Digit Translation Rules for VoIP” section on
page 157.
(Optional) Enables voice activity detection (VAD) by
disabling the transmission of packets during periods of
silence. VAD is enabled by default.
The minimum silence detection time for VAD can be
modified by using the voice vad-time global
configuration command.
The default for the vad command is enabled, which is normally the preferred configuration. If you are
operating on a high-bandwidth network and voice quality is of the highest importance, you should
disable VAD by using the no vad command. This results in better voice quality, but also requires higher
bandwidth for voice. For example, a broad industry average for VAD savings on links T1 and up is from
30 to 35 percent of the overall bandwidth.
Note
The music threshold that is configured by using the music-threshold voice-port command can affect
VAD performance.
Some codecs come with built-in VAD algorithms (specifically, G.729 Annex B and G.723.1 symmetric).
VAD can be used with all other codecs.
Configuring VoFR Dial Peers
To configure VoFR dial peers, see the “Configuring Voice over Frame Relay” chapter.
Configuring VoATM Dial Peers
To configure VoATM dial peers, see the “Configuring Voice over ATM” chapter.
Verifying POTS and VoIP Dial Peer Configurations
You can check the validity of your dial peer configuration by performing the following tasks:
•
If you have relatively few dial peers configured, you can use the show dial-peer voice command to
verify that the configuration is correct. To display a specific dial peer or to display all configured
dial peers, use this command. The following is sample output from the show dial-peer voice
command for a specific VoIP dial peer:
router# show dial-peer voice 10
VoiceOverIpPeer10
tag = 10, dest-pat = \Q',
incall-number = \Q+14087',
group = 0, Admin state is up, Operation state is down
Permission is Answer,
type = voip, session-target = \Q',
sess-proto = cisco, req-qos = bestEffort,
acc-qos = bestEffort,
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fax-rate = voice, codec = g729r8,
Expect factor = 10,Icpif = 30, VAD = disabled, Poor QOV Trap = disabled,
Connect Time = 0, Charged Units = 0
Successful Calls = 0, Failed Calls = 0
Accepted Calls = 0, Refused Calls = 0
Last Disconnect Cause is ""
Last Disconnect Text is ""
Last Setup Time = 0
•
To show the dial peer that matches a particular number (destination pattern), use the show dialplan
number command. The following example displays the VoIP dial peer associated with the
destination pattern 51234:
router# show dialplan number 51234
Macro Exp.: 14085551234
VoiceOverIpPeer1004
tag = 1004, destination-pattern = \Q+1408555....',
answer-address = \Q',
group = 1004, Admin state is up, Operation state is up
type = voip, session-target = \Qipv4:1.13.24.0',
ip precedence: 0
UDP checksum = disabled
session-protocol = cisco, req-qos = best-effort,
acc-qos = best-effort,
fax-rate = voice, codec = g729r8,
Expect factor = 10, Icpif = 30,
VAD = enabled, Poor QOV Trap = disabled
Connect Time = 0, Charged Units = 0
Successful Calls = 0, Failed Calls = 0
Accepted Calls = 0, Refused Calls = 0
Last Disconnect Cause is ""
Last Disconnect Text is ""
Last Setup Time = 0
Matched: +14085551234
Digits: 7
Target: ipv4:172.13.24.0
Troubleshooting Tips
You can troubleshoot your dial peer configurations by performing the following tasks:
•
Ping the associated IP address to confirm connectivity. If you cannot successfully ping your
destination, refer to the Cisco IOS IP Configuration Guide.
•
To verify that the operational status and administrative status of the dial peer is up, use the
show dial-peer voice command.
Note
To activate a dial peer, the answer-address, incoming called-number, or
destination-pattern with port or session-target command must be configured in the
dial peer.
•
To verify that the data is configured correctly on both routers, use the show dialplan number
command on the local and remote routers.
•
If you have configured number expansion, use the show num-exp command to check that the partial
number on the local router maps to the correct full E.164 telephone number on the remote router.
•
If you have configured translation rules, use the test translation-rule command to verify digit
manipulation.
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•
If you have configured a codec value, make sure that the same codec value is configured in both VoIP
dial peers on either side of the connection. You can verify the configured codec value by using the
show dial-peer voice or show dialplan number command.
•
To verify that the output string the router dials is correct, use the debug voip ccapi inout command.
•
To check Real-Time Transport Protocol (RTP) packets, use the debug cch323 rtp command.
•
To check logical channel negotiation, use the debug cch323 h245 command.
•
To check the call setup, use the debug cch323 h225 command.
Dial Peer Overview
Before setting up a dial plan, you should understand how the router matches dialed strings to inbound
and outbound dial peers. How the router matches dialed strings directly affects the digits that your users
have to dial, in addition to the digits that are collected and then forwarded or played out to the telephony
interface, such as a PBX, key system, or PSTN.
The following sections describe basic concepts on how the router selects a matching dial peer:
Note
•
Two-Stage Dialing, page 137
•
Variable-Length Matching, page 138
•
Matching Inbound Dial Peers, page 139
•
Inbound Dial Peers for IVR Applications, page 140
•
Matching Outbound Dial Peers, page 140
•
Default Routes for Outbound Call Legs, page 141
Unless otherwise noted, the concepts described in this section apply to VoIP, VoFR, and VoATM dial
peers.
Two-Stage Dialing
With two-stage dialing, when a voice call enters the network, the originating router collects dialed digits
until it can match an outbound dial peer. As soon as the router matches a dial peer, it immediately places
the call and forwards the associated dial string. No additional dialed digits are collected. The digits and
wildcards that are defined in the destination pattern determine how many digits the originating router
collects before matching the dial peer. Any digits dialed after the first dial peer is matched are dropped.
For example, if the dialed string is “1234599” and the originating router matches a dial peer with a
destination pattern of 123.., then the digits “99” are not collected. The call is placed immediately after
the digit “5” is dialed, and the dial string “12345” is forwarded to the next call leg.
On the terminating router, the left-justified digits that explicitly match the terminating POTS dial peer
are stripped off. Any trailing wildcard digits are considered excess digits. The terminating router
forwards these excess digits to the telephony interface. For example, if the dial string “1234599” is
matched on a terminating router to a destination pattern of “123..,” the digits “4599” are excess digits
and are forwarded to the telephony interface.
Figure 31 illustrates how the originating router collects a dial string and the terminating router forwards
the digits to the telephony device.
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Figure 31
Collecting and Forwarding Dialed Digits
Number 5551000 is
dialed. All digits are
sent to Router A.
555 is stripped off, 1000
is sent to PBX to complete
call at proper extension.
Router A sends
5551000 to Router B.
Router A
V
5554000
Router B
WAN
10.1.1.1
IP network
WAN
10.1.1.2
PBX
V
dial-peer voice 1 pots
destination-pattern 555 . . . .
port 1/0/0
35832
5551000
dial-peer voice 2 voip
destination-pattern 555 . . . .
session-target ipv4:10.1.1.2
The examples in Table 14 demonstrate how the originating router collects dialed digits for a given
destination pattern in the outbound voice-network dial peer.
Table 14
Digit Collection Based on Destination Pattern
Dialed Digits
Destination Pattern
Dial String Collected1
5551234
5......
5551234
5551234
555....
5551234
5551234
555
555
555123499
555....
5551234
1. These examples apply only to two-stage dialing, in which the router collects the dialed string digit by digit. If DID is enabled
in the inbound POTS dial peer, the router performs one-stage dialing, which means that the full dialed string is used regardless
of the destination pattern that is matched.
The router defaults to two-stage dialing unless you configure DID. For information on configuring DID,
see the “DID for POTS Dial Peers” section on page 142.
Variable-Length Matching
When matching dial peers, the router defaults to variable-length matching, which means that as long as
the left-justified digits in the dial string match the configured pattern in the dial peer, any digits beyond
the configured pattern are ignored for the purposes of matching. For example, dial string 5551212 would
match both of the following dial peers:
dial-peer voice 1 voip
destination-pattern 555
session target ipv4:10.10.1.1
dial-peer voice 2 voip
destination-pattern 5551212
session target ipv4:10.10.1.2
To disable variable-length matching for a dial peer, add the dollar sign ($) to the end of the destination
pattern, as shown:
dial-peer voice 1 voip
destination-pattern 555$
session target ipv4:10.10.1.1
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The $ character in the above configuration prevents this dial peer from being matched for dial string
5551212 because the extra digits beyond 555 are considered in the matching.
With two-stage dialing, the router collects the dialed string digit by digit. It attempts to match a dial peer
after each digit is received. As soon as it finds a match, it immediately routes the call. For example, given
the following configurations, the router would immediately match dial string 5551212 to dial peer 1.
dial-peer voice 1 voip
destination-pattern 555
session target ipv4:10.10.1.1
dial-peer voice 2 voip
destination-pattern 5551212
session target ipv4:10.10.1.2
If the router is performing two-stage dialing and you want to make sure that the full dial string is
collected before a dial peer is matched, you can use the timeout T-indicator as in variable-length dial
plans. For example, after the router waits until the full dial string is collected, dial string 5551212 would
match both of the following dial peers:
dial-peer voice 1 voip
destination-pattern 555T
session target ipv4:10.10.1.1
dial-peer voice 2 voip
destination-pattern 5551212T
session target ipv4:10.10.1.2
How the router selects a dial peer also depends on whether the dial peer is being matched for the inbound
or outbound call leg. For more information, see the “Matching Inbound Dial Peers” section on page 139
and the “Matching Outbound Dial Peers” section on page 140.
Matching Inbound Dial Peers
To match inbound call legs to dial peers, the router uses three information elements in the call setup
message and four configurable dial peer attributes. The three call setup elements are:
•
Called number or dialed number identification service (DNIS)—A set of numbers representing the
destination, which is derived from the ISDN setup message or CAS DNIS.
•
Calling number or automatic number identification (ANI)—A set of numbers representing the
origin, which is derived from the ISDN setup message or CAS ANI.
•
Voice port—The voice port carrying the call.
The four configurable dial peer attributes are:
•
Incoming called-number—A string representing the called number or DNIS. It is configured by
using the incoming called-number dial-peer configuration command in POTS or MMoIP dial
peers. For more information, see the “Identifying Voice and Modem Calls” section on page 144.
•
Answer address—A string representing the calling number or ANI. It is configured by using the
answer-address dial-peer configuration command in POTS or VoIP dial peers and is used only for
inbound calls from the IP network. For more information, see the “Answer Address for VoIP”
section on page 142.
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•
Destination pattern—A string representing the calling number or ANI. It is configured by using the
destination-pattern dial-peer configuration command in POTS or voice-network dial peers. For
more information, see the “Destination Pattern” section on page 120.
•
Port—The voice port through which calls to this dial peer are placed.
The router selects an inbound dial peer by matching the information elements in the setup message with
the dial peer attributes. The router attempts to match these items in the following order:
1.
Called number with incoming called-number
2.
Calling number with answer-address
3.
Calling number with destination-pattern
4.
Incoming voice port with configured voice port
The router must match only one of these conditions. It is not necessary for all the attributes to be
configured in the dial peer or that every attribute match the call setup information; only one condition
must be met for the router to select a dial peer. The router stops searching as soon as one dial peer is
matched and the call is routed according to the configured dial peer attributes. Even if there are other
dial peers that would match, only the first match is used.
Note
For a dial peer to be matched, its administrative state must be up. The dial peer administrative state
is up by default when it is configured with at least one of these commands: incoming called-number,
answer-address, or destination-pattern. If destination-pattern is used, the voice port or session
target must also be configured.
Inbound Dial Peers for IVR Applications
To identify an interactive voice response (IVR) application to handle inbound calls, the originating
router must match a POTS dial peer. You configure which IVR application handles incoming voice calls
by using the application dial-peer configuration command. If the router is unable to match an inbound
dial peer, or if the inbound dial peer does not specify an application, the default application handles the
call. The following configuration shows an example of specifying an IVR application for an inbound
POTS call leg:
dial-peer voice 571 pots
application tr6
destination-pattern 5714954
port 0:D
Matching Outbound Dial Peers
How the router selects an outbound dial peer depends on whether DID is configured in the inbound POTS
dial peer. If DID is not configured in the inbound POTS dial peer, the router collects the incoming dialed
string digit by digit. As soon as one dial peer is matched, the router immediately places the call using
the configured attributes in the matching dial peer.
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If DID is configured in the inbound POTS dial peer, the router uses the full incoming dial string to match
the destination pattern in the outbound dial peer. With DID, the setup message contains all the digits
necessary to route the call; no additional digit collection is required. If more than one dial peer matches
the dial string, all of the matching dial peers are used to form a rotary group. The router attempts to place
the outbound call leg using all of the dial peers in the rotary group until one is successful. For more
information on rotary groups, see the “Hunt Groups and Preferences” section on page 144.
For information on configuring DID, see the “DID for POTS Dial Peers” section on page 142.
Default Routes for Outbound Call Legs
Default routes reduce the number of dial peers that must be configured when calls that are not terminated
by other dial peers are sent to a central router, usually for forwarding to a PBX. A default route is a dial
peer that automatically matches any call that is not terminated by other dial peers. For example, in the
following configuration, the destination pattern 8... is a voice default route because all voice calls with
a dialed string that starts with 8 followed by at least three additional digits will either match on 8208 or
end up with 8..., which is the last-resort voice route used by the router if no other dial peer is matched.
dial-peer voice 8 pots
destination-pattern 8208
port 1/1
!
dial-peer voice 1000 pots
destination-pattern 8...
port 1/1
A default route could also be defined by using a single wildcard character with the timeout T-indicator
in the destination pattern, as shown in the following example:
dial-peer voice 1000 voip
destination-pattern .T
session-target ipv4:10.10.1.2
You should be careful, however, when using the T-indicator for default routes. Remember, when
matching dial peers for outbound call legs, the router places the call as soon as it finds the first matching
dial peer. The router could match on this dial peer immediately even if there were another dial peer with
a more explicit match and a more desirable route.
Note
The timeout T-indicator is appropriate only for two-stage dialing. If the router is configured for
one-stage dialing, which means that DID is configured in the inbound POTS dial peer, then the
timeout T-indicator is unnecessary.
Configuring Dial Peer Matching Features
You can define the attributes that the router uses to match dial peers by configuring specific dial peer
features. These dial peer matching features are described in the following sections:
•
Answer Address for VoIP, page 142
•
DID for POTS Dial Peers, page 142
•
Identifying Voice and Modem Calls, page 144
•
Hunt Groups and Preferences, page 144
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Note
•
Numbering Type Matching, page 147
•
Class of Restrictions, page 148
Unless otherwise noted, the concepts described in this section apply to VoIP, VoFR, and VoATM dial
peers.
Answer Address for VoIP
The answer-address command can be used to select the inbound dial peer for VoIP calls, instead of
using the destination pattern. If the answer-address command is configured in VoIP or POTS dial peers,
the router attempts to match the calling number to the string configured as the answer address before
attempting to match a destination pattern in any dial peer. The following dial peer would match any
inbound VoIP call that had a calling number of 5551212.
dial-peer voice 2 voip
answer-address 5551212
session target ipv4:192.168.1.1
For more information, see the “Matching Inbound Dial Peers” section on page 139.
Note
The answer-address command is not supported for VoFR or VoATM dial peers.
DID for POTS Dial Peers
The Direct Inward Dialing (DID) feature in dial peers enables the router to use the called number (DNIS)
to directly match an outbound dial peer when receiving an inbound call from a POTS interface. When
DID is configured on the inbound POTS dial peer, the called number (DNIS) is automatically used to
match the destination pattern for the outbound call leg.
Unless otherwise configured, when a voice call comes into the router, the router presents a dial tone to
the caller and collects digits until it can identify an outbound dial peer. This process is called two-stage
dialing. After the outbound dial peer is identified, the router forwards the call through to the destination
as configured in the dial peer.
You may prefer that the router use the called number (DNIS) to find a dial peer for the outbound call
leg—for example, if the switch connecting the call to the router has already collected all the dialed digits.
DID enables the router to match the called number to a dial peer and then directly place the outbound
call. With DID, the router does not present a dial tone to the caller and does not collect digits; it forwards
the call directly to the configured destination. This is called one-stage dialing.
Figure 32 shows a call scenario using DID.
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Figure 32
VoIP Call Using DID
Customers dial local
order number 5552020
PSTN
5274200
0:D
IP network
Inbound POTS dial peer
has DID configured.
Gateway matches called
number to VoIP dial peer.
10.1.1.2
V
Terminating gateway
answers call and sends
to call center.
36498
V
In Figure 32, the POTS dial peer that matches the incoming called-number has direct-inward-dial
configured:
dial-peer voice 100 pots
incoming called-number 5552020
direct-inward-dial
port 0:D
The direct-inward-dial command in the POTS dial peer tells the gateway to look for a destination
pattern in a dial peer that matches the DNIS. For example, if the dialed number is 5552020, the gateway
matches the following VoIP dial peer for the outbound call leg:
dial-peer voice 101 voip
destination-pattern 5552020
session target ipv4:10.1.1.2
The call is made across the IP network to 10.1.1.2, and a match is found in that terminating gateway:
dial-peer voice 555 pots
destination-pattern 5552020
port 0:D
prefix 5274200
This dial peer matches on the dialed number and changes that number to 52744200 with the prefix
command. The result is that the user dials a number, gets connected, and never knows that the number
reached is different from the number dialed.
Note
DID for POTS dial peers is not the same as analog DID for Cisco routers which enables DID trunk
service from the PSTN.
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Configuring Dial Peer Matching Features
To configure a POTS dial peer for DID, use the following commands beginning in global configuration
mode:
Step 1
Command
Purpose
Router(config)# dial-peer voice number pots
Enters dial-peer configuration mode and defines a
local dial peer that will connect to the POTS network.
The number is one or more digits identifying the dial
peer. Valid entries are from 1 to 2147483647.
Step 2
Specifies DID for this POTS dial peer.
Router(config-dial-peer)# direct-inward-dial
Note
DID is configured for inbound POTS dial peers only.
Identifying Voice and Modem Calls
When a Cisco router is handling both modem and voice calls, it needs to identify the service type of the
call—that is, whether the incoming call to the router is a modem or a voice call. When the router handles
only modem calls, the service type identification is handled through modem pools. Modem pools
associate calls with modem resources based on the called number (DNIS). In a mixed environment,
where the router receives both modem and voice calls, you need to identify the service type of a call by
using the incoming called-number command.
If the incoming called-number command is not configured, the router attempts to resolve whether an
incoming call is a modem or voice call on the basis of the interface over which the call comes. If the call
comes in over an interface associated with a modem pool, the call is assumed to be a modem call; if a
call comes in over a voice port associated with a POTS dial peer, the call is assumed to be a voice call.
To identify the service type of a call as voice, use the following commands beginning in global
configuration mode:
Command
Purpose
Step 1
Router(config)# dial-peer voice number {pots | voip
vofr | voatm}
Step 2
Router(config-dial-peer)# incoming called-number
number
|
Enters dial peer configuration mode.
Defines the telephone number that identifies voice
calls associated with this dial peer.
Hunt Groups and Preferences
The router supports the concept of hunt groups, sometimes called rotary groups, in which multiple dial
peers are configured with the same destination pattern. Because the destination of each POTS dial peer
is a single voice port to a telephony interface, hunt groups help ensure that calls get through even when
a specific voice port is busy. If the router is configured to hunt, it can forward a call to another voice port
when one voice port is busy.
For example, in the following configuration for Router A, four POTS dial peers are configured with
different destination patterns. Because each dial peer has a different destination pattern, no backup is
available if the voice port mapped to a particular dial peer is busy with another call.
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With a hunt group, if a voice port is busy, the router hunts for another voice port until it finds one that is
available. In the following example for Router B, each dial peer is configured using the same destination
pattern of 3000, forming a dial pool to that destination pattern.
Router A (Without Hunt Groups)
Router B (With Hunt Groups and Preferences)
dial-peer voice 1 pots
destination-pattern 3001
port 1/1
!
dial-peer voice 2 pots
destination-pattern 3002
port 1/2
!
dial-peer voice 3 pots
destination-pattern 3003
port 1/3
!
dial-peer voice 4 pots
destination-pattern 3004
port 1/4
dial-peer voice 1 pots
destination pattern 3000
port 1/1
preference 0
!
dial-peer voice 2 pots
destination pattern 3000
port 1/2
preference 1
!
dial-peer voice 3 pots
destination pattern 3000
port 1/3
preference 2
!
dial-peer voice 4 pots
destination pattern 3000
port 1/4
preference 3
To give specific dial peers in the pool a preference over other dial peers, you can configure the preference
order for each dial peer by using the preference command. The router attempts to place a call to the dial
peer with the highest preference. The configuration example given for Router B shows that all dial peers
have the same destination pattern, but different preference orders.
The lower the preference number, the higher the priority. The highest priority is given to the dial peer
with preference order 0. If the same preference is defined in multiple dial peers with the same destination
pattern, a dial peer is selected randomly.
By default, dial peers in a hunt group are selected according to the following criteria, in the order listed:
1.
Longest match in phone number—Destination pattern that matches the greatest number of dialed
digits. For example, if one dial peer is configured with a dial string of 345.... and a second dial
peer is configured with 3456789, the router would first select 3456789 because it has the longest
explicit match of the two dial peers.
2.
Explicit preference—Priority configured by using the preference dial peer command.
3.
Random selection—All destination patterns weighted equally.
You can change this default selection order or choose different methods for hunting dial peers by using
the dial-peer hunt global configuration command. An additional selection criteria is “least recent use,”
which selects the destination pattern that has waited the longest since being selected.
You can mix POTS and voice-network dial peers when creating hunt groups. This can be useful if you
want incoming calls to be sent over the packet network, except that if network connectivity fails, you
want to reroute the calls back through the PBX to the PSTN. This type of configuration is sometimes
referred to as hairpinning. Hairpinning is illustrated in Figure 33.
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Figure 33
Voice Call Using Hairpinning
Network
connection
fails
V
1/0:1
10.1.1.2
IP network
37357
1/1:0
PSTN
The following configuration shows an example of sending calls to the PSTN if the IP network fails:
dial-peer voice 101 voip
destination-pattern 472....
session target ipv4:192.168.100.1
preference 0
!
dial-peer voice 102 pots
destination-pattern 472....
prefix 472
port 1/0:1
preference 1
You cannot use the same preference numbers for POTS and voice-network dial peers within a hunt group.
You can set a separate preference order for each dial peer type, but the preference order does not work
on both at the same time. For example, you can configure preference order 0, 1, and 2 for POTS dial
peers, and you can configure preference order 0, 1, and 2 for the voice-network dial peers, but the two
preference orders are separate. The system resolves preference orders among POTS dial peers first.
Configuring Dial-Peer Hunting Options
Dial-peer hunting is enabled by default. To disable dial-peer hunting on a dial peer, use the following
commands beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# dial-peer voice number {pots | vofr
voip}
Step 2
Router(config-dial-peer)# huntstop
|
Enters dial-peer configuration mode for the specified
dial peer.
(Optional) Disables dial-peer hunting on the dial
peer. Once you enter this command, no further
hunting is allowed if a call fails on the selected dial
peer.
Use the no huntstop command to reenable dial-peer hunting if it has been disabled.
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To configure dial peer hunting options for all dial peers, use the following commands in global
configuration mode:
Command
Purpose
Step 1 Router(config)# dial-peer hunt
hunt-order-number
Step 2 Router(config)# voice hunt {user-busy |
invalid-number | unassigned-number}
(Optional) Specifies the hunt selection order for dial peers in a hunt
group. Valid entries are 0 through 7. The default is 0.
•
0—Longest match in phone number, explicit preference, random
selection
•
1—Longest match in phone number, explicit preference, least
recent use
•
2—Explicit preference, longest match in phone number, random
selection
•
3—Explicit preference, longest match in phone number, least
recent use
•
4—Least recent use, longest match in phone number, explicit
preference
•
5—Least recent use, explicit preference, longest match in phone
number
•
6—Random selection
•
7—Least recent use
(Optional) Defines how the originating or tandem router handles rotary
dial-peer hunting if it receives a disconnect cause code from the
terminating router.
•
user-busy sets the router to continue dial-peer hunting if it
receives a user-busy disconnect cause code from a destination
router.
•
invalid-number sets the router to stop dial-peer hunting if it
receives a an invalid-number disconnect cause code from a
destination router.
•
unassigned-number sets the router to stop dial-peer hunting if it
receives an unassigned-number disconnect cause code from a
destination router.
Numbering Type Matching
A dial peer can be selected according to the type of number field in the called party number or calling
party number information element, in addition to matching the dial peer based on the configured
destination pattern, answer address, or incoming called number. The type of number value is selected by
using the numbering-type dial-peer configuration command.
For example, in the following configuration, the dialed string “4085559999” would match this dial peer
if the type of number field for the called party number is “national.”
dial-peer voice 408 voip
numbering-type national
destination-pattern 408.......
session target ipv4:10.1.1.2
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The following numbering types can be used:
•
Abbreviated—Abbreviated representation of the complete number as supported by this network
•
International—Number called to reach a subscriber in another country
•
National—Number called to reach a subscriber in the same country, but outside the local network
•
Network—Administrative or service number specific to the serving network
•
Reserved—Reserved for extension
•
Subscriber—Number called to reach a subscriber in the same local network
•
Unknown—Type of number is unknown by the network
For detailed information about these numbering types, see ITU-T Recommendation Q.931
Configuring Numbering-Type Matching
To configure numbering-type matching for a dial peer call leg, use the following commands beginning
in global configuration mode:
Command
Purpose
Step 1
Router(config)# dial-peer voice number {pots | voip
| vofr | voatm}
Enters dial peer configuration mode.
Step 2
Router(config-dial-peer)# numbering-type
{abbreviated | international | national | network |
reserved | subscriber | unknown}
Specifies the numbering type to match, as defined by
the ITU Q.931 specification.
Note
To match a dial peer using the numbering-type command, you must also configure the
destination-pattern, answer-address, or incoming called-number command.
Class of Restrictions
The Class of Restrictions (COR) feature provides the ability to deny certain call attempts based on the
incoming and outgoing class of restrictions provisioned on the dial peers. This functionality provides
flexibility in network design, allows users to block calls (for example, to 900 numbers), and applies
different restrictions to call attempts from different originators.
COR is used to specify which incoming dial peer can use which outgoing dial peer to make a call. Each
dial peer can be provisioned with an incoming and an outgoing COR list. The incoming COR list
indicates the capability of the dial peer to initiate certain classes of calls. The outgoing COR list
indicates the capability required for an incoming dial peer to deliver a call via this outgoing dial peer. If
the capabilities of the incoming dial peer are not the same or a superset of the capabilities required by
the outgoing dial peer, the call cannot be completed using this outgoing dial peer.
A typical application of COR is to define a COR name for the number that an outgoing dial peer serves,
then define a list that contains only that COR name, and assign that list as corlist outgoing for this
outgoing dial peer. For example, dial peer with destination pattern 5T can have a corlist outgoing that
contains COR 5x, as shown in the following configuration.
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The next step, in the typical application, is to determine how many call permission groups are needed,
and define a COR list for each group. For example, group A is allowed to call 5x and 6x, and group B is
allowed to call 5x, 6x, and 1900x. Then, for each incoming dial peer, we can assign a group for it, which
defines what number an incoming dial peer can call. Assigning a group means assigning a corlist
incoming to this incoming dial peer.
dial-peer cor
name 5x
name 6x
name 1900x
!
dial-peer cor
member 5x
member 6x
!
dial-peer cor
member 5x
member 6x
member 1900x
!
dial-peer cor
member 5x
!
dial-peer cor
member 6x
!
dial-peer cor
member 1900x
custom
list listA
list listB
list list5x
list list6x
list list1900x
! outgoing dialpeer 100, 200, 300
dial-peer voice 100 pots
destination-pattern 5T
corlist outgoing list5x
dial-peer voice 200 pots
destination-pattern 6T
corlist outgoing list6x
dial-peer voice 300 pots
destination-pattern 1900T
corlist outgoing list1900x
!
! incoming dialpeer 400, 500
dial-peer voice 400 pots
answer-address 525....
corlist incoming listA
dial-peer voice 500 pots
answer-address 526
corlist incoming listB
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Configuring Classes of Restrictions
To configure classes of restrictions for dial peers, use the following commands beginning in global
configuration mode:
Command
Purpose
Step 1
Router(config)# dial-peer cor custom
Enters COR configuration mode to specify classes of
restrictions to apply to dial peers.
Step 2
Router(config-dp-cor)# name class-name
Provides a name for a class of restrictions.
Note
Repeat this step for additional class names, as needed.
These class names are used to define the COR lists
configured in Step 4 and Step 5.
Step 3
Router(config-dp-cor)# exit
Exits COR configuration mode.
Step 4
Router(config)# dial-peer cor list list-name
Provides a name for a list of restrictions.
Step 5
Router(config-dp-corlist)# member class-name
Adds a COR class to this list of restrictions.
The member is a class named in Step 2.
Note
Repeat Step 4 and Step 5 to define another list and its
membership, as needed.
Step 6
Router(config-dp-corlist)# exit
Exits COR-list configuration mode.
Step 7
Router(config)# dial-peer voice number {pots
| voip}
Enters dial-peer configuration mode and defines a dial peer.
Step 8
Router(config-dial-peer)# corlist incoming
cor-list-name
Specifies the COR list to be used when this is the incoming dial
peer.
Step 9
Router(config-dial-peer)# corlist outgoing
cor-list-name
Specifies the COR list to be used when this is the outgoing dial
peer.
Note
Repeat Step 7 through Step 9 for additional dial peers,
as needed.
Verifying Classes of Restrictions
To check the validity of your classes of restrictions configuration, perform the following tasks:
•
Enter the show dial-peer voice command to learn whether the COR list fields are set as desired on
a dial peer:
Router# show dial-peer voice 210
VoiceEncapPeer210
information type = voice,
tag = 210, destination-pattern = `221',
answer-address = `', preference=0,
numbering Type = `unknown'
group = 210, Admin state is up, Operation state is up,
incoming called-number = `221', connections/maximum = 4/unlimited,
DTMF Relay = disabled,
Modem = system passthrough ,
huntstop = disabled,
application associated:
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
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type = pots, prefix = `221',
forward-digits default
session-target = `', voice-port = `1/0/8:D',
direct-inward-dial = enabled,
digit_strip = enabled,
•
Enter the show dial-peer cor command to display the COR names and lists you defined:
Router# show dial-peer cor
Class of Restriction
name:900block
name:800_call
name:Catchall
COR list <list1>
member:900block
member:800_call
COR list <list2>
member:900block
COR list <list3>
member:900block
member:800_call
member:Catchall
Configuring Digit Manipulation
The router may need to manipulate digits in a dial string before it passes the dial string to the telephony
device. This can be necessary, for instance, when calling PBXs with different capabilities to accept
digits, or for PSTN and international calls. You may need to consider different strategies for configuring
digit manipulation within your dial peers depending on your existing dial plan, the digits users are
expected to dial, and the capabilities of your PBX or key system unit (KSU). These digit-manipulation
options, in conjunction with the destination pattern, determine the dial string that the router forwards to
the telephony device.
The following dial peer digit-manipulation options are described in this section:
Note
•
Digit Stripping and Prefixes, page 151
•
Forward Digits, page 154
•
Number Expansion, page 155
•
Digit Translation Rules for VoIP, page 157
Unless otherwise noted, these concepts apply to VoIP, VoFR, and VoATM networks.
Digit Stripping and Prefixes
When the terminating router matches a dial string to an outbound POTS dial peer, by default the router
strips off the left-justified digits that explicitly match the destination pattern. Any remaining digits,
called excess digits, are forwarded to the telephony interface, such as a PBX or the PSTN. For more
information about excess digits, see the “Two-Stage Dialing” section on page 137.
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Some telephony interfaces require that any digits that are stripped from the dial string must be recovered
to support a particular dial plan. You can accomplish this either by using the no digit-strip dial-peer
configuration command to disable the default digit-stripping behavior or by using the prefix dial-peer
configuration command to add digits to the front of the dial string before it is forwarded to the telephony
interface. These commands are supported only in POTS dial peers.
The no digit-strip command disables the automatic digit-stripping function so that matching digits are
not stripped from the dialed string before it is passed to the telephony interface. For example, in the
following dial peer configuration, the entire seven-digit dialed string is passed to the telephony interface:
dial-peer voice 100 pots
destination-pattern 555....
no digit-strip
port 1/0:1
Disabling digit stripping is useful when the telephony interface requires the full dialed string. With some
dial plans, however, the dialed digits must be manipulated according to specific rules. The prefix
command can be used to add specific digits to the front of the dialed string before it is forwarded to the
telephony interface.
For example, consider a telephone whose E.164 called number is 1(408)555-1234. This telephone can
be reached within the company by dialing its extension number, 51234. If you configure a destination
pattern of “1408555....” (the periods represent wildcards) for the associated outbound POTS dial peer,
the terminating gateway will strip off the digits “1408555” when it receives a call for 1(408)555-1234.
For the terminating gateway to forward the call to the appropriate destination, the digit “5” needs to be
prepended to the remaining digits. In this case, you would configure a prefix of 5, as shown in the
following dial peer configuration.
dial-peer voice 100 pots
destination-pattern 1408555....
prefix 5
port 1/0:1
A prefix can also include commas (,). Each comma indicates a one-second pause in dialing. For example,
consider a telephone whose E.164 called number is 1(408)555-1234; to reach this device, you must dial
“9.” In this case, you might configure “1408.......” as the destination pattern, and “9” as the prefix.
In this example, the terminating router will strip the digits “1408” from the called number and append
the digit “9” to the front of the remaining digits, so that the actual number dialed is” 9,5551234.” The
router pauses for one second between dialing the “9” and the “5551234” to allow for a secondary dial
tone. In this example, you would configure the router as follows:
dial-peer voice 100 pots
destination-pattern 1408.......
prefix 9,
port 1/0:1
Using a comma with the prefix command is useful when the router must allow for a secondary dial tone;
otherwise the router does not wait for the dial tone before playing out excess digits. Putting commas in
the prefix makes the router pause one second per comma, allowing for a dial tone to occur before the
router plays out the digits.
Figure 34 shows an example of a network using the no digit-strip command. In this example, a central
site (Site D) is connected to remote sites through routers (Sites A, B, and C), as well as through a Centrex
system for sites still using the PSTN (Sites E and F). The Centrex service requires the full 7-digit dial
string to complete calls. The dial peers are configured with a fixed-length 7-digit dial plan.
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Figure 34
Network with Digit Stripping Disabled or Prefixes Enabled
8204...
Site E
8205...
Site F
8201...
Site A
7 digits
Centrex
3 digits
7 digits
8202...
Site B
Frame
Relay
Site D
3 digits
7 digits
3 digits
35949
8203...
Site C
7 digits
When Site E (8204...) dials 8201999, the full 7-digit dialed string is passed through the Centrex to the
router at Site D. Router D matches the destination pattern 8201... and forwards the 7-digit dial string
to Router A. Router A matches the destination pattern 8201... , strips off the matching 8201, and
forwards the remaining 3-digit dial string to the PBX. The PBX matches the correct station and
completes the call to the proper extension.
Calls in the reverse direction are handled similarly, but because the Centrex service requires the full
7-digit dial string to complete calls, the POTS dial peer at Router D is configured with digit stripping
disabled. Alternatively, digit stripping could be enabled and the dial peer could instead be configured
with a 4-digit prefix, in this case 8204, which would result in forwarding the full dial string to the
Centrex service.
Router A
Router D
dial-peer voice 1 pots
destination-pattern 8201...
port 1/0:1
!
dial-peer voice 4 vofr
destination-pattern 8204...
session target s0 2
!
dial-peer voice 5 vofr
destination-pattern 8205...
session target s0 2
!
dial-peer voice 4 pots
destination-pattern 8204...
no digit-strip
port 1/0:1
!
dial-peer voice 5 pots
destination-pattern 8205...
no digit-strip
port 1/0:1
!
dial-peer voice 1 vofr
destination-pattern 8201...
session target s0 1
!
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Forward Digits
The forward-digits command controls the number of digits that are stripped before the dialed string is
passed to the telephony interface. On outbound POTS dial peers, the terminating router normally strips
off all digits that explicitly match the destination pattern in the terminating POTS dial peer. Only digits
matched by the wildcard pattern are forwarded. The forward-digits command can be used to forward a
fixed number of dialed digits, or all dialed digits, regardless of the number of digits that explicitly match
the destination pattern.
For example, the forward-digits 4 command tells the router to forward the last four digits in the dialed
string. The forward-digits all command instructs the router to forward the full dialed string. If the length
of the dialed string is longer than the length of the destination pattern, the forward-digits extra
command forwards the extra trailing digits. Extra digits are not forwarded, however, if the dial-peer
destination pattern is variable length; for example, 123T, 123...T.
The forward-digits command is supported only in POTS dial peers.
Figure 35 shows an example of routing voice calls through a PBX using forward digits. In this
configuration, Routers T1 and T2 are tandem nodes that must support forward digits so that calls from
Routers A, B, or C can make a call to extension 8208.
Figure 35
Routing Voice Calls Through a PBX Using Forward Digits
A
8…
T2
8208
T1
8200
1/1
8205
B
Frame Relay
8…
8…
8…
1/1
C
8…
8209
1/1
35945
1/1
In this example, all digits matched with destination 8... are forwarded to the appropriate port. For a
call from Router A to reach extension 8208, the call first terminates at Router T1, which plays out the
digits 8208 to the voice port connected to the PBX. The PBX then routes the voice call to Router T2.
The forward-digits all command is used here, but the forward-digits 4 command could also be used in
this example.
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The following dial peer configurations are required on each router for this example:
Router T1
Router T2
dial-peer voice 1 vofr
destination-pattern 8200
session-target s0 1
!
dial-peer voice 6 vofr
destination-pattern 8205
session-target s0 6
!
dial-peer voice 10 vofr
destination-pattern 8209
session-target s0 10
!
dial-peer voice 1 pots
destination-pattern 8...
forward-digits all
port 1/1
dial-peer voice 8 pots
destination-pattern 8208
port 1/1
!
dial-peer voice 1000 pots
destination-pattern 8...
forward-digits all
port 1/1
!
dial-peer voice 9999 pots
destination-pattern ....
forward-digits all
port 1/1
Router A
dial-peer voice 1 pots
destination-pattern 8200
port 1/1
!
dial-peer voice 1000 vofr
destination-pattern 8...
session-target s0 1
Number Expansion
In most corporate environments, the telephone network is configured so that you can reach a destination
by dialing only a portion (an extension number) of the full E.164 telephone number. You can define an
extension number as the destination pattern for a dial peer. The router can be configured to recognize the
extension number and expand it into its full E.164 dialed number when the num-exp global
configuration command is used with the destination-pattern dial-peer configuration command.
Number expansion is a globally applied rule that enables you to define a set of digits for the router to
prepend to the beginning of a dialed string before passing it to the remote telephony device. This reduces
the number of digits that a user must dial to reach a remote location. Number expansion is similar to
using a prefix, except that number expansion is applied globally to all dial peers.
Using a simple telephony-based example, suppose that John works in a company where employees
extensions are reached by dialing the last four digits of the full E.164 telephone number. The E.164
telephone number is 555-2123; John’s extension number is 2123. Suppose that every employee on John’s
floor has a telephone number that begins with the same first four digits: 5552. You could define each dial
peer’s destination pattern using each extension number, and then use number expansion to prepend the
first four digits onto the extension. In this example, the router could be configured as follows:
num-exp 2... 5552...
!
dial peer voice 1 pots
destination pattern 2123
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Number expansion can also be used to replace a dialed number with another number, as in the case of
call forwarding. Suppose that for some reason, John needs to have all of his telephone calls forwarded
to another number, 555-6611. In this example, you would configure the router as follows:
num-exp 2123 5556611
!
dial peer voice 1 pots
destination pattern 5556611
In this example, every time the device receives a call for extension 2123, the dialed digits will be
replaced with 555-6611 and the call will be forwarded to that telephone.
Before you configure the num-exp command, it is helpful to map individual telephone extensions to
their full E.164 dialed numbers. This task can be done easily by creating a number expansion table.
Creating a Number Expansion Table
Figure 36 shows a network for a small company that wants to use VoIP to integrate its telephony network
with its existing IP network. The destination patterns (or expanded telephone numbers) associated with
Router A are 408 115-xxxx, 408 116-xxxx, and 408 117-xxxx, where xxxx identifies the individual dial
peers by extension. The destination pattern (or expanded telephone number) associated with Router B is
729 555-xxxx.
Figure 36
VoIP Example for Number Expansion
729 555-1001
729 555-1002
408 115-1001
729 555-1000
408 116-1002
0:D
WAN
10.1.1.1
IP network
WAN
10.1.1.2
V
Router B
36850
1:D
0:D
Router A
V
729 555-1003
408 117-1003
Table 15 shows the number expansion table for this scenario. The information included in this example
must be configured on both Router A and Router B.
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Table 15
Sample Number Expansion Table
Extension
Destination Pattern
Num-Exp Command Entry
5....
408115....
num-exp 5.... 408115....
6....
408116....
num-exp 6.... 408116....
7....
408117....
num-exp 7.... 408117....
1...
729555....
num-exp 1... 729555....
The period (.) character represents wildcards (such as extension numbers) in a telephone number.
Configuring Number Expansion
To expand an extension number into its full telephone number, use the following command in global
configuration mode:
Command
Purpose
Router(config)# num-exp extension-number expanded-number
Configures number expansion globally for all dial peers.
The extension-number argument defines the extension
number to expand into the full telephone number that is
specified by the expanded-number argument.
The expanded-number argument defines the full
telephone number or destination pattern to which the
extension number is expanded.
Verifying Number Expansion
You can check the validity of your number expansion configuration by performing the following tasks:
•
Enter the show num-exp command to confirm that you have mapped the telephone numbers
correctly.
•
Enter the show dialplan number command to see how a telephone number maps to a dial peer.
Digit Translation Rules for VoIP
Digit translation rules are used to manipulate the calling number (ANI) or called number (DNIS) digits
for a voice call, or to change the numbering type of a call. Translation rules are used to convert a
telephone number into a different number before the call is matched to an inbound dial peer or before
the call is forwarded by the outbound dial peer. For example, within your company you may dial a
five-digit extension to reach an employee at another site. If the call is routed through the PSTN to reach
the other site, the originating gateway must use translation rules to convert the five-digit extension into
the 10-digit format that is recognized by the central office switch.
Translation rules are defined by using the translation-rule command. After you define a set of
translation rules, you can apply the rules to all inbound VoIP calls, to all inbound calls that terminate at
a specific voice port, and to individual inbound or outbound call legs according to the dial peer.
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Note
Digit translation rules are not supported for inbound SIP calls.
The following example shows a dial peer that is configured to use translation-rule set 1, which contains
ten translation rules. The first rule defined is rule 0, in which 910 is the pattern that must be matched and
replaced, and 0 is the pattern that is substituted for 910.
translation-rule 1
rule 0 ^910 0
rule 1 ^911 1
rule 2 ^912 2
rule 3 ^913 3
rule 4 ^914 4
rule 5 ^915 5
rule 6 ^916 6
rule 7 ^917 7
rule 8 ^918 8
rule 9 ^919 9
!
!
dial-peer voice 2 voip
destination-pattern 91..........
translate-outgoing called 1
session target ras
The configuration above results in the stripping of the leading digits 91 from any called number that
begins with 91 before the number is forwarded by the outbound VoIP dial peer. Use the caret (^) symbol
to specify that the matched digits must occur at the start of a dial string.
Note
Wildcard symbols such as the period (.), asterisk (*), percent sign (%), plus sign (+), and question
mark (?) are not valid in translation rules. The router simply ignores these symbols when converting
a number if they are used in a translation rule.
Translation rules can also be used to change the numbering type for a call. For example, some gateways
may tag any number with more than 11 digits as an international number, even when the user must dial
a 9 to reach an outside line. The following example shows a translation rule that converts any called
number that starts with 91, and that is tagged as an international number, into a national number without
the 9 before sending it to the PSTN.
translation-rule 20
rule 1 91 1 international national
!
!
dial-peer voice 10 pots
destination-pattern 91..........
translate-outgoing called 20
port 1:D
!
Note
Using digit translation rules with the num-exp or prefix command is not recommended unless it is
the only way to minimize confusion.
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Configuring Digit Translation Rules
To create digit translation rules, perform the tasks in the following procedure:
•
Creating Digit Translation Rules (Required)
To apply digit translation rules to VoIP calls, perform one or more of the following procedures:
•
Applying Translation Rules to Inbound POTS Calls (Optional)
•
Applying Translation Rules to Inbound VoIP Calls (Optional)
•
Applying Translation Rules to Outbound Call Legs (Optional)
Creating Digit Translation Rules
To enter translation-rule configuration mode and specify a set of translation rules, use the following
commands beginning in global configuration mode:
Step 1
Command
Purpose
Router(config)# translation-rule name-tag
Defines a digit translation-rule set and enters
translation-rule configuration mode. All subsequent
commands that you enter in this mode before you exit
will apply to this translation-rule set.
The name-tag argument represents a unique number
that identifies the set of translation rules. Valid
entries are from 1 to 2147483647.
Step 2
Router(config-translate)# rule name-tag
input-matched-pattern substituted-pattern
[match-type substituted-type]
Defines an individual translation rule. This command
can be entered up to 11 times to add an individual
translation rule to the translation rule set defined in
Step 1.
The name-tag argument represents a unique number
that identifies this individual translation rule. Valid
entries are from 0 to 10.
The input-matched-pattern argument defines the
digit string that must be matched, and then replaced
with the substituted-pattern. The substituted-pattern
argument defines the digit string that replaces the
input-matched-pattern.
The optional match-type argument defines the
numbering-type that you want to replace with the
numbering-type defined in substituted-type. Enter
any for the match-type if you want to match on any
numbering-type.
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Command
Purpose
Otherwise, enter one of the following keywords for
each of these arguments:
•
abbreviated
•
international
•
national
•
network
•
reserved
•
subscriber
•
unknown
For a description of these numbering-types, see the
“Numbering Type Matching” section on page 147.
To create additional individual translation rules to include in the translation-rule set, repeat Step 2.
Note
Applying translation rules to more than one call leg in an end-to-end call is not recommended.
Applying Translation Rules to Inbound POTS Calls
To apply a translation rule set to all inbound POTS calls that terminate on the same voice port, use the
following commands beginning in global configuration mode:
Step 1
Command
Purpose
Router(config)# voice-port location
Specifies the voice port through which the call
enters the router.
The voice-port command syntax is
platform-specific. For more information about the
syntax of this command, see the “Voice Port
Configuration” chapter.
Step 2
Router(config-voiceport)# translate {called |
calling} name-tag
Specifies the translation rule set to apply to the
called number or calling number.
The called keyword applies the translation rule to
the called party number. The calling keyword
applies the translation rule to the calling party
number.
The name-tag argument is the reference number of
the translation rule. Valid entries are 1 through
2147483647.
Note
When this method is used, the digit translation rules are executed first before the inbound POTS dial
peer is matched.
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Applying Translation Rules to Inbound VoIP Calls
To apply a translation rule set to all inbound VoIP calls that originate at an H.323 gateway, use the
following command in global configuration mode:
Command
Purpose
Router(config)# voip-incoming translation-rule {called |
calling} name-tag
Specifies the translation rule set to apply to all
inbound VoIP call legs that originate from an H.323
gateway.
The called keyword applies the translation rule to
the called party number. The calling keyword
applies the translation rule to the calling party
number.
The name-tag argument is the reference number of
the translation rule. Valid entries are 1 through
2147483647.
Note
When using this method, the digit translation rules are executed first before the inbound VoIP dial
peer is matched.
Note
Digit translation rules are not supported for inbound session initiation protocol (SIP) calls.
Applying Translation Rules to Outbound Call Legs
To apply a translation rule set to an outbound VoIP or POTS call leg, use the following commands
beginning in global configuration mode:
Step 1
Command
Purpose
Router(config)# dial-peer voice number voip
Enters dial-peer configuration mode to configure a
VoIP dial peer.
or
Step 2
Router(config)# dial-peer voice number pots
Enters dial-peer configuration mode to configure a
POTS dial peer.
Router(config-dial-peer)# translate-outgoing {called
| calling} name-tag
Specifies the translation rule set to apply to the
calling number or called number.
The called keyword applies the translation rule to
the called party number. The calling keyword
applies the translation rule to the calling party
number.
The name-tag argument is the reference number of
the translation rule. Valid entries are 1 through
2147483647.
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Note
Translation rules that are configured in a dial peer using the translate-outgoing command are not
applied to inbound call legs. When using two-stage dialing, the translation rules that are configured
in the voice port using the translate command are applied twice; after the inbound dial peer is
matched, and again after the digits are collected.
Note
If the prefix command is also configured in the dial peer, the translate-outgoing command is
executed first.
Verifying Digit Translation
To verify the configuration of a digit translation rule, enter the show translation-rule EXEC command.
The following example shows output for a specific translation rule:
Router# show translation-rule 10
Translation rule address: 0x62C4F4B0
Tag name: 10
Translation rule in_used 1
**** Xrule rule table *******
Rule : 1
in_used state: 1
Match pattern: 555.%
Sub pattern: 1408555
Match type: subscriber
Sub type: international
**** Xrule rule table *******
Rule : 2
in_used state: 1
Match pattern: 91.%
Sub pattern: 1
Match type: international
Sub type: national
**** Xrule rule table *******
Rule : 3
in_used state: 1
Match pattern: 527.%
Sub pattern: 1408527
Match type: subscriber
Sub type: international
To verify whether a digit translation rule functions as expected, enter the test translation-rule EXEC
command. The following example shows that when translation rule 10 is used, the number 5551212 is
converted to 14085551212:
Router# test translation-rule 10 5551212
The replaced number: 14085551212
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This chapter describes quality of service (QoS) for voice and has the following sections:
•
QoS for Voice Overview, page 163
•
QoS for Voice Configuration Prerequisites, page 172
•
QoS for Voice Configuration Task List, page 172
•
QoS for Voice Configuration Examples, page 175
For a complete description of the commands used in this chapter, refer to the Cisco IOS Voice, Video,
and Fax Command Reference. To locate documentation of other commands that appear in this chapter,
use the command reference master index or search online.
To identify the hardware platform or software image information associated with a feature in this
chapter, use the Feature Navigator on Cisco.com to search for information about the feature or refer to
the software release notes for a specific release. For more information, see the “Identifying Supported
Platforms” section in the “Using Cisco IOS Software” chapter.
QoS for Voice Overview
Networks today are carrying more data than ever in the form of bandwidth-intensive, real-time voice,
video, and data, which stretch network capability and resources. Cisco IOS software provides QoS
solutions that help to solve problems caused by increasing traffic demands on a network.
QoS refers to the ability of a network, whether the network is a complex network, small corporate
network, Internet service provider (ISP), or enterprise network, to provide better service to selected
network traffic over various technologies, including Frame Relay, ATM, Ethernet and 802.1 networks,
and SONET, as well as IP-routed networks that may use any or all of these underlying technologies.
The primary goals of QoS are to provide better and more predictable network service by providing
dedicated bandwidth, controlled jitter and latency, and improved loss characteristics. QoS achieves these
goals by providing tools for managing network congestion, shaping network traffic, using expensive
wide-area links more efficiently, and setting traffic policies across the network.
QoS provides these benefits:
•
Control over bandwidth, equipment, and wide-area facilities. As an example, you can limit the
bandwidth consumed over a backbone link by file transfer protocol (FTP) or queueing of an
important database access.
•
More efficient use of network resources—Network analysis management and accounting tools,
enable you to know what your network is being used for and ensure that you are servicing the most
important traffic to your business.
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•
Tailored services—QoS enables ISPs to offer carefully tailored grades of service differentiation to
their customers.
•
Coexistence of mission-critical applications—Cisco QoS technologies make certain that bandwidth
and minimum delays required by time-sensitive multimedia and voice applications are available; and
that other applications using the link get their fair service without interfering with mission-critical
traffic.
•
Foundation for a fully integrated network—Cisco QoS technologies fully integrates a multimedia
network, for example, by implementing weighted fair queueing (WFQ) to increase service
predictability and IP Precedence signaling for traffic differentiation. Also available is ReSerVation
Protocol (RSVP), which allows you to take advantage of dynamically signaled QoS.
The basic QoS architecture has three components necessary to deliver QoS across a network comprising
heterogeneous technologies (IP, ATM, LAN switches, etc.) as follows:
•
QoS within a single network element (for example, queueing, scheduling, and traffic shaping tools)
•
QoS signaling techniques for coordinating QoS from end-to-end between network elements
•
QoS policy, management, and accounting functions to control and administer end-to-end traffic
across a network
The next section describes the tools that Cisco IOS software provides in each section of the architecture,
which, when combined, can create end-to-end QoS or simply solve specific problems at various points
in the network.
For more information regarding the concepts and complexities of QoS, refer to the Cisco IOS Quality of
Service Solutions Configuration Guide and Cisco IOS Quality of Service Solutions Command Reference.
For more information about the configuration of playout delay (jitter), see the “Configuring Voice Ports”
chapter; and, for information about dial peers, see the “Configuring Dial Plans, Dial Peers, and Digit
Manipulation” chapter.
QoS for Voice Tools
Cisco offers many tools for implementing QoS for voice. In general, each network has individual
problems that you can solve using one or more of Cisco QoS tools. Voice over IP (VoIP) comes with its
own set of problems (packet loss, jitter, and handling delay) and QoS can help solve some of these
problems. Some of the problems QoS cannot solve are propagation delay, codec delay, sampling delay,
and digitalization delay.
The International Telecommunication Union Telecommunication Standardization Sector (ITU-T) G.114
recommendation suggests no more than 150 milliseconds (ms) of end-to-end delay to maintain good
voice quality.
This section contains a high-level overview of the following:
•
Edge Functions, page 165
•
Packet Classification, page 167
•
RSVP, page 167
•
IP RTP Priority, page 169
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Edge Functions
As VoIP networks are designed, edge functions usually correspond to wide-area networks (WANs) that
have less than a T1 or E1 line of bandwidth from the central site. The following concepts are discussed:
•
Bandwidth Limitations, page 165
•
Real-Time Transport Protocol, page 165
•
Queueing, page 166
Bandwidth Limitations
The first issue of major concern when designing a VoIP network is bandwidth constraints. Depending
upon which codec you use and how many voice samples you want per packet, the amount of bandwidth
per call can increase drastically. For a list of bandwidth consumed by codec, see Table 16.
Table 16
Codec Type and Sample Size Effects on Bandwidth
Codec
Bandwidth
Consumed
Bandwidth Consumed
with cRTP (2-Byte Header)
Sample Latency
G.729 with one 10-ms sample per frame
40 kbps
9.6 kbps
10 ms
G.729 with four 10-ms samples per frame
16 kbps
8.4 kbps
40 ms
G.729 with two 10-ms samples per frame
24 kbps
11.2 kbps
20 ms
G.711 with one 10-ms sample per frame
112 kbps
81.6 kbps
10 ms
G.711 with two 10-ms samples per frame
96 kbps
80.8 kbps
20 ms
In the table, 24 kbps of bandwidth is consumed when an 8-kbps codec is used. The amount of consumed
bandwidth is affected by the codec used. For example, if the G.729 codec is used for two 10-ms samples,
the amount of bandwidth consumed is 20 bytes per frame, which works out to 8 kbps. The packet headers
that include IP, RTP, and User Datagram Protocol (UDP) add 40 bytes to each frame. The header is twice
the amount of the payload.
If the G.729 codec is used with two 10-ms samples, without RTP header compression, 24 kbps are
consumed in each direction per call. Although this might not be a large amount for T1 (1.544-mbps), E1
(2.048-mbps), or higher circuits, it is a large amount (42 percent) for a 56-kbps circuit.
Also, the bandwidth does not include layer 2 headers (PPP, Frame Relay, etc.). It includes headers from
layer 3 (network layer) and above only. Therefore, the same G.729 call can consume different amounts
of bandwidth based upon which data link layer is used (Ethernet, Frame Relay, PPP, and etc.).
Real-Time Transport Protocol
To reduce the large percentage of bandwidth consumed by a G.729 voice call, you can use compressed
Real-Time Transport Protocol (cRTP). cRTP enables you to compress the 40-byte IP/RTP/UDP header
to 2 to 4 bytes most of the time.
With cRTP, the amount of traffic per VoIP call is reduced from 24 kbps to 11.2 kbps. This is a major
improvement for low-bandwidth links. A 56-kbps link, for example, can now carry four G.729 VoIP calls
at 11.2 kbps each. Without cRTP, only two G.729 VoIP calls at 24 kbps can be used.
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To avoid the unnecessary consumption of available bandwidth, cRTP is used on a link-by-link basis. This
compression scheme reduces the IP/RTP/UDP header to 2 bytes when UDP checksums are not used, or
4 bytes when UDP checksums are used.
cRTP uses some of the same techniques as TCP header compression. In TCP header compression, the
first factor-of-two reduction in data rate occurs because half of the bytes in the IP and TCP headers
remain constant over the life of the connection.
The big gain, however, comes from the fact that the difference from packet to packet is often constant,
even though several fields change in every packet. Therefore, the algorithm can simply add 1 to every
value received. By maintaining both the uncompressed header and the first-order differences in the
session state shared between the compressor and the decompressor, cRTP must communicate only an
indication that the second-order difference is zero. In that case, the decompressor can reconstruct the
original header without any loss of information, simply by adding the first-order differences to the saved,
uncompressed header as each compressed packet is received.
Just as TCP/IP header compression maintains shared state for multiple simultaneous TCP connections,
this IP/RTP/UDP compression must maintain state for multiple session contexts. A session context is
defined by the combination of the IP source and destination addresses, the UDP source and destination
ports, and the RTP synchronization source (SSRC) field. A compressor implementation might use a hash
function on these fields to index a table of stored session contexts.
The compressed packet carries a small integer, called the session context identifier, or CID, to indicate
in which session context that packet should be interpreted. The decompressor can use the CID to index
its table of stored session contexts.
cRTP can compress the 40 bytes of header down to 2 to 4 bytes most of the time. As such, about 98
percent of the time the compressed packet will be sent. Periodically, however, an entire uncompressed
header must be sent to verify that both sides have the correct state. Sometimes, changes occur in a field
that is usually constant, such as the payload type field. In such cases, the IP/RTP/UDP header cannot be
compressed, so an uncompressed header must be sent.
You should use cRTP on any WAN interface where voice bandwidth is a concern and a high proportion
of RTP traffic exists.
Queueing
Queueing is like the concept of first in first out (FIFO), which means that the first in line is the first to
get out of the line. FIFO queueing was the first type of queueing to be used in routers, and it is still useful,
depending upon the network topology. In networks today, with a variety of applications, protocols, and
users, a way to classify different traffic is required.
Cisco has several queueing tools that enable a network administrator to specify what type of traffic is
special or important and to queue the traffic based upon that information. The most popular technique is
WFQ.
There are the several queueing types that are listed below. For more information, see the Cisco IOS
Quality of Service Solutions Configuration Guide and Cisco IOS Quality of Service Solutions Command
Reference:
•
Weighted fair queueing
•
Custom queueing
•
Priority queueing
•
Class-based (CB) WFQ
•
Priority queuing (PQ) with CB-WFQ
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Packet Classification
To achieve your intended packet delivery, you must know how to properly weight WFQ. There are
different weighting techniques and ways to use them in various networks to achieve the degree of QoS
you require.
IP Precedence
IP Precedence is a value defined by the three bits in the type of service (ToS) field in an IP header. IP
Precedence enables a router to group traffic flows based upon the eight precedence settings and to queue
traffic based upon that information as well as on source address, destination address, and port numbers.
Policy Routing
Policy routing is a routing scheme that forwards packets to specific interfaces based on user-configured
policies. Such policies might specify that traffic sent from a particular network should be forwarded out
one interface, while all other traffic should be forwarded out another interface.
With policy-based routing, you can configure a defined policy for traffic flows and not have to rely
completely on routing protocols to determine traffic forwarding and routing. Policy routing also enables
you to set the IP Precedence field so that the network can utilize different classes of service.
You can base policies on IP addresses, port numbers, protocols, or the size of packets. You can use one
of these descriptors to create a simple policy, or you can use all of them to create a complicated policy.
All packets received on an interface with policy-based routing enabled are passed through enhanced
packet filters known as route maps. The route maps dictate where the packets are forwarded.
RSVP
RSVP enables endpoints to signal the network with the kind of QoS needed for a particular application.
Most networks are designed to assume what QoS applications require. Network administrators can use
RSVP as dynamic access lists. Using RSVP means that network administrators do not need to be
concerned with port numbers of IP packet flows because RSVP signals that information during its
original request.
RSVP is an out-of-band, end-to-end signaling protocol that requests a certain amount of bandwidth and
latency with each network hop that supports RSVP. If a network node (router) does not support RSVP,
RSVP moves onto the next hop. A network node has the option to approve or deny the reservation based
upon the load of the interface to which the service is requested.
VoIP Call Admission Control
Cisco VoIP call admission control (CAC) applications use RSVP to limit the accepted voice load on the
IP network and guarantee the QoS levels of calls. The VoIP CAC using RSVP synchronizes RSVP
signaling with Cisco H.323 Version 2 signaling to ensure that the bandwidth reservation is established
in both directions before a call moves to the alerting phase (ringing). This ensures that the called party
phone rings only after the resources for the call have been reserved. Using RSVP-based admission
control, VoIP applications can reserve network bandwidth and react appropriately if bandwidth
reservation fails.
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Prior to Cisco IOS release 12.1(3)XI and 12.1(5)T, VoIP gateways used H.323 Version 1 (Slow Connect)
procedures when initiating calls requiring bandwidth reservation. This feature, which is enabled by
default, allows gateways to use H.323 Version 2 (Fast Connect) for all calls, including those requiring
RSVP.
To enable backward compatibility, commands are available to force the originating gateway to initiate
calls using Slow Connect procedures if the terminating gateway is running Cisco IOS Release 12.1(1)T
or later. You can configure Slow Connect globally for all VoIP calls by using the h323 call start
voice-service configuration command, or configure Slow Connect per individual VoIP dial peer by using
the call start voice-class configuration command.
A timer can be set by using the call rsvp-sync serv-timer command to limit the number of seconds that
the terminating gateway waits for bandwidth reservation setup before proceeding with the call setup or
releasing the call, depending on the configured QoS level in the dial peers.
Synchronized RSVP is attempted for a VoIP call as long as the requested (desired) QoS for the associated
dial peer is set to controlled-load or guaranteed-delay. If the requested QoS level is set to the default of
best-effort, bandwidth reservation is not attempted. If RSVP reservation is attempted but fails, the
acceptable QoS for the dial peer determines the outcome of the call. When the acceptable QoS is
configured for best effort, the call setup proceeds, but without any bandwidth reservation in place. When
the acceptable QoS on either gateway is configured for other than best effort, and the RSVP reservation
fails, the call is released. The requested QoS and acceptable QoS are configured through Cisco IOS
software by using the req-qos and acc-qos dial-peer configuration commands, respectively.
Table 17 summarizes the results of nine call setup scenarios using Fast Connect, based on the QoS levels
configured in the VoIP dial peers at the originating and terminating gateways. The table does not include
cases in which the requested QoS is best-effort and the acceptable QoS is other than best-effort and is
valid only for calls using Fast Connect procedures.
Table 17
Call Results Based on Configured QoS Levels
Originating Gateway
Call
Scenarios Requested QoS
Acceptable QoS
Terminating Gateway
Requested QoS
Acceptable QoS
Results
1
controlled-load
controlled-load
controlled-load
controlled-load
Call proceeds only if both RSVP
or
or
or
or
reservations succeed.
guaranteed-delay guaranteed-delay guaranteed-delay guaranteed-delay
2
controlled-load
controlled-load
controlled-load
best-effort
or
or
or
guaranteed-delay guaranteed-delay guaranteed-delay
Call proceeds only if both RSVP
reservations succeed.
3
controlled-load
controlled-load
best-effort
or
or
guaranteed-delay guaranteed-delay
Call is released.
4
controlled-load
best-effort
or
guaranteed-delay
controlled-load
controlled-load
Call proceeds only if both RSVP
or
or
reservations succeed.
guaranteed-delay guaranteed-delay
5
best-effort
controlled-load
or
guaranteed-delay
controlled-load
best-effort
or
guaranteed-delay
Call proceeds regardless of RSVP
results. If RSVP reservation fails,
call receives best-effort service.
6
best-effort
controlled-load
or
guaranteed-delay
best-effort
Call proceeds with best-effort
service.
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Configuring Quality of Service for Voice
Traffic Policing for Voice Networks
Table 17
Call Results Based on Configured QoS Levels (continued)
Originating Gateway
Call
Scenarios Requested QoS
Acceptable QoS
Terminating Gateway
Requested QoS
Acceptable QoS
Results
7
best-effort
best-effort
controlled-load
controlled-load
Call is released.
or
or
guaranteed-delay guaranteed-delay
8
best-effort
best-effort
controlled-load
best-effort
or
guaranteed-delay
Call proceeds with best-effort
service.
9
best-effort
best-effort
best-effort
Call proceeds with best-effort
service.
best-effort
The following are the benefits of using CAC with RSVP:
•
VoIP gateways default to H.323 Version 2 (Fast Connect) for all calls.
•
Called-party phone rings only after bandwidth reservation is confirmed.
•
QoS for voice calls is guaranteed across the IP network.
The following are restrictions on VoIP CAC using RSVP:
•
To support RSVP-based QoS with H.323 Version 2 (Fast Connect), the originating and terminating
gateways must be running Cisco IOS Release 12.1(3)XI or 12.1(5)T, or later.
•
To support RSVP-based QoS with H.323 Version 1 (Slow Connect), Cisco H.323 Version 2 gateways
must be running Cisco IOS Release 12.1(1)T or later.
•
RSVP with multicast is not supported.
IP RTP Priority
When WFQ is enabled and IP RTP Priority is configured, a strict priority queue is created. You can use
IP RTP Priority to enable use of the strict priority queueing scheme for delay-sensitive data. You can
identify voice traffic by its UDP port numbers and classify it into a priority queue. The result is voice
traffic that has strict priority service in preference to all other traffic. This is the most highly
recommended classification scheme for VoIP networks on lower-bandwidth links (768 kbps and below).
Traffic Policing for Voice Networks
The preceding sections cover ways you can queue different flows of traffic and then prioritize those
flows. That is an important part of QoS. Sometimes, however, it is necessary to actually regulate or limit
the amount of traffic an application is allowed to send across various interfaces or networks.
Cisco IOS software has a few tools that enable network administrators to define how much bandwidth
an application or even a user can use. These features have two different tools: rate-limiting and shaping.
The main difference between these two traffic-regulation tools is that rate-limiting tools drop traffic
based upon policing, and shaping tools generally buffer the excess traffic while waiting for the next open
interval to transmit the data.
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Configuring Quality of Service for Voice
Traffic Shaping for Voice Networks
The similarities are in that both the rate-limiting and shaping tools identify when traffic exceeds the
thresholds set by the network administrator. Often, these two tools are used together. Traffic shaping is
used at the edge of the network (on customer premises) to make sure the customer is utilizing the
bandwidth for business needs. Rate-limiting tools often used in service provider networks to ensure that
a subscriber does not exceed the amount of bandwidth set by contract with the service provider.
You can rate-limit traffic by precedence, Media Access Control (MAC) address, IP addresses, or other
parameters. Network administrators also can configure access lists to create even more granular
rate-limiting policies.
Traffic Shaping for Voice Networks
Cisco IOS QoS software includes two types of traffic-shaping tools: Generic Traffic Shaping (GTS) and
Frame Relay traffic shaping (FRTS). The two traffic-shaping methods are similar in implementation,
although their command-line interfaces differ somewhat and they use different types of queues to contain
and shape traffic that is deferred.
If a packet is deferred, GTS uses a WFQ to hold the delayed traffic. FRTS uses either a custom queue
(CQ) or a priority queuing (PQ) to hold the delayed traffic, depending on what you configured. FRTS
also supports WFQ to hold delayed traffic.
Traffic shaping enables you to control the traffic going out of an interface to match its flow to the speed
of the remote, target interface and to ensure that the traffic conforms to policies contracted for it. Thus,
you can shape traffic adhering to a particular profile to meet downstream requirements, thereby
eliminating bottlenecks in topologies with data-rate mismatches.
You use traffic shaping primarily to do the following:
•
Control usage of available bandwidth
•
Establish traffic policies
•
Regulate traffic flow to avoid congestion
You can also use traffic shaping to do the following:
•
Configure an interface if you have a network with different access rates. Suppose one end of the link
in a Frame Relay network runs at 256 kbps and the other end runs at 128 kbps. Sending packets at
256 kbps could cause the applications using the link to fail.
•
Configure an interface to offer a subrate service. In this case, traffic shaping enables you to use the
router to partition your T1 or T3 links into smaller channels.
Traffic shaping prevents packet loss. It is especially important to use traffic shaping in Frame Relay
networks because the switch cannot determine which packets take precedence and, therefore, which
packets should be dropped when congestion occurs. It is critical for VoIP that you control latency. By
limiting the amount of traffic and traffic loss in the network, you can smooth out traffic patterns and give
priority to real-time traffic.
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Configuring Quality of Service for Voice
High-Speed Transport
High-Speed Transport
High-speed transport is defined as any interface higher than T1 speed. The QoS mechanisms required to
configure a high-speed transport are as follows:
•
Packet Over SONET/SDH (POS)—Prioritizes traffic on this high-speed interface up to OC-48.
•
Modified deficit round robin (MDRR)—Extends Deficit Round Robin (DRR) to provide priority for
real-time traffic such as VoIP. Within MDRR, IP packets are mapped to different CoS queues based
on precedence bits. All the queues are serviced in round-robin fashion except for one: the priority
queue used to handle voice traffic.
•
IP and ATM—Maps IP prioritization onto ATM by configuring precedence values to an IP packet to
different ATM PVCs. The IP prioritization enables the network administrator to have different
PVCs, allocating more important traffic over a variable bit rate (VBR) ATM circuit and less
important traffic over an unspecified bit rate (UBR) ATM circuit; or IP prioritization onto ATM
using queueing techniques such as WFQ to prioritize different flows by PVC.
Refer to the Cisco IOS Quality of Service Configuration Guide and Cisco IOS Quality of Service
Command Reference for detailed information.
Congestion Avoidance
Congestion avoidance works by dropping packets from different flows, causing applications to slow the
amount of traffic being sent.
WRED
Random Early Detection (RED) is a congestion avoidance mechanism, and Weighted RED (WRED) is
the Cisco IOS software implementation of dropping traffic to avoid global synchronization. WRED
combines the capabilities of the RED algorithm with IP precedence. This combination provides for
preferential traffic handling for higher-priority packets. It can selectively discard lower-priority traffic
when the interface starts to get congested and provide differentiated performance characteristics for
different classes of service. To fully comprehend how WRED works, you must understand TCP
packet-loss behavior.
TCP
A stream of data sent on a TCP connection is delivered reliably and in order to the destination.
Transmission is made reliable through the use of sequence numbers and acknowledgments. Segments
(segments sequentially numbered) carry an acknowledgment number, which is the sequence number of
the next expected data octet of transmissions in the reverse direction. When the TCP transmits a segment,
it puts a copy on a retransmission queue and starts a timer; when the acknowledgment for that data is
received, the segment is deleted from the queue. If the acknowledgment is not received before the timer
runs out, the segment is retransmitted.
To govern the flow of data into a TCP, flow control mechanisms are used. The data-receiving TCP reports
a window to the sending TCP. This window specifies the number of octets, starting with the
acknowledgment number that the data-receiving TCP is currently prepared to receive.
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QoS for Voice Configuration Prerequisites
QoS for Voice Configuration Prerequisites
The following are tasks that must be performed prior to configuring QoS for voice:
Note
•
Establish a working IP network. For information about configuring IP, see the Cisco IOS IP
Routing Configuration Guide.
•
Configure your VoIP gateway for H.323. To support RSVP-based QoS with H.323 Version 2
(Fast Connect), the originating and terminating gateways must be running Cisco IOS
Release 12.1(3)XI or 12.1(5)T, or later. For information about configuring the gateway, refer to the
Software Configuration Guide for Cisco 3600 and Cisco 2600 Series Routers or Configuring H.323
VoIP Gateway for Cisco Access Platforms and the “Configuring H.323 Gateways” chapter.
•
Enable RSVP on the appropriate interfaces on your gateways by using the ip rsvp bandwidth
interface configuration command. You must also enable fair queueing on these interfaces by using
the fair-queue interface configuration command. For information about enabling RSVP and fair
queueing, refer to the Cisco IOS Quality of Service Solutions Command Reference.
•
Set the QoS levels in your dial peers by using the req-qos and acc-qos dial-peer configuration
commands. For information about configuring QoS levels, see the “Configuring Dial Plans, Dial
Peers, and Digit Manipulation” chapter.
An inbound plain old telephone service (POTS) dial peer is not required if the originating and
terminating gateways have outbound VoIP dial peers configured to reach the calling number at the
far end and if the VoIP dial peers use the same QoS parameters. Configure an inbound POTS dial peer
if the corresponding outbound VoIP dial peers at the originating and terminating gateways do not
have matching QoS configurations, or if calls can be established in only one direction (for example,
or if calls can be made from gateway A to gateway B, but not from gateway B to gateway A).
For information on how to configure playout delay, echo cancellation, and voice levels, see the
“Configuring Voice Ports” chapter.
QoS for Voice Configuration Task List
Refer to the Cisco IOS Quality of Service Configuration Guide and Cisco IOS Quality of Service
Command Reference for tasks that enable QoS for your network. To configure the H.323 Gateway, see
the “Configuring H.323 Gateways” chapter.
The following sections describe optional configuration tasks for the VoIP Call Admission Control Using
RSVP feature. The tasks in the first section are for configuring synchronization:
•
Configuring Synchronization and the Reservation Timer, page 173 (Optional)
Use the following tasks only if you require backward compatibility with H.323, Version 2 (Slow
Connect) gateways running a release earlier than Cisco IOS Release 12.1(3)XI or 12.1(5)T (must be
Cisco IOS Release 12.1(1)T or later):
•
Configuring Slow Connect for VoIP Globally, page 173 (Optional)
•
Configuring Slow Connect for a Specific Dial Peer, page 174 (Optional)
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QoS for Voice Configuration Task List
Configuring Synchronization and the Reservation Timer
Synchronization between RSVP and the H.323 voice signaling protocol is enabled by default; no
configuration tasks are required to enable this feature. To enable the feature if the no call rsvp sync
command was used to disable it, use the following commands in global configuration mode:
Command
Purpose
Step 1
Router(config)# call rsvp sync
Enables synchronization between RSVP and the H.323
voice signaling protocol.
Step 2
Router(config)# call rsvp-sync resv-timer seconds
Sets the timer for reservation requests. The default is 10
seconds.
Configuring Slow Connect for VoIP Globally
To make an H.323 gateway backward-compatible with a destination gateway, use the following
commands beginning in global configuration mode. This procedure is optional and selects Slow Connect
globally for all VoIP services.
Command
Purpose
Step 1
Router(config)# voice service voip
Enters voice-service configuration mode for VoIP services.
Step 2
Router(conf-voi-serv)# h323 call start slow
Forces the H.323 gateway to use Slow Connect procedures.
Note
Note
To restore the default of Fast Connect, use the
h323 call start fast command.
The previous procedure selects Slow Connect globally for all VoIP calls. To change the type of
connect procedures for calls associated with a specific dial peer, use the following procedure:
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Configuring Quality of Service for Voice
Monitoring and Maintaining RSVP Call Admission Control
Configuring Slow Connect for a Specific Dial Peer
Note
This procedure is optional and selects Slow Connect for a specific VoIP dial peer.
To make an H.323 gateway backward-compatible with a destination gateway, use the following
commands beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# voice class h323 tag
Enters voice-class configuration mode and creates a voice
class for H.323 attributes.
Step 2
Router(config-class)# call start slow
Forces the H.323 gateway to use Slow Connect procedures.
The default is slow. The keyword system causes the H.323
gateway to use the connect procedure that is configured in
the voice-service configuration (see Configuring Slow
Connect for VoIP Globally).
or
Router(config-class)# call start system
Note
If you require Fast Connect for a specific dial peer,
use the call start fast command to restore the
default when configuring the Slow Connect for
VoIP globally.
Step 3
Router(config-class)# exit
Exits voice-class configuration mode and returns to global
configuration mode.
Step 4
Router(config)# dial-peer voice number voip
Enters dial-peer configuration mode for the VoIP dial peer.
Step 5
Router(config-dial-peer)# voice-class h323 tag
Assigns the voice class attributes to the dial peer, including
the H.323 connect procedure that was selected in Step 2.
Verifying the RSVP CAC Configuration
To verify that RSVP-based call admission control is configured correctly, enter the show
running-config privileged EXEC command to display the command settings for the router, as shown in
the “QoS for Voice Configuration Examples” section.
Monitoring and Maintaining RSVP Call Admission Control
To display the configuration parameters for RSVP synchronization and statistics for calls that initiate
RSVP, use the following commands in privileged EXEC mode:
Command
Purpose
Step 1
Router# show call rsvp-sync conf
Displays the RSVP synchronization configuration.
Step 2
Router# show call rsvp-sync stats
Displays statistics for calls that attempted RSVP
reservation.
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QoS for Voice Configuration Examples
QoS for Voice Configuration Examples
Refer to the Cisco IOS Quality of Service Configuration Guide and Cisco IOS Quality of Service
Command Reference for more information.
The following examples display the screen output using the show running-config command:
•
RSVP Synchronization Examples, page 175
•
H.323 Slow Connect by Voice Service Example, page 176
•
H.323 Slow Connect by Dial Peer Example, page 176
RSVP Synchronization Examples
The following examples show that calls can be made in either direction between gateway A and
gateway B, which are connected to POTS phones, with phone numbers 711 and 712, respectively. The
requested QoS indicates that RSVP setup must be complete before the destination phone rings. The
acceptable QoS indicates that the call is released if the RSVP setup fails or is not complete within the
allotted time.
3/0/0
V
10.10.107.107
Gateway B
IP
network
10.10.107.108
V
2/0/0
711
712
Gateway A
Gateway B
call rsvp-sync
call rsvp-sync resv-timer 15
!
interface Ethernet0/0
ip address 10.10.107.107 10.255.255.255
fair-queue 64 256 31
ip rsvp bandwidth 1000 1000
!
voice-port 3/0/0
!
dial-peer voice 712 voip
destination-pattern 712
session target ipv4:10.10.107.108
req-qos controlled-load
acc-qos controlled-load
!
dial-peer voice 711 pots
destination-pattern 711
port 3/0/0
call rsvp-sync
call rsvp-sync resv-timer 15
!
interface Ethernet0/0
ip address 10.10.107.108 10.255.255.255
fair-queue 64 256 31
ip rsvp bandwidth 1000 1000
!
voice-port 2/0/0
!
dial-peer voice 711 voip
destination-pattern 711
session target ipv4:10.10.107.107
req-qos controlled-load
acc-qos controlled-load
!
dial-peer voice 712 pots
destination-pattern 712
port 2/0/0
36034
Gateway A
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QoS for Voice Configuration Examples
H.323 Slow Connect by Voice Service Example
The following example shows that Slow Connect is configured globally for all VoIP calls because the
h323 call start slow command is used in the voice service configuration:
dial-peer voice 712 voip
destination-pattern 712
session target ipv4:10.10.107.108
req-qos controlled-load
acc-qos controlled-load
!
voice service voip
h323 call start slow
The following example shows the same basic configuration but demonstrates that when the
call start system command is used in the voice class configuration, the gateway defaults to the connect
procedure that is configured in the voice service; otherwise the dial peer configuration takes precedence
(see the section “H.323 Slow Connect by Dial Peer Example”).
dial-peer voice 712 voip
voice-class h323 2
destination-pattern 712
session target ipv4:10.10.107.108
req-qos controlled-load
acc-qos controlled-load
!
voice class h323 2
call start system
!
voice service voip
h323 call start slow
!
H.323 Slow Connect by Dial Peer Example
The following example shows that calls from VoIP dial peer 712 use Slow Connect procedures because
the call start slow command is configured in the voice class assigned to the dial peer:
dial-peer voice 712 voip
voice-class h323 2
destination-pattern 712
session target ipv4:10.10.107.108
req-qos controlled-load
acc-qos controlled-load
!
voice class h323 2
call start slow
!
voice service voip
h323 call start fast
!
Note
The h323 call start fast voice-service command is ignored because the voice class configuration
takes precedence, unless the call start system voice-class command is used (see the section “H.323
Slow Connect by Voice Service Example”).
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H.323 Support and Other VoIP Call
Control Signaling
Configuring Media Gateway Control Protocol
and Related Protocols
This chapter describes the concepts and configuration procedures for Media Gateway Control Protocol
(MGCP). MGCP defines the call control relationship between VoIP gateways that translate audio signals
to and from the packet network, and call agents (CAs). The CAs are responsible for processing the calls.
Note
An earlier implementation of the protocol, Simple Gateway Control Protocol (SGCP), is no
longer available as a standalone product. MGCP supports SGCP functionality for those
customers who want SGCP capabilities. For more information on SGCP, see Simple Gateway
Control Protocol Support on the Cisco MC3810 and Cisco 3600 Series Routers.
This chapter includes the following sections:
•
MGCP Configuration Overview, page 180
•
MGCP Prerequisite Tasks, page 183
•
MGCP Configuration Task List, page 184
•
MGCP Configuration Examples, page 190
Cisco IOS Release 12.2 supports the MGCP 0.1, SGCP 1.1+, SIP, and H.323 protocols on these
platforms:
•
Cisco 2600 series modular access routers
•
Cisco 3640 and Cisco 3660 multiservice platforms
•
Cisco AS5300 universal access server
•
Cisco uBR924 cable access router
•
Cisco Voice Gateway 200 (VG200)
For a complete description of the commands used in this chapter, refer to the Cisco IOS Voice, Video,
and Fax Command Reference. To locate documentation of other commands that appear in this chapter,
use the command reference master index or search online.
To identify the hardware platform or software image information associated with a feature in this
chapter, use the Feature Navigator on Cisco.com to search for information about the feature or refer to
the software release notes for a specific release. For more information, see the “Identifying Supported
Platforms” section in the “Using Cisco IOS Software” chapter.
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MGCP Configuration Overview
MGCP Configuration Overview
MGCP is an extension of the earlier version of the protocol SGCP and supports SGCP functionality in
addition to several enhancements. Systems using SGCP can easily migrate to MGCP, and MGCP
commands are available to enable the SGCP capabilities.
An MGCP gateway handles the translation between audio signals and the packet network. The gateways
interact with a CA, also called a Media Gateway Controller (MGC) that performs signal and call
processing on gateway calls. In the MGCP configurations that Cisco IOS supports, the gateway can be
a Cisco router, access server, or cable modem, and the CA is a server from a third-party vendor.
Configuration commands for MGCP define the path between the call agent and the gateway, the type of
gateway, and the type of calls handled by the gateway.
MGCP uses endpoints and connections to construct a call. Endpoints are sources of or destinations for
data, and can be physical or logical locations in a device. Connections can be point-to-point or
multipoint.
Similar to SGCP, MGCP uses UDP for establishing audio connections over IP networks. However,
MGCP also uses hairpinning to return a call to the PSTN when the packet network is not available.
Creating a call connection involves a series of signals and events that describe the connection process.
This information might include such indicators as the off-hook status, a ringing signal, or a signal to play
an announcement. These events and signals are specific to the type of endpoint involved in the call.
MGCP groups these events and signals into packages. A trunk package, for example, is a group of events
and signals relevant to a trunking gateway, while an announcement package groups events and signals
for an announcement server. MGCP supports seven package types that are as follows:
•
Trunk
•
Line
•
Dual-tone multifrequency (DTMF)
•
Generic media
•
Real-time Transport Protocol (RTP)
•
Announcement server
•
Script
The trunk package and line package are supported by default on certain types of gateways. Although
configuring a gateway with additional endpoint package information is optional, you may want to
specify the packages for your endpoints to add to or to override the defaults.
MGCP provides the following benefits:
•
Alternative dial tone for voice over IP environments—Deregulation in the telecommunications
industry gives competitive local exchange carriers (CLECs) opportunities to provide toll bypass
from the incumbent local exchange carriers (ILECs) by using VoIP. MGCP enables a VoIP system
to control call setup and teardown and CLASS features for less sophisticated gateways.
•
Configuration requirements for static VoIP network dial peers has been removed—When MGCP is
used as the call agent in a VoIP environment, configuring static VoIP network dial peers is not
required, and so the configuration is simplified. The MGCP call agent provides functions similar to
VoIP network dial peers.
Note
POTS dial peer configuration is still required.
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MGCP Configuration Overview
•
Migration paths—Systems using earlier versions of the protocol can migrate easily to MGCP.
•
Multiple protocols support and investment protection—Cisco IOS Release 12.2 supports
concurrently on the same hardware and software the MGCP Version 0.1, SGCP 1.1+, SIP, and H.323
protocols. VoIP solutions can use any of these popular protocols. Changing protocols for new
network solutions can be done without disrupting the current network or investing in new systems.
•
Varied network needs supported as follows:
– IXCs that have no legacy TDM equipment in their networks and want to deploy a fully featured
network that offers both long-distance services to corporate customers and connectivity to local
exchange carriers or other IXCs with traditional TDM equipment.
– IXCs who have TDM equipment in their networks and want to relieve the congestion in the
network using data technologies to carry voice traffic or to cap the growth of TDM ports. In
these situations, the packet network provides basic switched trunking without services or
features.
– Competitive CLECs who want to provide residential and enhanced services.
– Dial access customers who want enhanced SS7 access capabilities and increased performance,
reliability, scalability, and lower costs.
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MGCP Configuration Overview
Supported Gateways
MGCP supports residential and trunking gateways and each gateway is described in the following
sections.
Residential Gateway
A residential gateway (RGW) provides an interface between analog (RJ-11) calls from a telephone and
the VoIP network. Examples of RGWs include cable modems and the Cisco 2600 series routers. See
Figure 37 for an illustration of an RGW configuration.
Figure 37
Residential and Trunking Gateways
Trunking GW
Trunking GW
MGCP
MGCP
PSTN
IP
MGC (CA)
MGC (CA)
MGCP
IP
MGCP
Residential GW
Phone
Phone
Residential GW
35621
Residential GW
Phone
Phone
RGW functionality supports analog POTS calls for both SGCP and MGCP on the Cisco 2600 series
routers and Cisco uBR924 cable access router:
•
Call waiting and stutter dial tone are supported on the Cisco 2600 series router and Cisco uBR924
cable access router.
•
On-hook caller ID, distinctive ringing, and ring splash are supported only on the Cisco uBR924
cable access router.
•
A default call agent address can be specified for each FXS port on the Cisco uBR924 cable access
router.
•
Modem and fax calls are supported on the Cisco 2600 series router and Cisco uBR924 cable access
router.
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MGCP Prerequisite Tasks
Trunking Gateway
A trunking gateway (TGW) provides an interface between trunks on the public switched telephone
network (PSTN) and a VoIP network. A trunk can be a DS0, T1, or E1 line. Examples of TGWs include
access servers and routers. See Figure 38 for an illustration of a TGW configuration.
Figure 38
Trunking Gateways
MGC (CA)
MGC (CA)
ISUP
MGCP
ISUP
MGCP
SS7 GW
SS7 GW
IP
Trunking GW
Trunking GW
35619
PSTN
Phone
Phone
TGW functionality supports SGCP and MGCP on the following platforms:
•
Cisco AS5300 universal access servers and the Cisco 3660 router for SS7 calls.
•
Cisco AS5300 universal access servers with SGCP 1.1+ protocol for Feature Group D Operator
Services (FGD-OS) 911 outgoing calls on T1 lines.
•
Cisco AS5300 universal access servers for PRI/ISDN signaling. These calls are backhauled to the
CA.
•
Cisco AS5300 universal access servers and Cisco 3660 routers for T1 and E1 interfaces.
•
Cisco AS5300 universal access servers and Cisco 3660 routers for modem and fax calls.
MGCP Prerequisite Tasks
Complete the following tasks on your network before configuring MGCP:
•
Configure IP routing.
•
Configure voice ports.
•
Configure VoIP.
•
Configure the call agent.
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MGCP Configuration Task List
MGCP Configuration Task List
To configure MGCP, perform the tasks in the following sections. Each task in the list is identified as
either optional or required.
•
Do at least one of the following tasks, depending on your network configuration (required):
– Configuring a TGW for MGCP, page 184
– Configuring a TGW for SGCP, page 186
– Configuring an RGW, page 187
– Verifying the TGW or RGW Configuration, page 190
Note
•
Blocking New Calls and Gracefully Terminating Existing Calls, page 190 (optional)
•
Monitoring and Maintaining MGCP, page 190 (optional)
RGWs are configured only with MGCP.
Configuring a TGW for MGCP
To configure a TGW for MGCP, use the following commands beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# mgcp
Initiates the MGCP application.
Step 2
Router(config)# mgcp call-agent
[ipaddr|hostname] [port] service-type mgcp
Specifies the call agent’s IP address or domain name, the
port, and gateway control service type. The keywords and
arguments are as follows:
Step 3
Router(config)# controller t1 number
Cisco IOS Voice, Video, and Fax Configuration Guide
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•
ipaddr—Call agent’s IP address.
•
hostname—Call agent’s hostname, using the format
host.domain.ext.
•
port—Port for the call agent to use. Valid values are
from 1025 through 65535.
•
service-type—Type of gateway control service
supported by the call agent. Valid values are mgcp and
sgcp. For MGCP configurations, use mgcp.
Specifies the channel number of the T1 trunk to be used for
analog calls.
Configuring Media Gateway Control Protocol and Related Protocols
MGCP Configuration Task List
Step 4
Command
Purpose
Router(config-controller)# ds0-group
channel-number timeslots range type none
service mgcp
Configures the channelized T1 time slots to accept the
analog calls. The keywords and arguments are as follows:
•
channel-number—DS0 group number. Valid values are
from 0 to 23 for T1 interfaces and from 0 to 30 for E1
interfaces.
•
timeslots range—DS0 time slot range of values. Valid
values are from 1 to 24 for T1 interfaces and from 1 to
31 for E1 interfaces. The default value is 24.
•
type—Signaling type to be applied to the selected
group. For MGCP functionality, use none.
•
service—Type of service supported on the gateway.
Valid values are mgcp and sgcp. For MGCP
configurations, use mgcp.
Step 5
Router(config-controller)# exit
Exits controller configuration mode and returns to global
configuration mode.
Step 6
Router(config)# mgcp restart-delay value
(Optional) Specifies the delay value sent in the RSIP
graceful teardown method. The value range is from 0 to 600
seconds; the default is 0.
Step 7
Router(config)# mgcp package-capability
{s-package | dtmf-package | gm-package |
rtp-package | trunk-package | script-package}
(Optional) Specifies the event packages that are supported
on the gateway. The set of supported packages varies with
the type of gateway (TGW or RGW). The keywords are as
follows:
•
as-package—Announcement server package.
•
dtmf-package—DTMF package.
•
gm-package—Generic media package.
•
rtp-package—RTP package.
•
trunk-package (default)—Trunk package.
•
script-package—Script package. Available only on the
Cisco AS5300 universal access server.
Step 8
Router(config)# mgcp default-package
{as-package | dtmf-package | gm-package |
rtp-package | trunk-package}
(Optional) Specifies the event package that should act as the
default. Overrides the mgcp package-capability default
package.
Step 9
Router(config)# mgcp dtmf-relay {codec |
low-bit-rate} mode {cisco | out-of-band}
(Optional) Used for relaying digits through the IP network.
The keywords and arguments are as follows:
•
codec—G.711 or a G.726 codec.
•
low-bit-rate—Low-bit-rate codec other than G.711 and
G.726.
•
cisco—Removes DTMF tone from the voice stream and
sends FRF.11 with special payload 121 for DTMF
digits.
•
out-of-band—Removes DTMF tone from the voice
stream and does not send FRF.11.
The default is no mgcp dtmf-relay for all codecs.
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Configuring Media Gateway Control Protocol and Related Protocols
MGCP Configuration Task List
Step 10
Command
Purpose
Router(config)# mgcp modem passthru {cisco |
ca}
(Optional) Configures the gateway for modem and fax
data.The keywords and arguments are as follows:
•
cisco—Switching to the G.711 codec when the gateway
detects a modem or fax tone to allow the analog data to
pass-through.
•
ca (default)—Switching to the CA that switches to
G.711 codec when the gateway detects a modem or fax
tone to allow the analog data to pass through.
The no form of the command disables support for modem
and fax data.
Step 11
Router(config)# mgcp sdp simple
(Optional) Specifies use of a subset of the session
description protocol (SDP). Some call agents require this
subset to send data through the network. The default is
no mgcp sdp simple.
Configuring a TGW for SGCP
To configure a TGW for SGCP, use the following commands beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# mgcp
Initiates the MGCP application.
Step 2
Router(config)# mgcp call-agent [ipaddr | hostname]
[port] service-type sgcp
Specifies the call agent’s IP address or domain name,
the port, and gateway control service type. The
keywords and arguments are as follows:
•
ipaddr—Call agent’s IP address.
•
hostname—Call agent’s hostname, using the
format host.domain.ext.
•
port—Port for the call agent to use. Valid values
are from 1025 through 65535.
•
service-type—Type of gateway control service
supported by the call agent. Valid values are
mgcp and sgcp. For SGCP configurations, use
sgcp.
Step 3
Router(config)# controller t1 number
Specifies the channel number of the T1 trunk to be
used for analog calls.
Step 4
Router(config-controller)# ds0-group channel-number
timeslots range type {none | fgdos} [tone_type]
[addr_info] service {sgcp | voice}
Configures the channelized T1 time slots to accept the
analog calls. For type none, use service sgcp. For type
fgdos, use service voice.
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Configuring Media Gateway Control Protocol and Related Protocols
MGCP Configuration Task List
Command
Purpose
The keywords and arguments are as follows:
•
channel-number—DS0 group number. Valid
values are 1 to 23 for T1 interfaces and from 0 to
30 for E1 interfaces.
•
timeslots range—DS0 time slot range of values.
Valid values are from 1 to 24 for T1 interfaces
and from 1 to 31 for E1 interfaces. The default
value is 24.
•
type—Signaling type to be applied to the selected
group. For SGCP functionality, use none or
fgdos.
•
tone_type—Tone type supported by the signaling
type. For signaling type fgdos, the valid value is
mf. This parameter is available if type is fgdos.
•
addr_info—Calling and called party numbers are
used. For type fgdos, the valid value is dnis-ani.
This parameter is available if type is fgdos.
•
service—Type of service on the gateway. For
SGCP configurations, valid values are sgcp or
voice. For signaling type none, use sgcp. For
signaling type fgdos, use voice.
Configuring an RGW
To configure an RGW, use the following commands beginning in global configuration mode:
Step 1
Command
Purpose
Router(config)# mgcp
Initiates the MGCP application.
Note
Step 2
Step 3
Router(config)# mgcp call-agent [ipaddr | hostname]
[port] service-type mgcp
Router(config)# dial-peer voice number pots
RGWs are configured only with MGCP.
Specifies the call agent IP address or domain name,
the port, and gateway control service type. The
keywords and arguments are as follows:
•
ipaddr—Call agent’s IP address.
•
hostname—Call agent’s hostname, using the
format host.domain.ext.
•
port—Port for the call agent to use. Valid values
are from 1025 through 65535.
•
service-type—Type of gateway control service
supported by the call agent. Valid values are
mgcp or sgcp. For MGCP configurations, use
mgcp.
Sets up the dial peer for a voice port.
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MGCP Configuration Task List
Command
Purpose
Step 4
Router(config-dial-peer)# application MGCPAPP
Selects the MGCP application to run on the voice
port.
Step 5
Router(config-dial-peer)# exit
Exits dial peer configuration mode and returns to
global configuration mode.
Step 6
Router(config)# mgcp package-capability {line-package
| dtmf-package | gm-package | rtp-package}
(Optional) Specifies the event packages that are
supported on the gateway. The set of supported
packages varies with the type of gateway (TGW or
RGW). The keywords are as follows:
Step 7
Router(config)# mgcp default-package [line-package |
dtmf-package | gm-package]
•
line-package (default)—Line package.
•
dtmf-package—DTMF package.
•
gm-package—Generic media package.
•
rtp-package—RTP package.
(Optional) Specifies the event package that should
act as the default. Overrides the mgcp
package-capability command.
Configuring the Cisco Voice Gateway 200 to Support Cisco CallManager
The Cisco Voice Gateway 200 functions as an RGW and uses the configuration steps shown in the
Configuring an RGW section. In addition, the Cisco Voice Gateway 200 has the capability of using
MGCP with Cisco CallManager for administration and redundant call agent features. This capability
requires additional configuration steps.
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MGCP Configuration Task List
To configure the Cisco Voice Gateway 200 so it can be controlled by Cisco CallManager using MGCP,
use the following commands in global configuration mode:
Command
Step 1
Router(config)# ccm-manager MGCP
Step 2
Router(config)# ccm-manager redundant host hostname1
hostname2
Step 3
Router(config)# ccm-manager switchback {graceful |
immediate | schedule-time hh:mm | uptime-delay
minutes}
Purpose
Enables support for Cisco CallManager within
MGCP.
Identifies one or two backup Cisco CallManager
servers. The arguments hostname1 and hostname2
specify the first and second backup servers,
respectively, using the dotted decimal format.
Specifies how the gateway behaves if the primary
server becomes unavailable and later becomes
available again. The keywords and arguments are as
follows:
•
graceful—Completes all outstanding calls before
returning the gateway to the control of the
primary Cisco CallManager server.
•
immediate—Returns the gateway to the control
of the primary Cisco CallManager server without
delay, as soon as the network connection to the
server is reestablished.
•
schedule-time hh:mm—Returns the gateway to
the control of the primary Cisco CallManager
server at the specified time, where hh:mm is the
time according to a 24-hour clock. If the gateway
reestablishes a network connection to the primary
server after the configured time, the switchback
will occur at the specified time on the following
day.
•
uptime-delay minutes—Returns the gateway to
the control of the primary Cisco CallManager
server when the primary server runs for a
specified number of minutes after a network
connection is reestablished to the primary server.
Valid values are from 1 to 1440 (from 1 minute to
24 hours).
To force the Cisco Voice Gateway 200 to use the backup Cisco CallManager server, use the following
command in privileged EXEC mode:
Command
Purpose
Router# ccm-manager switchover-to-backup
Redirects the Cisco Voice Gateway 200 gateway to the
backup Cisco CallManager server.
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Configuring Media Gateway Control Protocol and Related Protocols
MGCP Configuration Examples
Verifying the TGW or RGW Configuration
To verify the configuration settings for all platforms and protocols, use the following commands in
global configuration mode:
Command
Purpose
Step 1
Router(config)# show running-config
Displays the current configuration settings.
Step 2
Router(config)# show ccm-manager
Displays the current configuration settings on the Cisco Voice
Gateway 200.
Blocking New Calls and Gracefully Terminating Existing Calls
You can block all new MGCP calls to the router and gracefully terminate all existing active calls, which
means that an active call is not terminated until the caller hangs up. To block all new calls, use the
following commands in global configuration mode:
Command
Purpose
Step 1
Router(config)# mgcp block-newcalls
Prevents the gateway from accepting new calls.
Step 2
Router(config)# no mgcp block-newcalls
Restarts normal MGCP call operation.
Monitoring and Maintaining MGCP
To monitor the MGCP configuration, use the following commands in privileged EXEC mode:
Command
Purpose
Step 1
Router# show mgcp [connection | endpoint
| statistics]
Displays all active MGCP connections on the router.
Step 2
Router# debug mgcp [all | errors |
events | packets | parser]
Turns on debugging for the gateway.
Step 3
Router# clear mgcp statistics
Resets the MGCP statistical counters.
MGCP Configuration Examples
This section provides configuration examples for each of the supported platforms:
•
Configuring the Cisco AS5300 As a TGW with MGCP: Example, page 191
•
Configuring the Cisco AS5300 As a TGW with SGCP: Example, page 192
•
Configuring the Cisco 3660 as a TGW with MGCP: Example, page 193
•
Configuring the Cisco uBR924 as an RGW: Example, page 194
•
Configuring the Cisco 2620 as an RGW: Example, page 196
•
Configuring the Cisco Voice Gateway 200 as an RGW: Example, page 197
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Configuring Media Gateway Control Protocol and Related Protocols
MGCP Configuration Examples
Configuring the Cisco AS5300 As a TGW with MGCP: Example
The following example illustrates a configuration only for MGCP calls. FGD-OS calls are not supported.
version 12.2
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname A
!
resource-pool disable
!
ip subnet-zero
ip ftp username smith
ip host B 209.165.200.225
ip host C 209.165.200.226
ip domain-name cisco.com
ip name-server 209.165.202.129
!
mgcp
mgcp request timeout 10000
mgcp call-agent 192.168.10.10 2302
mgcp restart-delay 5
mgcp package-capability gm-package
mgcp package-capability dtmf-package
mgcp package-capability trunk-package
mgcp package-capability rtp-package
mgcp package-capability as-package
mgcp package-capability mf-package
mgcp package-capability script-package
mgcp default-package trunk-package
mta receive maximum-recipients 0
!
controller T1 0
framing esf
clock source line primary
linecode b8zs
ds0-group 0 timeslots 1-24 type none service
!
controller T1 1
framing esf
clock source line secondary 1
linecode b8zs
ds0-group 0 timeslots 1-24 type none service
!
controller T1 2
framing esf
linecode b8zs
ds0-group 0 timeslots 1-24 type none service
!
controller T1 3
framing esf
linecode b8zs
ds0-group 0 timeslots 1-24 type none service
!
voice-port 0:0
!
voice-port 1:0
!
voice-port 2:0
!
voice-port 3:0
mgcp
mgcp
mgcp
mgcp
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MGCP Configuration Examples
!
interface Ethernet0
ip address 192.168.10.9 255.255.255.0
no ip directed-broadcast
!
interface FastEthernet0
ip address 172.22.91.73 255.255.255.0
no ip directed-broadcast
shutdown
duplex auto
speed auto
!
no ip classless
ip route 0.0.0.0 0.0.0.0 172.22.91.1
ip route 209.165.200.225 255.255.255.255 192.168.0.1
no ip http server
!
line con 0
exec-timeout 0 0
transport input none
line aux 0
line vty 0 4
login
!
end
Configuring the Cisco AS5300 As a TGW with SGCP: Example
The following example illustrates a configuration that supports MGCP and FGD-OS calls:
version 12.2
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname A
!
resource-pool disable
!
ip subnet-zero
ip ftp username smith
ip host B 209.165.200.225
ip host C 209.165.200.226
ip domain-name cisco.com
ip name-server 209.165.202.129
!
mgcp
mgcp request timeout 10000
mgcp call-agent 192.168.10.10 2302 sgcp
mta receive maximum-recipients 0
!
controller T1 0
framing esf
clock source line primary
linecode b8zs
ds0-group 0 timeslots 1-24 type none service mgcp
!
controller T1 1
framing esf
clock source line secondary 1
linecode b8zs
ds0-group 0 timeslots 1-24 type fgd-os mf dnis-ani service voice
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MGCP Configuration Examples
!
controller T1 2
framing esf
linecode b8zs
ds0-group 0 timeslots 1-24 type none service mgcp
!
controller T1 3
framing esf
linecode b8zs
ds0-group 0 timeslots 1-24 type none service mgcp
!
!voice-port 0:0
!
voice-port 1:0
!
voice-port 2:0
!
voice-port 3:0
!
interface Ethernet0
ip address 192.168.10.9 255.255.255.0
no ip directed-broadcast
!
interface FastEthernet0
ip address 172.22.91.73 255.255.255.0
no ip directed-broadcast
shutdown
duplex auto
speed auto
!
no ip classless
ip route 0.0.0.0 0.0.0.0 172.22.91.1
ip route 209.165.200.225 255.255.255.255 192.168.0.1
no ip http server
!
line con 0
exec-timeout 0 0
transport input none
line aux 0
line vty 0 4
login
!
end
Configuring the Cisco 3660 as a TGW with MGCP: Example
The following example illustrates a platform that does not support FGD-OS calls.
version 12.2
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname A
!
memory-size iomem 40
voice-card 1
!
ip subnet-zero
!
mgcp 4000
mgcp call-agent 209.165.202.129 4000
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MGCP Configuration Examples
mgcp package-capability gm-package
mgcp package-capability dtmf-package
mgcp package-capability rtp-package
mgcp package-capability as-package
isdn voice-call-failure 0
cns event-service server
!
controller T1 1/0
framing esf
clock source internal
ds0-group 1 timeslots 1-24 type none service mgcp
!
controller T1 1/1
framing esf
clock source internal
ds0-group 1 timeslots 1-24 type none service mgcp
!
voice-port 1/0:1
!
voice-port 1/1:1
!
interface FastEthernet0/0
ip address 209.165.202.140 255.255.255.0
no ip directed-broadcast
load-interval 30
duplex auto
speed auto
!
interface FastEthernet0/1
no ip address
no ip directed-broadcast
no ip mroute-cache
load-interval 30
shutdown
duplex auto
speed auto
!
ip default-gateway 209.165.202.130
ip classless
ip route 209.165.200.225 255.255.255.255 FastEthernet0/0
no ip http server
!
snmp-server engineID local 00000009020000107BD8CD80
snmp-server community public RO
!
line con 0
exec-timeout 0 0
transport input none
line aux 0
line vty 0 4
login
!
end
Configuring the Cisco uBR924 as an RGW: Example
The following example illustrates a platform that does not support FGD-OS calls.
version 12.2
no service pad
service timestamps debug uptime
service timestamps log uptime
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MGCP Configuration Examples
no service password-encryption
!
hostname A
!
logging buffered 200000 debugging
!
clock timezone - -8
ip subnet-zero
no ip routing
no ip domain-lookup
ip host A 192.168.147.91
ip host C 209.165.200.224
ip host D 209.165.200.225
!
mgcp
mgcp call-agent 192.168.10.10 2490
mgcp package-capability gm-package
mgcp package-capability dtmf-package
mgcp package-capability line-package
mgcp default-package line-package
!
voice-port 0
input gain -3
!
voice-port 1
input gain -3
!
dial-peer voice 1 pots
application MGCPAPP
port 1
!
dial-peer voice 2 pots
application MGCPAPP
port 0
!
interface Ethernet0
ip address 192.168.147.91 255.255.255.0
no ip directed-broadcast
no ip route-cache
no ip mroute-cache
!
interface cable-modem0
ip address negotiated
no ip directed-broadcast
no ip route-cache
no ip mroute-cache
cable-modem downstream saved channel 459000000 20
cable-modem downstream saved channel 699000000 19 2
cable-modem mac-timer t2 100000
no cable-modem compliant bridge
bridge-group 59
bridge-group 59 spanning-disabled
!
ip default-gateway 10.1.1.1
ip classless
no ip http server
!
line con 0
exec-timeout 0 0
transport input none
line vty 0 4
login
!
end
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MGCP Configuration Examples
Configuring the Cisco 2620 as an RGW: Example
The following example illustrates a platform that does not support FGD-OS calls.
version 12.2
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname D
!
memory-size iomem 10
ip subnet-zero
!
mgcp
mgcp call-agent 172.20.5.20
mgcp package-capability gm-package
mgcp package-capability dtmf-package
mgcp package-capability line-package
mgcp package-capability rtp-package
mgcp default-package line-package
cns event-service server
!
voice-port 1/0/0
!
voice-port 1/0/1
!
dial-peer voice 1 pots
application MGCPAPP
port 1/0/0
!
dial-peer voice 2 pots
application MGCPAPP
port 1/0/1
!
interface Ethernet0/0
no ip address
no ip directed-broadcast
shutdown
!
interface Serial0/0
no ip address
no ip directed-broadcast
no ip mroute-cache
shutdown
no fair-queue
!
interface Ethernet0/1
ip address 172.20.5.25 255.255.255.0
no ip directed-broadcast
!
interface Serial0/1
no ip address
no ip directed-broadcast
shutdown
!
ip default-gateway 209.165.202.130
ip classless
ip route 209.165.200.225 255.255.255.224 Ethernet0/1
no ip http server
!
line con 0
exec-timeout 0 0
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MGCP Configuration Examples
transport input none
line aux 0
line vty 0 4
login
!
end
Configuring the Cisco Voice Gateway 200 as an RGW: Example
The following example illustrates the configuration of the Cisco Voice Gateway 200 as an RGW.
version 12.2
no service single-slot-reload-enable
no service pad
service timestamps debug datetime msec
service timestamps log uptime
no service password-encryption
!
hostname MainVG200
!
ip subnet-zero
no ip finger
ip host dirt 172.16.1.129
!
mgcp
mgcp call-agent 172.20.71.44
call rsvp-sync
!
ccm-manager switchback immediate
ccm-manager redundant-host 172.20.71.47
ccm-manager mgcp
!
interface FastEthernet0/0
ip address 172.21.10.14 255.255.255.0
duplex auto
speed auto
!
ip default-gateway 172.21.10.1
ip classless
ip route 0.0.0.0 0.0.0.0 FastEthernet0/0
ip route 172.16.0.0 255.255.0.0 172.20.82.1
no ip http server
!
access-list 199 permit udp any any range 16384 32766
access-list 199 permit ip host 10.51.26.6 any
access-list 199 permit ip host 10.51.16.7 any
queue-list 2 protocol ip 2 list 199
queue-list 2 default 5
queue-list 2 queue 2 byte-count 2880 limit 16
queue-list 2 queue 5 limit 1
priority-list 1 protocol ip high list 199
priority-list 1 default low
!
voice-port 1/0/0
!
voice-port 1/0/1
!
voice-port 1/1/0
!
voice-port 1/1/1
!
dial-peer voice 1 pots
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!
dial-peer voice 111 pots
application mgcpapp
port 1/1/1
!
gateway
!
line con 0
exec-timeout 0 0
transport input none
line aux 0
line vty 0 4
password lab
login
!
end
Cisco IOS Voice, Video, and Fax Configuration Guide
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H.323 Applications
This chapter provides an overview of the H.323 standard from the International Telecommunication
Union Telecommunication Standardization Sector (ITU-T), of the Cisco H.323-compliant gatekeeper, of
the Cisco H.323-compliant gateway, and of the Cisco H.323-compliant features. Cisco IOS software
complies with the mandatory requirements and several of the optional features of the H.323 Version 2
specification. The chapter contains the following sections:
•
The H.323 Standard, page 200
•
H.323 Feature Overview, page 211
•
H.323 Restrictions, page 235
•
H.323 Prerequisite Tasks, page 237
•
H.323 Configuration Task List, page 238
Refer to the ITU-T H.323 standard for more in-depth information about the overall H.323 standard.
For a complete description and for examples of configuring Cisco gatekeepers, see the chapter
“Configuring H.323 Gatekeepers and Proxies.”
For a complete description and for examples of configuring Cisco gateways, see the chapter
“Configuring H.323 Gateways.”
For more information on configuring Cisco H.323 features, see the “MGCP and Related Protocols,”
“Configuring SIP,” “Voice over IP Overview,” and “Dial Plans, Dial Peers, and Digit Manipulation”
chapters. For general information regarding the H.323 Standard, refer to the ITU-T H.323 specifications.
For a more complete description of the H.323-compliant gatekeeper and H.323 Version 2 standard
support upgrade commands used in this chapter, refer to the Cisco IOS Voice, Video, and Fax Command
Reference. To locate documentation for other commands that appear in this chapter, use the command
reference master index or search online.
To identify the hardware platform or software image information associated with a feature in this
chapter, use the Feature Navigator on Cisco.com to search for information about the feature or refer to
the software release notes for a specific release. For more information, see the “Identifying Supported
Platforms” section in the “Using Cisco IOS Software” chapter.
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H.323 Applications
The H.323 Standard
The H.323 Standard
The H.323 standard provides for sending and receiving audio, video, and data on an IP-based
internetwork. The following sections provide a basic overview of network components and how they
relate to each other:
•
H.323 Terminals, page 201
•
H.323 Gateways, page 201
•
Configuring ISDN Redirect Number Support, page 201
•
H.323 Proxies, page 202
•
H.323 Gatekeepers, page 202
•
Gatekeeper Zones, page 202
•
MCUs, page 202
•
How Terminals, Gatekeepers, and Proxies Work Together, page 203
•
How Terminals, Gatekeepers, and Gateways Work Together, page 205
•
How Terminals, Gatekeepers, Proxies, and MCUs Work Together, page 206
•
Call Signaling Procedures, page 209
Figure 39 shows a typical H.323 network.
Figure 39
Gatekeeper in an H.323 Network
Cisco gatekeeper
MCU
H.323 terminal
H.323 terminal
Corporate LAN
Router
Cisco
proxy
Gateway
H.320 terminal
(over ISDN)
Internet
Real-time
network
Telephone
network
H.324 terminal
(over POTS)
H.323 terminal
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H.323 Applications
The H.323 Standard
H.323 Terminals
An H.323 terminal is an endpoint in the network that provides for real-time, two-way communications
with another H.323 terminal, gateway, or multipoint control unit (MCU). The communications consist
of control, indications, audio, moving color video pictures, or data between the two terminals. A terminal
may provide audio only; audio and data; audio and video; or audio, data, and video. The terminal can be
a computer-based video conferencing system or other device.
A gatekeeper supports a broad variety of H.323 terminal implementations from many different vendors.
These terminals must support the standard H.323 Registration, Admission, and Status (RAS) protocol to
function with the gatekeeper.
H.323 Gateways
An H.323 gateway is an endpoint on the LAN that provides real-time communications between H.323
terminals on the LAN and other ITU terminals on a WAN or to other H.323 gateways.
Gateways allow H.323 terminals to communicate with devices that are running other protocols. They
provide protocol conversion between the devices that are running different types of protocols. For
example, Figure 40 shows a gateway between an H.323 terminal and a non-H.323 terminal.
Figure 40
Gateway Between an H.323 Terminal and an H.320 Terminal
H.323 gateway
H.323
endpoint
Protocol
translation
and media
transcoding
Non-H.323
endpoint
H.320 terminal
11025
H.323 terminal
Configuring ISDN Redirect Number Support
Voice over IP (VoIP) supports the redirecting call feature of the VoIP gateway for ISDN calls. The
redirecting number is an optional field of the Q.931 setup message.
When a local exchange carrier (LEC) switch detects an incoming call that is destined for a busy or
nonanswering party, the switch formulates a Q.931 setup message with the redirecting number field set
to the original destination number and sends it to the gateway. The called party number of the setup
message will be set to one of the destination number (Dialed Number Identification Service [DNIS])
access numbers of the gateway.
If a redirect number is present on an incoming call, it is used in place of the DNIS. To configure ISDN
redirect number support, see the “Dial Plans, Dial Peers, and Digit Manipulation” chapter.
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The H.323 Standard
H.323 Proxies
H.323 proxies are special types of gateways that relay H.323 calls to another H.323 endpoint. They can
be used to isolate sections of an H.323 network for security purposes, to manage quality of service
(QoS), or to perform special application-specific routing tasks.
H.323 Gatekeepers
An H.323 gatekeeper is an H.323 entity on the LAN that provides address translation and that controls
access to the LAN for H.323 terminals, gateways, and MCUs.
Gatekeepers are optional nodes that manage endpoints in an H.323 network. The endpoints communicate
with the gatekeeper using the RAS protocol.
Endpoints attempt to register with a gatekeeper on startup. When they wish to communicate with another
endpoint, they request admission to initiate a call using a symbolic alias for the endpoint, such as an
E.164 address or an e-mail address. If the gatekeeper decides that the call can proceed, it returns a
destination IP address to the originating endpoint. This IP address may not be the actual address of the
destination endpoint, but it may be an intermediate address, such as the address of a proxy or a
gatekeeper that routes call signaling.
Note
Although the gatekeeper is an optional H.323 component, it must be included in the network if
proxies are used.
Gatekeeper Zones
An H.323 endpoint is an H.323 terminal, gateway, or MCU. An endpoint can call and be called.
H.323 endpoints are grouped into zones. Each zone has one gatekeeper that manages all the endpoints
in the zone. A zone is an administrative convenience similar to a Domain Name System (DNS) domain.
(Because a zone is, by definition, the area of control of a gatekeeper, you will find the terms “zone name”
and “gatekeeper name” used synonymously in this chapter.)
Note
The maximum number of local zones defined in a gatekeeper should not exceed 100.
MCUs
An MCU is an endpoint on the network that allows three or more endpoints to participate in a multipoint
conference. It controls and mixes video, audio, and data from endpoints to create a robust multimedia
conference. An MCU may also connect two endpoints in a point-to-point conference, which may later
develop into a multipoint conference.
Note
Some terminals have limited multipoint control built into them. These terminals may not require an
MCU that includes all the functionality mentioned.
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The H.323 Standard
How Terminals, Gatekeepers, and Proxies Work Together
When endpoints are brought online, they first attempt to discover their gatekeeper. They discover their
gatekeeper either by sending multicast a discovery request or by being configured with the address and,
optionally, with the name of the gatekeeper and by sending a unicast discovery request. Following
successful discovery, each endpoint registers with the gatekeeper. The gatekeeper keeps track of which
endpoints are online and available to receive calls.
There are three ways to set up calls between various endpoints, as described in the following sections:
•
Interzone Call with Proxy, page 204
•
Interzone Call Without Proxy, page 203
•
Interzone Call with Proxy, page 204
Intrazone Call
Intrazone calls occur within the same zone.
If terminal TA1 wants to make an intrazone call to terminal TB1 in Zone 1, the following sequence of
events occurs:
1.
TA1 asks GK1 for permission to call TB1.
2.
GK1 returns the address of TB1 to TA1.
3.
TA1 then calls TB1.
Figure 41 illustrates these events.
Figure 41
Intrazone Call
Zone 1
3
TA1
2
GK1
11019
1
TB1
Interzone Call Without Proxy
Interzone calls occur between two or more zones.
If terminal TA1 in Zone 1 wants to call terminal TA2 in Zone 2 without the use of a proxy, the following
sequence of events occurs:
1.
TA1 asks GK1 for permission to call TA2.
2.
TA2 is not in the GK1 zone. GK1 locates GK2 as the TA2 gatekeeper. GK1 then asks GK2 for the
TA2 address.
3.
GK2 returns the TA2 address to GK1.
4.
GK1 returns the address to TA1.
5.
TA1 calls TA2.
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Figure 42 illustrates these events.
Interzone Call Without Proxy
Zone 2
Zone 1
TB1
5
TA1
1
P1
4
TA2
2
GK2
GK1
3
11020
Figure 42
Interzone Call with Proxy
One reason for using a proxy is to isolate addressing information in one zone from another. When such
isolation is desired, zones are configured as inaccessible on the gatekeepers. (Other reasons for using
proxies are discussed later in this document.)
If terminal TA1 in Zone 1 wants to call terminal TA3 in Zone 3, the following sequence of events occurs:
1.
TA1 asks GK1 for permission to call TA3.
2.
GK1 locates GK3 as the TA3 gatekeeper. GK1 asks GK3 for the TA3 address.
3.
GK3 responds with the P3 address instead of the TA3 address, to hide the TA3 identity.
4.
GK1 knows that to get to P3, the call must go through P1. So GK1 returns the P1 address to TA1.
5.
TA1 calls P1.
6.
P1 consults GK1 to discover the true destination of the call (which is TA3 in this example).
7.
GK1 instructs P1 to call P3.
8.
P1 calls P3.
9.
P3 consults GK3 for the true destination, which is TA3.
10. GK3 gives the TA3 address to P3.
11. P3 completes the call to TA3.
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Figure 43 illustrates these events.
Figure 43
Interzone Call with Proxy
Zone 1
TA1
5
1
4
TB1
6
GK1
P1
7
3
8
9
P3
11
GK3
10
TA3
11021
2
Zone 3
How Terminals, Gatekeepers, and Gateways Work Together
Gateways provide protocol conversion between terminals that run different types of protocols. Gateways
communicate with gatekeepers using the RAS protocol. The gatekeeper maintains resource availability
information, which it uses to select the appropriate gateway during the admission of a call. In Figure 44,
the following conditions exist:
•
TA1 is an H.323 terminal that is registered to GK1.
•
GW1 is an H.323-to-H.320 gateway that is registered to GK1.
•
TA2 is an H.320 terminal.
Figure 44 illustrates these events.
Intrazone Call Through Gateway
Zone 1
H.323
TA1
3
GW1
1
GK1
2
4
TA2
11017
Figure 44
H.320
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The H.323 Standard
A call from TA1 to TA2 is set up as follows:
1.
TA1 asks GK1 for permission to connect to the TA2 E.164 address.
2.
The gatekeeper looks through its local registrations and does not find any H.323 terminals that are
registered with that E.164 address, so the gatekeeper assumes that it is an H.320 terminal that is
outside the scope of H.323. The gatekeeper instructs TA1 to connect to the GW1 IP address.
3.
TA1 connects to GW1.
4.
GW1 completes the call to TA2.
A call from TA2 to TA1 is set up as follows:
1.
TA2 calls GW1 and provides the TA1 E.164 address as the final destination.
2.
GW1 sends a message to GK1 asking to connect to that address.
3.
GK1 gives GW1 the address of TA1.
4.
GW1 completes the call with TA1.
Figure 45 illustrates these events.
Figure 45
Gateways Provide Translation Between Terminal Types
GK1
3
2
4
1
GW1
11018
H.323
TA1
TA2
H.320
How Terminals, Gatekeepers, Proxies, and MCUs Work Together
When MCUs are brought online, they first attempt to discover their gatekeeper. As with terminals and
proxies, MCUs discover their gatekeeper either by multicasting a discovery request or by being
configured with the name and address of the gatekeeper and unicasting a discovery request. Following
successful discovery, the MCU registers with the gatekeeper. The gatekeeper keeps track of which
endpoints are online and available to receive calls.
There are three ways to set up an MCU conference call, as described in the following sections:
•
Intrazone MCU Conference Call, page 207
•
Interzone MCU Conference Call Without Proxy, page 207
•
Interzone MCU Conference Call with Proxy, page 208
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The H.323 Standard
Intrazone MCU Conference Call
An MCU conference in Zone 1 is created with the conference ID CompanyMeeting. The MCU
reregisters itself with the gatekeeper, with the new conference ID appended to its list of existing aliases.
If terminals TA1, TA2, and TA3 in Zone 1 want to join CompanyMeeting, the following sequence of
events occurs:
1.
TA1, TA2, and TA3 join the conference by asking GK1 for permission to call the given conference
ID.
2.
GK1 returns the address of the MCU to TA1, TA2, and TA3.
3.
TA1, TA2, and TA3 then call the MCU.
Figure 46 illustrates these events.
Figure 46
Intrazone Call with MCU
Zone 1
GK1
1
2
2
1
1
2
TA2
TA1
TA3
3
3
MCU
11022
3
Interzone MCU Conference Call Without Proxy
The MCU in Zone 2 creates a conference with conference ID [email protected] The MCU
reregisters itself with GK2, with the new conference ID appended to its list of existing aliases. Terminals
TA1, TB1, and TC1 in Zone 1 want to join the MCU conference call with the conference ID
[email protected] in Zone 2. The following sequence of events occurs:
1.
TA1, TB1, and TC1 ask GK1 for permission to join the conference.
2.
GK1 locates GK2 for the remote zone that contains conference [email protected] using
DNS or information configured on GK1. GK1 sends a request to GK2 to recover the MCU address.
3.
GK2 gives the MCU address to GK1.
4.
GK1 gives the MCU address to TA1, TB1, and TC1, and it instructs these endpoints to set up the
call with the MCU.
5.
TA1, TB1, and TC1 then call the MCU.
Figure 47 illustrates these events.
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The H.323 Standard
Interzone MCU Conference Call Without Proxies
Zone 1
5
TA1
4
1
4
GK1
Zone 2
1
4
5
MCU
TB1
5
1
GK2
TC1
2
11023
Figure 47
3
Interzone MCU Conference Call with Proxy
One main reason for using a proxy is to isolate addressing information in one zone from another. When
such isolation is desired, zones are configured to be inaccessible on the gatekeepers.
The MCU in Zone 3 creates a conference with the conference ID [email protected] The
MCU reregisters itself with the gatekeeper, using the new conference ID appended to its list of existing
aliases. Terminals TA1, TB1, and TC1 in Zone 1 want to join the MCU conference with the conference
ID [email protected] in Zone 3. The following sequence of events occurs:
1.
TA1, TB1, and TC1 ask GK1 for permission to join the conference [email protected]
2.
GK1 locates GK3 for the remote zone that contains conference [email protected]
GK1 asks GK3 for the MCU address.
3.
GK3 responds with the PX3 address instead of the MCU address. GK1 knows that to get to PX3 the
call should go through P1.
4.
GK1 gives the P1 address to TA1, TB1, and TC1.
5.
TA1, TB1, and TC1 call P1.
6.
P1 consults GK1 to discover the true call destination, which is [email protected] in this
example.
7.
GK1 instructs P1 to call P3.
8.
P1 calls P3.
9.
P3 consults with GK3 to discover the true call destination, which is [email protected]
in this example.
10. GK3 gives the MCU address to PX3.
11. P3 completes the call with the MCU.
Figure 48 illustrates these events.
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The H.323 Standard
Figure 48
Interzone MCU Conference Call with Proxy
Zone 3
Zone 1
8
P1
P3
6
5
9
TA1
1
TB1
1
4
TC1
10
MCU
1
4
4
GK3
GK1
2
11024
5
7
11
5
3
Call Signaling Procedures
Two important phases of H.323 call signaling are call setup and call termination. The following two
examples demonstrate the call setup and call termination processes in relation to gatekeepers and
gateways.
Call Setup—Both Gateways Registered to the Same Gatekeeper
In Figure 49, both gateways are registered to the same gatekeeper, and the gatekeeper has chosen direct
call signaling. Gateway 1 (the calling gateway) initiates the admission request (ARQ) (1)/admission
confirmation (ACF) (2) exchange with that gatekeeper. The gatekeeper returns the call signaling channel
address of Gateway 2 (the called gateway) in the ACF. Gateway 1 then sends the setup (3) message to
Gateway 2 using that transport address. If Gateway 2 wishes to accept the call, it initiates an ARQ
(5)/ACF (6) exchange with the gatekeeper. Gateway 2 sends an alerting (7) message to Gateway 1. (If
Gateway 2 receives an admission reject [ARJ] (6) message instead of an ACF message, it sends a release
complete message to Gateway 1 instead of the alerting message.) Gateway 2 responds with the connect
(8) message, which contains an H.245 control channel transport address for use in H.245 signaling.
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Figure 49
Both Gateways Registered to the Same Gatekeeper
Gateway 1
Gatekeeper 1
Gateway 2
ARQ (1)
ACF/ARJ (2)
Setup (3)
Call proceeding
(4)
ARQ (5)
ACF/ARJ (6)
RAS messages
Call signaling messages
56581
Aler ting (7)
Connect (8)
Call Termination
Either gateway may terminate a call in one of the following ways:
1.
It discontinues transmission of video at the end of a complete picture and then closes all logical
channels for video.
2.
It discontinues transmission of data and then closes all logical channels for data.
3.
It discontinues transmission of voice and then closes all logical channels for voice.
4.
It transmits the H.245 endSessionCommand message in the H.245 control channel, indicating to the
far end that it wishes to disconnect the call and then discontinues H.245 message transmission.
5.
It waits to receive the endSessionCommand message from the other gateway and then closes the
H.245 control channel.
6.
If the call signaling channel is open, a release complete message is sent and the channel is closed.
7.
The gateway clears the call by using the procedures defined below.
An endpoint receiving an endSessionCommand message without first having transmitted it carries out
steps 1 and 7 above, except that in Step 5, the gateway waits for the endSessionCommand message from
the first endpoint.
Terminating a call may not terminate a conference; a conference may be explicitly terminated using an
H.245 message (dropConference). In this case, the gateways wait for the multipoint controller to
terminate the calls as described.
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Call Clearing with a Gatekeeper
In networks that contain a gatekeeper, the gatekeeper needs to know about the release of bandwidth.
After performing steps 1 to 6 above, each endpoint transmits an H.225.0 disengage request (DRQ)
message (3) to its gatekeeper (shown in Figure 50). The gatekeeper responds with a disengage confirm
(DCF) message (4). After sending the DRQ message, the endpoints do not send further unsolicited
information request response (IRR) messages that relate to that call to the gatekeeper. At this point, the
call is terminated. Figure 50 shows the direct call model.
The DRQ and DCF messages are sent on the RAS channel.
Figure 50
Call Termination Direct Call Model
Gateway 1
Gatekeeper 1
Gateway 2
EndSessionCom
mand (1)
)
Command (1
EndSession
Release Compl
ete (2)
DRQ (3)
DCF (4)
DRQ (3)
DCF (4)
56580
RAS messages
Call signaling messages
H.245 messages
H.323 Feature Overview
This section includes the following subsections:
•
Source Call Signal Address, page 212
•
H.323 Version 2 Support, page 213
– Lightweight Registration, page 214
– Improved Gateway Selection Process, page 214
– Gateway Resource Availability Reporting, page 215
– Support for Single-Proxy Configurations, page 215
– Registration of E.164 Addresses for Gateway-Attached Devices, page 215
– Tunneling of Redirecting Number Information Element, page 215
– DTMF Relay, page 216
– H.245 Tunneling of DTMF Relay in Conjunction with Fast Connect, page 217
– Translation of FXS Hookflash Relay, page 217
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H.323 Feature Overview
– H.235 Security, page 219
– GKTMP and RAS Messages, page 219
– RAS Message Fields, page 220
– Multizone Features, page 224
– Codec Negotiation, page 225
– Supported Codecs, page 225
– H.245 Empty Capabilities Set, page 226
•
H.323 Version 2 Fast Connect, page 226
•
H.450.2 Call Transfer, page 227
•
H.450.3 Call Deflection, page 228
•
Gateway Support for Alternate Endpoints, page 228
•
Gatekeeper C Code Generic API for GKTMP in a UNIX Environment, page 228
•
Gateway Support for a Network-Based Billing Number, page 228
•
Gateway Support for Voice-Port Description, page 229
•
H.323 Signaling, page 229
– In-Band Tones and Announcements, page 229
– End-to-End Alerting, page 231
– Cut-Through of Voice Path, page 231
– H.245 Initiation, page 231
– Overlap Dialing, page 232
•
Configurable Timers in H.225.0, page 232
•
Answer Supervision Reporting, page 232
•
Gateway-to-Gatekeeper Billing Redundancy, page 233
•
Ecosystem Gatekeeper Interoperability, page 233
– AltGKInfo in GRJ Messages, page 234
– AltGKInfo in RRJ Messages, page 234
Source Call Signal Address
Source call signal address allows a source call-signal address field to be included in the ARQ.
Previously, in the Cisco IOS implementation of H.323 gateway software, if the terminating gateway was
registered to an H.323 gatekeeper and used RAS, the ARQ message sent for each incoming call did not
contain the H.225.0 source call signal address (CSA). The source CSA is an optional parameter in the
ARQ message. The source CSA is also an optional parameter in the H.225.0 call setup message sent by
the originating endpoint.
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Source call signal address also allows for the source CSA parameter to be included in the ARQ message,
as illustrated by the message sequence shown in Figure 51.
Figure 51
EP1
Source Call Signal Message Sequence
ARQ
GK1
EP2
ACF
Setup Message with SrcCSA1 (SrcCSA = IP Addr. of EP1 + dynamic Port)
Call Proceeding
ACF
GK2
28906
ARQ with SrcCSA1
In the message sequence shown in Figure 51, the ARQ messages are enhanced to send the source CSA.
The originating gateway (EP1) sends the H.225.0 setup message to the destination gateway. The setup
message contains the source CSA parameter, which is the combination of the IP address of the originator
and the dynamic TCP port number used or obtained for the H.225.0 call signaling channel. If the
terminating gateway (EP2) accepts the call upon receipt of the setup message, the gateway sends an ARQ
message to the gatekeeper. The terminating gateway retrieves the source CSA parameter sent by the
originating gateway in the setup message. It then sends an ARQ message to the gatekeeper with the
source CSA parameter. The CSA parameter is optional and has the same value as the source CSA in the
received setup message. If the setup message does not contain the source CSA parameter, the terminating
gateway determines the source CSA by using the H.225.0 call-signaling TCP socket connection of the
peer endpoint, which it uses in the ARQ message.
If the originating gateway is registered to a gatekeeper and RAS is used as the session target, the
originating gateway also sends an ARQ message. This ARQ does not include the optional source CSA
parameter.
H.323 Version 2 Support
Cisco software complies with the mandatory requirements and several of the optional features of the
H.323 Version 2 specification. Cisco H.323 Version 2 software enables gatekeepers, gateways, and
proxies to send and receive all the required fields in H.323 Version 2 messages. Cisco H.323 Version 2
features include the following:
•
Lightweight Registration, page 214
•
Improved Gateway Selection Process, page 214
•
Gateway Resource Availability Reporting, page 215
•
Support for Single-Proxy Configurations, page 215
•
Registration of E.164 Addresses for Gateway-Attached Devices, page 215
•
Tunneling of Redirecting Number Information Element, page 215
•
DTMF Relay, page 216
•
H.245 Tunneling of DTMF Relay in Conjunction with Fast Connect, page 217
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•
Translation of FXS Hookflash Relay, page 217
•
H.235 Security, page 219
•
GKTMP and RAS Messages, page 219
•
RAS Message Fields, page 220
•
Multizone Features, page 224
•
Codec Negotiation, page 225
•
Supported Codecs, page 225
•
H.245 Empty Capabilities Set, page 226
Lightweight Registration
Before the release of its H.323 Version 2 software, Cisco gateways reregistered with the gatekeeper every
30 seconds. Each registration renewal used the same process as the initial registration, even though the
gateway was already registered with the gatekeeper. These registration renewals generated considerable
overhead at the gatekeeper.
Cisco H.323 Version 2 software defines a lightweight registration procedure that still requires the full
registration process for initial registration but that uses an abbreviated renewal procedure to update the
gatekeeper and minimize overhead.
Lightweight registration requires each endpoint to specify a time-to-live (TTL) value in its registration
request (RRQ) message. When a gatekeeper receives an RRQ message with a TTL value, it returns an
updated TTL timer value in a registration confirmation (RCF) message to the endpoint. Shortly before
the TTL timer expires, the endpoint sends an RRQ message with the KeepAlive field set to TRUE, which
refreshes the existing registration.
It is not required that an H.323 Version 2 endpoint indicate a TTL in its registration request. If the
endpoint does not indicate a TTL, the gatekeeper assigns one and sends it to the gateway in the RCF
message. No configuration changes are permitted during a lightweight registration, so all fields other
than the endpointIdentifier, gatekeeperIdentifier, tokens, and TTL are ignored. In the case of H.323
Version 1 endpoints that cannot process the TTL field in the RCF, the gatekeeper probes the endpoint
with information requests (IRQs) for a predetermined grace period to see if the endpoint is still alive.
Improved Gateway Selection Process
Cisco H.323 Version 2 software improves the gateway selection process as follows:
•
When more than one gateway is registered in a zone, the updated zone prefix command allows
selection priorities to be assigned to these gateways on the basis of the dialed prefix.
•
Gateway resource reporting allows the gateway to notify the gatekeeper when H.323 resources are
getting low. The gatekeeper uses this information to determine which gateway it will use to complete
a call.
The gatekeeper maintains a separate gateway list, ordered by priority, for each of its zone prefixes. If a
gateway does not have an assigned priority for a zone prefix, it defaults to priority 5, which is the median.
To explicitly bar the use of a gateway for a zone prefix, the gateway must be defined as having a priority
0 for that zone prefix.
When selecting gateways, the gatekeeper identifies a target pool of gateways by performing a longest
zone prefix match; then it selects from the target pool according to priorities and resource availability.
If all high-priority gateways are busy, a low-priority gateway might be selected.
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Gateway Resource Availability Reporting
To allow gatekeepers to make intelligent call routing decisions, the gateway reports the status of its
resource availability to its gatekeeper. Resources that are monitored are digital signal level 0 (DS0)
channels and digital signal processor (DSP) channels. In Cisco IOS Release 12.1, this feature is available
only on the AS5300 platform.
The gateway reports its resource status to the gatekeeper using the RAS Resource Availability Indication
(RAI). When a monitored resource falls below a configurable threshold, the gateway sends a RAI to the
gatekeeper indicating that the gateway is almost out of resources. When the available resources then
cross over another configurable threshold, the gateway sends a RAI indicating that the resource depletion
condition no longer exists. Resource reporting thresholds are configured by using the resource
threshold command. The upper and lower thresholds are separately configurable to prevent the gateway
from operating sporadically because of the availability or lack of resources.
Support for Single-Proxy Configurations
Cisco H.323 Version 2 software supports single-proxy, two-proxy, and no-proxy calls. Proxies can also
be independently configured to meet the needs of inbound and outbound call scenarios.
Registration of E.164 Addresses for Gateway-Attached Devices
If phones are connected directly to the gateway, the Cisco H.323 Version 2 gateway allows fully qualified
E.164 numbers to be registered with the gatekeeper. When configuring the gateway, use the register
command to register these E.164 numbers.
Tunneling of Redirecting Number Information Element
An incoming PRI setup message may contain either a Redirecting Number (RDN) Information Element
(IE) or an Original Called Number (OCN) IE. These IEs indicate that the call has been redirected
(forwarded) and that each message contains the following:
•
The destination number (DN) that was originally called
•
The reason for the call being redirected
•
Other related information
OCN IE is a Nortel variant of the RDN IE.
The H.323 Version 2 gateway passes the entire RDN or OCN IE from an incoming PRI message into the
H.225.0 setup message. The IE is encapsulated in the nonStandardData field within the user-to-user
information element (UUIE) of the H.225.0 setup message. The nonStandardData field can contain the
encapsulated RDN or OCN IE and a tunneled global, signaling, and control standard QSIG message, or
it can contain only the OCN or RDN. Cisco and other third-party H.323 endpoints can access the
redirected information by decoding the nonStandardData field. In accordance with the H.225.0
specification, the nonStandardData is ignored by third-party endpoints and causes no interoperability
problems.
For redirected PRI calls that are routed to a Cisco gateway, that are sent using H.323 to another Cisco
gateway, and that exit the gateway using PRI, the RDN/OCN IE is tunneled from the source gateway to
the destination gateway. The incoming PRI setup message is tunneled through H.225.0 and is encoded
into the outgoing PRI setup message by the destination gateway.
Tunneling the RDN or OCN IE is important for applications such as Unified Messaging servers that need
to know the telephone number that was originally dialed so as to access the correct account information.
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DTMF Relay
Dual-Tone Multifrequency (DTMF) is the tone generated on a touchtone phone when the keypad digits
are pressed. During a call, DTMF may be entered to access interactive voice response (IVR) systems,
such as voice mail and automated banking services.
In previous releases of Cisco IOS software, DTMF is transported in the same way as voice. This
approach can result in problems accessing IVR systems. Although DTMF is usually transported
accurately when using high-bit-rate voice codecs such as G.711, low-bit-rate codecs such as G.729 and
G.723.1 are highly optimized for voice patterns and tend to distort DTMF tones. As a result, IVR systems
may not correctly recognize the tones.
DTMF relay solves the problem of DTMF distortion by transporting DTMF tones “out-of-band” or
separate from the encoded voice stream. Cisco H.323 Version 2 software introduces the following three
options to the existing dtmf-relay command for sending DTMF tones out-of-band:
•
A Cisco proprietary RTP-based method (dtmf-relay cisco-rtp command)
•
H.245 signal (dtmf-relay h245-signal command)
•
H.245 alphanumeric (dtmf-relay h245-alphanumeric command)
If none of these options is selected, DTMF tones are transported in-band and encoded in the same way
as voice traffic.
The dtmf-relay cisco-rtp command sends DTMF tones in the same Real-Time Protocol (RTP) channel
as voice. However, the DTMF tones are encoded differently from the voice samples and are identified
by a different RTP payload type code. This method accurately transports DTMF tones, but because it is
proprietary, it requires the use of Cisco gateways at both the originating and terminating endpoints of
the H.323 call.
The dtmf-relay h245-signal and dtmf-relay h245-alphanumeric commands are modes of DTMF
transport defined by the ITU H.245 standard. These methods separate DTMF digits from the voice
stream and send them through the H.245 signaling channel instead of the RTP channel. The tones are
transported in H.245 user input indication messages. The H.245 signaling channel is a reliable channel,
so the packets that transport the DTMF tones are guaranteed to be delivered. However, because of the
overhead that is generated by using a reliable protocol, and depending on network congestion conditions,
the DTMF tones may be slightly delayed. This delay is not known to cause problems with existing
applications.
The dtmf-relay h245-signal command relays a more accurate representation of a DTMF digit than does
the dtmf-relay h245-alphanumeric command because tone duration information is included along with
the digit value. This information is important for applications requiring that a key be pressed for a
particular length of time. For example, one popular calling card feature allows the caller to terminate an
existing call by pressing the # key for more than 2 seconds and then making a second call without having
to hang up in between. This feature is beneficial because the access number and personal identification
number (PIN) code do not need to be dialed again. Outside-line access charges, which are common at
hotels, may also be avoided.
The dtmf-relay h245-alphanumeric command simply relays DTMF tones as ASCII characters. For
instance, the DTMF digit 1 is transported as the ASCII character 1. There is no duration information
associated with tones in this mode. When the Cisco H.323 gateway receives a DTMF tone using this
method, it will generate the tone on the Public Switched Telephone Network (PSTN) interface of the call
using a fixed duration of 500 milliseconds. All systems that are H.323 Version 2-compliant are required
to support the dtmf-relay h245-alphanumeric command, but support of the dtmf-relay h245-signal
command is optional.
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The ability of a gateway to receive DTMF digits in a particular format and the ability to send digits in
that format are independent functions. No configuration is necessary to receive DTMF digits from
another H.323 endpoint using any of the methods described. The Cisco H.323 Version 2 gateway is
capable of receiving DTMF tones transported by any of these methods at all times.
However, to send digits out-of-band using one of these methods, two conditions must be met:
•
The chosen method of DTMF relay must be enabled during dial-peer configuration using the
dtmf-relay command.
•
The peer (the other endpoint of the call) must indicate during call establishment that it is capable of
receiving DTMF in that format.
More than one DTMF relay option may be enabled for a particular dial peer. If more than one option is
enabled, and if the peer indicates that it is capable of receiving DTMF in more than one of these formats,
the gateway will send DTMF using the method among the supported formats that it considers to be the
most preferred. The preferences are defined as follows:
•
dtmf-relay cisco-rtp (highest preference)
•
dtmf-relay h245-signal
•
dtmf-relay h245-alphanumeric
If the peer is not capable of receiving DTMF in any of the modes that were enabled, DTMF tones will
be sent in-band.
When the Cisco H.323 Version 2 gateway is involved in a call to a Cisco gateway that is running a version
of Cisco IOS software prior to Release 12.0(5)T, DTMF tones will be sent in-band because those systems
do not support DTMF relay.
See the “Configuration Task List” section in the “Configuring H.323 Gateways and Proxies” chapter for
an example of configuring DTMF relay.
H.245 Tunneling of DTMF Relay in Conjunction with Fast Connect
Through H.245 tunneling, H.245 messages are encapsulated within H.225.0 messages without using a
separate H.245 TCP connection. When tunneling is enabled, one or more H.245 messages can be
encapsulated in any H.225.0 message. H.245 tunneling is not supported as a stand-alone feature;
initiation of H.245 tunneling procedures can be initiated only by using the dtmf-relay command and
only from an active fast connect call. Furthermore, if dtmf-relay is configured on a Version 2 VoIP dial
peer and the active call has been established by using fast connect, tunneling procedures initiated by the
opposite endpoint are accepted and supported.
H.245 tunneling is backward compatible with H.323 Version 1 configurations.
Translation of FXS Hookflash Relay
A hookflash indication is a brief on-hook condition that occurs during a call. It is not long enough in
duration to be interpreted as a signal to disconnect the call. Create a hookflash indication by quickly
depressing and then releasing the hook on your telephone.
PBXs and telephone switches are frequently programmed to intercept hookflash indications and use
them as a way to allow a user to invoke supplemental services. For example, your local service provider
may allow you to enter a hookflash as a means of switching between calls if you subscribe to a call
waiting service.
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In the traditional telephone network, a hookflash results in a voltage change on the telephone line.
Because there is no equivalent of this voltage change in an IP network, the ITU H.245 standard defines
a message representing a hookflash. To send a hookflash indication using this message, an H.323
endpoint sends an H.245 user input indication message containing a “signal” structure with a value of
“!”. This value represents a hookflash indication.
Cisco H.323 Version 2 software includes limited support for relaying hookflash indications using the
H.245 protocol. H.245 user input indication messages containing hookflash indications that are received
on the IP call leg are forwarded to the plain old telephone service (POTS) call leg if the POTS interface
is Foreign Exchange Office (FXO). If the interface is not FXO, any H.245 hookflash indication that is
received is ignored. This support allows IP telephony applications to send hookflash indications to a
PBX through the Cisco gateway and thereby invoke the IOS supplementary services of the PBX if the
PBX supports access to those features using hookflash.
The gateway does not originate H.245 hookflash indications in this release. For example, it does not
forward hookflash indications from Foreign Exchange Station (FXS) interfaces to the IP network over
H.245.
The acceptable duration of a hookflash indication varies by equipment vendor and by country. Although
one PBX may consider a 250-millisecond on-hook condition to be a hookflash, another PBX may
consider this condition to be a disconnect. Therefore, the timing hookflash-out command allows the
administrator to define the duration of a hookflash signal generated on an FXO interface.
Figure 52 illustrates an FXS hookflash being translated to an H.245 user input.
Analog
phone
Translating an FXS Hookflash to an H.245 User Input
FXS
PBX
Cisco 2600
Cisco 2600
Ethernet
FXO
30764
Figure 52
In the Cisco H.323 Version 2 software, an FXS hookflash relay is generated only if the following two
conditions are met:
•
The other endpoint must support the reception of an H.245 hookflash and advertise this using the
“Receive User Input Capability” message during H.245 capabilities exchange.
•
The call must be established with either the h245-alphanumeric or h245-signal variant of the
dtmf-relay command.
This implies that the VoIP dial peer must be configured for dtmf-relay h245-alphanumeric or
h245-signal, but not cisco-rtp.
Enter the timing hookflash-input command on FXS interfaces to specify the maximum length in
milliseconds of a hookflash indication. If the hookflash lasts longer than the specified limit, then the
FXS interface processes the indication as an onhook.
The acceptable duration of a hookflash indication varies by equipment vendor and by country. One PBX
may consider a 250 milliseconds on-hook condition to be a hookflash; another PBX may consider this
condition to be a disconnect.
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H.235 Security
Security for RAS protocol signaling between H.323 endpoints and gatekeepers is enhanced in H.323
Version 2 software by including secure endpoint registration of the Cisco gateway to the Cisco
gatekeeper and secure per-call authentication. In addition, it provides for the protection of specific
messages related to Open Settlement Protocol (OSP) and to other messages as required via encryption
tokens. The authentication type is “password with hashing” as described in the ITU H.235 specifications.
Specifically, the encryption method is to use the MD5 algorithm, with password hashing. This
functionality is provided by the security token required-for command on the gatekeeper and the
security password command on the gateway.
The gatekeeper can interact with a RADIUS security server to perform the authentications. The gateway
can also authenticate an external application by using the Gatekeeper Transaction Message Protocol
(GKTMP) application programming interface (API).
Per-call authentication is accomplished by validating account and pin numbers that are entered by the
user connected to the calling gateway by using an IVR prompt.
The security mechanisms described above require the gateway and gatekeeper clocks to be synchronized
within 30 seconds of each other by using a Network Time Protocol (NTP) server.
GKTMP and RAS Messages
The GKTMP for the Cisco gatekeeper provides a transaction-oriented application protocol that allows
an external application to modify gatekeeper behavior by processing specified RAS messages.
A set of triggers can be specified that use RAS messages that can be recognized by the gatekeeper.
Triggers are specified filter conditions that must match each type of RAS message. The triggers can be
dynamically registered by using the external application, or this information can be configured by using
the command-line interface (CLI) on the gatekeeper.
When the gatekeeper receives a RAS message that meets the specified trigger conditions, it forwards the
message to the external application in a GKTMP message format. This message is text encoded and sent
over TCP. The external application can then modify fields in the message before returning it to the
gatekeeper for further processing, or it may return a RAS response to the gatekeeper to be forwarded to
the RAS client.
The following messages can be sent in GKTMP:
•
ACF—admission confirm
•
ARJ—admission reject
•
ARQ—admission request
•
LCF—location confirm
•
LRJ—location reject
•
LRQ—location request
•
RCF—registration confirm
•
RRJ—registration reject
•
RRQ—registration request
•
URQ—unregistration request
The application server interprets RAS messages in the following ways:
•
For RRQ and URQ, the application server performs gatekeeper authorization, storing endpoint RAS
gatekeeper IP addresses and maintaining gatekeeper resource control.
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Note
•
For ARQ and LRQ, the application server performs authorization and digit translation functions and
returns terminating IP addresses or a new E.164 address to the gatekeeper for reorigination by the
originating gateway.
•
For LCF and LRJ, the application server intercepts location responses from a distant gatekeeper and
modifies the message fields before responding to the originating gateway.
Cisco has developed an API that can be used to provide an interface to the Cisco gatekeeper. Refer
to the Cisco Gatekeeper External Interface Reference.
To configure the gatekeeper to receive trigger registrations from the external applications, specify the
registration port of the server using the server registration-port command. This command tells the
gatekeeper to listen for server connections.
You can also configure the gatekeeper to initiate the connection to a specified external application by
using the server trigger command to specify a set of static trigger conditions for a specified server. Only
one application server can be specified for each server trigger command. All RAS messages that do not
match the selection criteria for any external application are processed normally by the gatekeeper. The
show gatekeeper servers and debug gatekeeper servers commands can be entered to assist in the
configuration.
See the “Gatekeeper Transaction Message Protocol and RAS Messages Example” in the “Configuring
H.323 Gatekeepers and Proxies” chapter of this configuration guide.
RAS Message Fields
In support of the H.323 security and accounting features, fields have been added to several of the RAS
messages effective with Cisco IOS Release 12.0(7)T. In general, all the RAS messages sent by the
gateway, with the exception of the gateway request (GRQ), include authentication data in the
cryptoToken field. This section lists each of the messages that changed effective with Cisco IOS Release
12.0(7)T and describes the fields that have been added.
GRQ Message
When H.323 security is enabled on the gateway, the following fields are added to the GRQ message:
Field
Description
authenticationCapability
This field should have a value of pwdHash.
algorithmOIDs
The object ID for the MD5 algorithm. The object identifier (OID) used to
indicate MD5 will be {1 2 840 113549 2 5}.
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GCF Message
When H.323 security is enabled on the gateway, the following fields should be in the gateway
confirmation (GCF) message:
Field
Description
authenticationMode
This field should have a value of pwdHash.
algorithmOIDs
The object ID for the MD5 algorithm. The OID used to indicate MD5 will
be {1 2 840 113549 2 5}.
If the authenticationMode or the algorithm OIDs fields do not contain the values specified above, the
gatekeeper responds with a gatekeeper rejection (GRJ) message that contains a reject reason of
securityDenial. This prompts the gateway to resend the GRQ .
RRQ Message
If H.323 security is enabled on the gateway, the following fields are added to the RRQ message:
Field
Description
cryptoTokens
This field contains one of the cryptoToken types defined for the
CryptoH323Token field specified in H.225.0. Currently, the only type of
cryptoToken supported is cryptoEPPwdHash.
The following fields are contained within the cryptoEPPwdHash structure:
Field
Description
alias
The gateway alias, which is the H.323 ID of the gateway.
timestamp
The current time stamp.
token
The MD5 encoded PwdCertToken. This field contains the following:
timestamp—The same as the time stamp of cryptoEPPwdHash.
password—The password of the gateway.
generalID—The same gateway alias as the one included in the
cryptoEPPwdHash.
tokenID—The object ID.
ARQ Message
When H.323 security is enabled on the gateway, additional fields are included in the ARQ message. The
contents of the field depend on whether the ARQ message is sent from the source gateway or the
destination gateway.
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Source Gateway ARQ Message
If the ARQ message is sent from the source gateway, the following fields are included:
Field
Description
cryptoTokens
This field contains one of the cryptoToken types defined for the
CryptoH323Token field specified in H.225.0. Currently, the only type of
cryptoToken supported is cryptoEPPwdHash.
The following fields are contained within the cryptoEPPwdHash structure:
Field
Description
alias
The account number of the user or the H.323 ID of the gateway if endpoint
authentication is selected.
timestamp
The current time stamp.
token
The MD5 encoded PwdCertToken. This field contains the following:
•
timestamp—The same as the time stamp of cryptoEPPwdHash.
•
password—If “endpoint” is selected, this is the security password of
the gateway. Otherwise, it is the password or PIN of the user.
•
generalID—If “endpoint” is selected, this is the H.323 ID of the
gateway. Otherwise, it is the ID or account number of the user.
•
tokenID—The object ID.
Destination Gateway ARQ Message
If the ARQ message is sent from the destination gateway, the following fields are included:
Field
Description
cryptoTokens
This field contains one of the cryptoToken types defined for the
CryptoH323Token field specified in H.225.0. Currently, the only type of
cryptoToken supported is cryptoEPPwdHash.
The following fields are contained within the cryptoEPPwdHash structure:
Field
Description
alias
The alias (H.323 ID or E.164 address) of the destination gateway.
timestamp
The current time stamp.
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Field
Description
token
The MD5 encoded PwdCertToken. This field contains the following:
timestamp—The same as the time stamp of cryptoEPPwdHash.
password—The password of the destination gateway.
generalID—The same gateway alias as the one included in
cryptoEPPwdHash.
tokenID—The object ID.
ACF Message
If H.323 security is enabled on the gateway, the gatekeeper should include the billing-related information
from the nonStandardParameter field of the clearTokens structure. If the call is using a prepaid call
service, the clearTokens field should indicate the maximum call duration. In the case of prepaid call
service, the gateway will terminate the call if it exceeds the allowed time.
The following clearToken fields should be included in the ACF message:
Field
Description
nonStandard
The billing information for the call.
tokenOID
The generic billing object ID.
The following fields are contained within the nonStandardParameter structure:
Field
Description
nonStandardIdentifier
The generic billing object ID.
BillingInfo
The billing information. This field can contain the following:
•
bill_to—A string that identifies the subscriber that should be billed for
this call.
•
reference_id—A unique ID generated by the billing system.
•
billing_mode—Whether the call is being made using prepaid call
service (debit_mode) or not (credit_mode).
•
max_duration—The maximum duration allowed for the call. Used only
for prepaid call service.
•
balance—The account balance of the caller. For a billing mode of
credit_mode, this should be a negative value that represents the current
amount owed by the subscriber. Otherwise, this should be a positive
value that represents the credit remaining on the debit account of the
subscriber.
•
currency—The currency used in reporting the balance.
•
timezone—The time zone of the call, represented by a hexadecimal
string that indicates the difference in seconds between the location of
the caller and the Universal Time Coordinated (UTC).
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DRQ Message
The gateway sends a DRQ message when the call ends. If H.323 security is enabled on the gateway, the
call usage information is included in the DRQ message. The call usage information is sent in the
nonStandardParameter field of the ClearToken structure.
The following fields are contained within the nonStandardParameter structure:
Field
Description
duration
The duration of the call in seconds.
callLog
The call usage information. This field contains the following information:
•
DISCONNECT_REASON—The disconnect reason. Possible values
are as follows:
– DISCONNECT_NORMAL—The call ended normally.
– DISCONNECT_DISCONNECT—The call ended because of a
technical failure.
– DISCONNECT_ABANDONED—The call never took place; for
example, the remote phone was not answered.
– DISCONNECT_PREEMPT—The call was ended by the gateway.
This would be the disconnect reason issued if the call was ended
because the max_duration was exceeded.
•
DISCONNECT_STRING—A string that further describes the
disconnect reason.
•
TIME—The time at which the call started, indicated by a hexadecimal
string that represents the time, in seconds, since 00:00 January 1, 1970
UTC.
•
ORIGIN—Whether the call was inbound or outbound.
Multizone Features
Cisco multizone software enables the Cisco gateway to provide information to the gatekeeper using
additional fields in the RAS messages. The gatekeeper no longer terminates a call if it is unable to
resolve the destination E.164 phone number with an IP address.
Previously, the source gateway attempted to set up a call to a destination IP address as provided by the
gatekeeper in an admission confirm (ACF) message. If the gatekeeper was unable to resolve the
destination E.164 phone number to an IP address, the incoming call was terminated.
Multizone software allows a gatekeeper to provide additional destination information and modify the
destinationInfo field in the ACF message. The gateway will include the canMapAlias-associated
destination information in setting up the call to the destination gateway.
The gatekeeper indicates to the gateway that the call should be destined to a new E.164 number by
sending an ACF message with an IP address of 10.0.0.0 in the destCallSignalAddress field and the new
destination E.164 phone number in the destinationInfo field.
The gateway that receives such an ACF will fall back to routing the call on the basis of this new E.164
address and performing a relookup of the configured dial plan for the gateway. If the gateway routes the
call on the basis of the new E.164 address, the call might be routed back to the PSTN or to an H.323
endpoint.
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Codec Negotiation
Codec negotiation allows the gateway to offer several codecs during the H.245 capability exchange
phase and to ultimately settle on a single common codec during the call establishment phase. Offering
several codecs increases the probability of establishing a connection because there will be a greater
chance of overlapping voice capabilities between endpoints. Normally, only one codec can be specified
when a dial peer is configured, but codec negotiation allows a prioritized list of codecs associated with
a dial peer to be specified. During the call establishment phase the router will use the highest priority
codec from the list that it has in common with the remote endpoint. It will also adjust to the codec
selected by the remote endpoint so that a common codec is established for both the receive and send
voice directions.
When a call is originated, all the codecs associated with the dial peer are sent to the terminating endpoint
in the H.245 terminal capability set message. At the terminating endpoint, the gateway will advertise all
the codecs that are available in firmware in its terminal capability set. If there is a need to limit the codecs
advertised to a subset of the available codecs, a terminating dial peer must be matched that includes this
subset. The incoming called-number command in dial peer configuration mode can be used to force
this match.
Supported Codecs
The supported codecs are available for use with Cisco H.323 Version 2 software. Table 18 lists each
codec with a default packet size (in bytes) and a range.
Table 18
Note
Codec Default Packet Size
Codecs
Range (in
bytes)
Default (in
bytes)
Bit Rate
G.711ulaw
40–240
160
64 kbps
G.711alaw
40–240
160
64 kbps
G.723r63
24–240
24
6.3 kbps
G.723r53
20–240
20
5.3 kbps
G.723ar63
24–240
24
6.3 kbps
G.723ar53
20–240
20
5.3 kbps
G.726r32
20–240
40
32 kbps
G.726r24
15–240
30
24 kbps
G.726r16
10–240
20
16 kbps
G.728
10–240
10
16 kbps
G.729br8
10–240
20
8 kbps
G.729r8 pre-ietf
10–240
20
8 kbps
G.729r8
10–240
20
8 kbps
A separate codec for G.729 Annex B is included, which adds Annex B functionality to G.729. A
separate codec for G.723.1 Annex A adds Annex A functionality to G.723.1.
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Note
The Annex B functionality added to G.729 and the Annex A functionality added to G.723.1 are the
built-in, codec-specific voice-activated detection/calling tone (VAD/CNG) functions.
H.245 Empty Capabilities Set
Empty capabilities set support is a mandatory part of the H.323 Version 2 standard. It is used by
applications to redirect the voice media stream. This feature is particularly useful for applications such
as the following:
•
Selsius IP phones, which rely on a hub or call manager to direct the media stream to IP phones.
•
Unified messaging for which it is desirable to redirect the media stream to various message servers
for message playout.
The empty capabilities set feature was added to provide a way to redirect RTP streams. The RTP streams
are redirected as follows:
•
The sequence starts with the an empty capabilities set being received at an endpoint.
•
After an open logical channel (OLC) is established (or if in the middle of this process) one of the
endpoints sends an empty capabilities set message.
•
When the empty capabilities set message is received, the other endpoints close the logical channel
if any was opened with that endpoint and move to a pause state, waiting for a nonempty capability
set message.
After receiving the nonempty capabilities set message, the endpoint moves to the beginning of Phase B,
which is the initial communication and capabilities exchange, as described in H.323 Version 3 (June
1999), item 8.4.6.
In other words, the exchange of the capabilities message determines a master/slave relationship, and a
new OLC message is created to open a new logical channel with another endpoint. From this point on,
the RTP streams are sent to the new endpoint.
H.323 Version 2 Fast Connect
Fast connect allows endpoints to establish media channels without waiting for a separate H.245
connection to be opened. This streamlines the number of messages that are exchanged and the amount
of processing that must be done before endpoint connections can be established. A high-level view of
the fast connect procedures within the H.323 protocol follows:
1.
The calling endpoint transmits a setup message containing the fastStart element that contains a
sequence of encoded logical channel structures, each representing a different capability media type
for both “send” and “receive” directions.
2.
The called endpoint selects one or more of the media types offered by the calling endpoint for the
send and receive directions and returns its selections in a fastStart element in any H.225.0 message
up to and including connect. At this point, the called endpoint must be prepared to receive media
along any of the channels it selected.
3.
If H.245 procedures are needed and one or both of the endpoints do not support tunneling, a separate
H.245 connection is used.
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Fast Connect is not explicitly configurable. All H.323 Version 2 VoIP endpoints are capable of initiating
or accepting fast connect calls. It is assumed that the gateway is capable of sending and receiving fast
connect procedures unless its corresponding dial peer has been configured for the Resource Reservation
Protocol (RSVP). (In other words, the req-qos is set to a value other than the default of best-effort.) If
the dial peer has been configured for RSVP, traditional “slow” connect procedures are followed, and the
endpoint neither attempts to initiate fast connect nor responds to a fast connect request from its peer.
A terminating endpoint can reject fast connect by simply omitting the fastStart element from all H.225.0
messages up to and including connect. In this case, normal H.245 procedures are followed and a separate
H.245 TCP connection is established. So, if an endpoint does not support the fast connect procedures,
normal H.245 procedures are followed. In addition, certain conditions can cause a fast connect call to
fall back to normal H.245 procedures to complete the call.
Once a media connection has been opened (an audio path has been established), either endpoint has the
option of switching to H.245 procedures (if they are needed) by using H.245 tunneling, whereby H.245
messages are encapsulated within the h245Control element of H.225.0 messages.
The dtmf-relay command is the only H.245 cognizant command that can initiate H.245 tunneling
procedures from a fast connect call. If H.245 tunneling is active on the call, switching to a separate H.245
connection is not supported.
A Cisco terminating endpoint accepts a fast connect request only if a pair of symmetric codecs (codecs
that in both directions are equivalent or identical) can be selected from a list that has been offered. The
originating endpoint is constrained only by what it can send through the codec (or voice class codec list)
associated with the dial peer.
If the Cisco originating endpoint has offered multiple codecs and the terminating endpoint selects a pair
of asymmetric (mismatched) codecs, the originating endpoint initiates separate H.245 procedures to
correct the asymmetric codec situation.
Fast connect is backward compatible with H.323 Version 1 configurations.
Note
Because fast connect is compliant with H.323 Version 2 and because the majority of endpoints prefer
to establish a call by using fast connect procedures, this feature is not configurable. The H.323 fast
connect feature does not require any additional configuration beyond a working voice configuration.
H.450.2 Call Transfer
Call transfer allows an H.323 endpoint to redirect an answered call to another H.323 endpoint. Cisco
gateways support H.450.2 call transfer as the transferring and transferred-to party. The transferring
endpoint must be an H.450-capable terminal; the Cisco gateway cannot act as the transferring endpoint.
Gatekeeper-controlled or gatekeeper-initiated call transfer is not supported.
Note
Certain devices are limited in their support of H.450. The Cisco 1700 and uBR820 platforms do not
support IVR. Therefore, these platforms are not able to act as H.450 transferring endpoints.
H.450.2 specifies two variants of call transfer:
•
Transfer without consultation—The transferring endpoint supplies the number of the transferred-to
endpoint as part of the transfer request, and the two remote endpoints are transferred together. A
Cisco gateway cannot be the transferring endpoint.
•
Transfer with consultation—This feature is not currently supported.
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H.450.3 Call Deflection
Call deflection is a feature under H.450.3 Call Diversion (Call Forwarding) that allows a called H.323
endpoint to redirect the unanswered call to another H.323 endpoint. Cisco gateways support H.450.3 call
deflection as the originating, deflecting, and deflected-to gateway. The Cisco gateway as the deflecting
gateway supports invocation of call deflection only by using an incoming PRI QSIG message (call
deflection cannot be invoked by using any other trunk type).
If the deflecting endpoint is a Cisco gateway, the telephony endpoint on the PRI of the deflecting
gateway invokes call deflection by sending an equivalent QSIG reroute invoke request within a
FACILITY message to the gateway. The deflecting gateway then uses the procedures outlined in the
H.450.3 call deflection standard to transfer the call to another endpoint. Note that the initiation of
deflection using QSIG reroute invoke is valid only on calls that arrived as H.323 calls at the deflecting
gateway. In other words, for calls that arrive at the gateway through a telephony interface (such as a
hairpin call) or by using a non-H.323 IP protocol, QSIG reroute invoke is ignored.
Cisco H.323 Version 2 software does not support gatekeeper-controlled or gatekeeper-initiated call
deflection.
Note
Certain devices are limited in their support of the H.450 standard. The Cisco AS5800 universal
access server is not able to convert QSIG to H.450. The Cisco 1700 and uBR820 platforms do not
support IVR. Therefore, these devices are not able to act as H.450 deflecting endpoints.
Gateway Support for Alternate Endpoints
Alternate endpoints allow a gatekeeper to specify alternative destinations for a call when queried with
an ARQ by an originating gateway. If the first destination gateway fails to connect, the gateway tries all
the alternate destinations before going to the next dial peer rotary (if a rotary is configured).
Note
This feature is not supported by the Cisco gatekeeper; it is intended for use with third-party
gatekeepers that implement the alternate endpoint field in the ACF message. No support is provided
for the gateway to send a list of alternate endpoints in RRQ messages.
Gatekeeper C Code Generic API for GKTMP in a UNIX Environment
This API allows third-party applications that run in a UNIX host to send GKTMP messages to a
Cisco gatekeeper and receive GKTMP messages from a Cisco gatekeeper. This API may be used to
develop back-end services such as authentication, billing, and address translation.
Gateway Support for a Network-Based Billing Number
Gateway support for a network-based billing number informs the gatekeeper of the specific voice port
or T1/E1 span from which an incoming call entered the ingress gateway. This is done using a Cisco
proprietary, nonstandard field that has been added to the ARQ message sent by the ingress gateway. No
configuration is necessary for this feature.
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Gateway Support for Voice-Port Description
Gateway support for voice-port description provides the gatekeeper with a configurable string that
identifies the voice port or T1/E1 span from which an incoming call entered the ingress gateway. This is
done using a Cisco proprietary, nonstandard field that has been added to the ARQ message sent by the
ingress gateway. The string in the ARQ message corresponds to the setting of the voice-port description
command.
Gateway support for voice-port description is similar to the network-based billing number feature, but it
differs in two important respects:
•
The voice-port description field is only included in the ARQ message if the voice-port description
is configured through the CLI for the applicable voice port.
•
Because the voice-port description is configurable, the user can provide customer-specific
information to the gatekeeper. For example, the voice-port description can be configured to
correspond to the carrier identification code (CIC) for calls received on a particular T1/E1 span.
H.323 Signaling
When interworking with ISDN, with T-1 channel-associated signaling (CAS), and with E-1 R2 services
from the PSTN, H.323 signaling enables VoIP networks to properly signal the setup and teardown of
calls. In-band tones and announcements are generated as needed at the originating or terminating switch.
When a tone is played at the destination switch, the backward voice path from the called party to the
calling party is cut through early so that the calling party can hear the tone or announcement. To prevent
fraudulent calls, the voice path is cut through in both directions only after the connect message is
received from the destination. The call progress indicator, which signals the availability of in-band
communication, is carried end to end as required when interworking with ISDN and CAS protocols.
The H.323 signaling feature prevents unexpected behavior, such as early alerting (when an alert message
is returned immediately after a call proceeding message is sent), to ensure that the calling party does not
hear conflicting call progress information, such as a ringback tone followed by a busy tone, and does not
miss hearing a tone or announcement when one should play. Support for network-side ISDN and
reduction in the risk of speech clipping is also addressed.
The H.323 signaling feature is dependent on Cisco H.323 gateways, gatekeepers, and VoIP features.
H.323 signaling provides the following:
•
In-Band Tones and Announcements, page 229
•
End-to-End Alerting, page 231
•
Cut-Through of Voice Path, page 231
•
H.245 Initiation, page 231
•
Overlap Dialing, page 232
In-Band Tones and Announcements
In-band progress tones and announcements are required for PSTN services and for ISDN speech and 3.1
kHz voice services, per Bellcore and American National Standards Institute (ANSI) specifications. To
guarantee that in-band tones and announcements are generated when required and at the appropriate
switch, Cisco H.323 signaling software ensures that the progress indicator (PI) is carried end to end in
call-signaling messages between the called party and the calling party. The PI in outbound dial peers can
also be configured at the H.323 VoIP gateway, if necessary.
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The PI is an IE that signals when in-band tones and announcements are available. The PI controls
whether the local switch generates the appropriate tone or announcement or whether the remote switch
is responsible for the generation. For example, if the terminating switch generates the ringback tone, it
sends a PI of 1 or 8 in the alerting message. If the originating switch receives an alerting message without
a PI, it generates the ringback tone.
The specific PI that a switch sends in call messages, if any, depends on the model of the switch. To ensure
that in-band communication is generated appropriately, it may be necessary in some instances to
override the default behavior of the switch by manually configuring the PI at the Cisco H.323 gateway.
The PI is configurable in setup messages from the outbound VoIP dial peer, typically at the originating
gateway, and in alert, progress, and connect messages from the outbound POTS dial peer, typically at the
terminating gateway. The PI is configured by using the progress_ind dial-peer configuration command.
Table 19 shows the PI values that may be configured on the H.323 gateway.
Table 19
Configurable Progress Indicator Values for H.323 Gateways
PI
Description
Message Type
0
No progress indicator is included.
Setup
1
Call is not end-to-end ISDN; further call
Alert, setup, progress, connect
progress information may be available in-band.
2
Destination address is non-ISDN.
Alert, progress, connect
3
Origination address is non-ISDN.
Setup
8
In-band information or appropriate pattern is
now available.
Alert, progress, connect
When the interworking is between ISDN and non-ISDN networks, the originating gateway reacts as
follows:
•
If the originating switch does not include a PI in setup messages, the originating gateway assumes
that the originating switch is ISDN and expects the switch to generate the ringback tone. Determine
which device generates the ringback tone by using the progress_ind dial-peer configuration
command:
– To enable the terminating switch to generate the ringback tone, set the PI to 8 in the alert
messages on the terminating gateway. The progress indicator is configured in the POTS dial
peer.
– To enable the originating gateway to generate the ringback tone, set the PI to 3 in setup
messages on the originating gateway. The PI is configured in the VoIP dial peer.
Note
•
If the terminating gateway sends an alert message with no PI value, the originating
gateway generates the ringback tone. But if the terminating gateway sends an alert
message that has a PI of 1, 2, or 8, the originating gateway does not generate the ringback
tone.
The originating gateway cuts through the voice path in the backward direction when it receives a
progress or alert message that has a PI of 1, 2, or 8.
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Note
Pure ISDN calls may use different protocols at the originating and terminating ends. For example, a
call may originate on ETSI and terminate on NI2. If the two protocols are not compatible end to end,
the gateway drops all IEs from messages, including the progress indicator. Because a progress
indicator is required in all progress messages, the originating gateway inserts a PI of 1 in the progress
message. To avoid dropping IEs, use the isdn gateway-max-internetworking global configuration
command to prevent the gateway from checking protocol compatibility.
End-to-End Alerting
Early alerting is prevented in these ways:
•
For calls that terminate at an ISDN switch—The terminating gateway sends an alert message to the
originating gateway only after it receives an alert message from the terminating switch.
•
For calls that terminate at a CAS switch—The terminating gateway sends a progress message, rather
than an alert message, to the originating gateway after it receives a setup message.
Cut-Through of Voice Path
When tones and announcements are generated at the destination switch, the backward voice path from
the called party to the calling party is cut through before the tones and announcements are played. This
allows announcements, such as “The number you have called has been changed,” or allows tones for
error conditions, such as network congestion, to be forwarded to the calling party. To prevent fraudulent
calls, the originating gateway does not perform full cut-through until it receives a connect message from
the destination switch. Cut-through is performed as follows:
•
For calls that terminate at an ISDN switch—The terminating gateway performs backward
cut-through when it receives an alert or progress message and full cut-through (both directions)
when it receives a connect message. The originating gateway performs backward cut-through when
it receives a call proceeding message and full cut-through when it receives a connect message.
•
For calls that terminate at a CAS switch—The terminating gateway performs backward cut-through
after it sends a progress message and full cut-through (both directions) when it receives an off-hook
signal. The originating gateway performs backward cut-through when it receives a progress message
and full cut-through when it receives a connect message.
Note
If the originating or terminating gateway sends a call proceeding message and then receives
a call proceeding message with a progress indicator of 1, 2, or 8, the gateway converts this
call proceeding message into a progress message with a corresponding PI.
H.245 Initiation
To avoid speech clipping, H.245 capabilities are now initiated at the originating gateway at the earliest
possible moment, when the originating gateway receives a call proceeding message from the terminating
gateway. Previously, call proceeding messages were not passed end to end across the VoIP network;
H.245 was initiated only after the originating gateway received an alert message.
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Overlap Dialing
To enhance overlap dialing, the call proceeding message is now passed transparently from the
terminating switch to the originating switch if the originating switch does not include the
sending complete information element in the setup message. The call proceeding message notifies the
originating switch that the terminating switch has collected all dialed digits that are required to route the
call. If the originating switch sends a sending complete IE, the originating gateway responds with a
call proceeding message, and the session application drops the call proceeding message sent by the
terminating switch.
Configurable Timers in H.225.0
When a call is attempted, a TCP connection is made: the TCP socket connection is made for the signaling
that the H.225.0 protocol carries. When the timer expires, the call is timed out and attempted using
another dial peer, if one has been defined. Cisco configurable timers in H.225.0 software allow users to
configure the H.225.0 TCP connection timeout value for all outgoing call attempts (on a per-VoIP dial
peer basis).
In previous releases of Cisco IOS software, the call attempt timeout was 15 seconds and could not be
changed. In some cases, however, users might need a shorter timeout value to facilitate a faster failover.
In other cases, they might need a greater timeout value.
Configurable timers in H.225 address those needs by allowing the user to override the default of 15
seconds and configure the timeout value.
See the “H.323 Configuration Task List” section for information on how to configure timers in H.225.0.
Answer Supervision Reporting
Answer supervision reporting is an enhancement to the information request (IRR) Registration,
Admission, and Status (RAS) protocol message that enables gatekeepers to maintain call accounting
information by reporting the call connection time of connected calls to the gatekeeper.
In H.323 configurations, the endpoint (gateway) uses direct call-routed signaling. Gatekeepers do not
have real-time knowledge or control over the state of a call and are dependent on the endpoints to provide
them with necessary real-time information, such as call connect time, call termination time, and call
termination reason.
When a call ends, the gateway sends a DRQ message with the BillingInformationToken (which contains
the duration of the call) to the gatekeeper. If for some reason the gatekeeper does not receive the DRQ
message, the gatekeeper will not have the information about when the call started or the duration of the
call, which is necessary to maintain accounting information.
Answer supervision reporting allows the call connection time to be reported to the gatekeeper upon the
connection of a call and at periodic intervals thereafter. Answer supervision reporting adds a proprietary
Cisco parameter, the call connection time, to the perCallInfo parameter in the nonStandardData field,
which is located in the IRR message. When a connect message is received, the originating gateway sends
the unsolicited IRR message to its gatekeeper. On sending a connect message, the terminating gateway
sends the unsolicited IRR message to its gatekeeper. If the ACF message has a nonzero value for the IRR
frequency parameter, the gateway sends the unsolicited IRR message to its gatekeeper at periodic
intervals, which are determined by the value in the IRRfrequency parameter.
With the exception of containing the call connection time in the perCallInfo parameter, the IRR message
and its functionality remain the same.
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Gateway-to-Gatekeeper Billing Redundancy
Gateway-to-gatekeeper billing enhances the accounting capabilities of the Cisco H.323 gateway and
provides support for VocalTec™ gatekeepers. Gateway-to-gatekeeper billing redundancy provides for
redundant billing information to be sent to an alternate gatekeeper if the primary gatekeeper to which a
gateway is registered becomes unavailable.
During the process of establishing a call, the primary gatekeeper sends an ACF message to the registered
gateway. The ACF message includes the billing information of the user and an access token. To provide
the billing information to an alternate gatekeeper if the primary gatekeeper is unavailable when the call
session ends, the access token information sent in the ACF message is also included in the DRQ message
that is sent to the alternate gatekeeper.
This features enables the alternate gatekeeper to obtain the billing information required to successfully
complete the transaction.
For further information on configuring gateway-to-gatekeeper billing redundancy, refer to Cisco H.235
Accounting and Security Enhancements for Cisco Gateways, Cisco H.323 Gateway Security and
Accounting Enhancements, and Gateway Support for Alternate Gatekeeper.
Ecosystem Gatekeeper Interoperability
Ecosystem gatekeeper interoperability adds support for the alternate gatekeeper field (altGKInfo) in the
gatekeeper rejection (GRJ), registration rejection (RRJ), and admission rejection (ARJ) messages. This
allows a gateway to move between gatekeepers during the GRQ, RRQ, and ARQ phases. There is no
need for gateway reconfiguration or for a gatekeeper failover in the gateway.
Gateways can be configured to switch from their primary gatekeeper to an alternate gatekeeper if a
failure or outage occurs. If an outage occurs and gateways move from one gatekeeper to another, there
may be an imbalance in the number of gateways registered to each gatekeeper. The ecosystem gatekeeper
interoperability feature helps to restore the balance (when the outage has been corrected) by allowing
some of the gateways to be moved back to their proper gatekeepers.
The altGKInfo consists of two subfields: the alternateGatekeeper and the altGKisPermanent flag. The
alternateGatekeeper is the list of alternate gatekeepers. The altGKisPermanent is a flag that indicates
whether the gatekeepers in the associated alternateGatekeeper field are permanent or temporary.
•
If the current state of the altGKisPermanent flag is TRUE, the new altGKInfo of any RAS message
received from one of the alternate gatekeepers is accepted and the new list will replace the existing
list.
•
If the current state of the altGKisPermanent flag is FALSE, the altGKInfo of any RAS message
received from one of the alternate gatekeepers will be ignored.
If the current permanent gatekeeper becomes nonresponsive and the altGKisPermanent flag is set to
FALSE, the gateway sets the internal state of the altGKisPermanent flag to TRUE. This allows the
gateway to accept the alternate gatekeeper list from one of the gatekeepers in the existing alternate
gatekeeper list.
The handling of the altGKInfo field varies depending on whether it is included in a GRJ or an RRJ
message.
For further information on configuring ecosystem gatekeeper interoperability, refer to Gateway Support
for Alternate Gatekeepers, Configuring H.323 VoIP Gateway for Cisco Access Platforms, and Ecosystem
Gatekeeper Interoperability Enhancements.
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AltGKInfo in GRJ Messages
When the gateway accepts the alternate gatekeeper list from the GRJ, the gateway sends a GRQ message
to a gatekeeper on the list. The selection is based on priority of the alternate gatekeepers. Each alternate
gatekeeper is tried until a GCF message is received.
If the gateway receives a GRJ message without the AltGKInfo field, it accepts the rejection. Because this
is the first phase for the gateway to contact a gatekeeper, the gateway is considered lost without a
gatekeeper.
During the GRQ phase, the gateway ignores the value of the altGKisPermanent flag in any RAS message
and sets the value internally to TRUE.
AltGKInfo in RRJ Messages
When the gateway accepts the alternate gatekeeper list from the first RRJ message, the gateway
retransmits an RRQ message to a gatekeeper on the alternate gatekeeper list. The selection is based on
priority of the alternate gatekeepers.
The retransmission of the RRQ message depends on the type of RRQ (full or lightweight), the current
state of the altGKisPermanent flag, and the current state of the needToRegister flag of each alternate
gatekeeper as follows:
•
If the state of the altGKisPermanent flag is TRUE and the state of the needToRegister flag is NO,
the gateway will retransmit the full RRQ to an alternate gatekeeper for full RRQs and a lightweight
RRQ for lightweight RRQs.
•
If the state of the altGKisPermanent flag is TRUE and the state of the needToRegister flag is YES,
the gateway will retransmit the full RRQ to an alternate gatekeeper for full RRQs and lightweight
RRQs.
•
If the state of the altGKisPermanent flag is FALSE and the state of the needToRegister flag is NO,
the gateway will retransmit a lightweight RRQ for lightweight RRQs and nothing for full RRQs.
•
If the state of the altGKisPermanent flag is TRUE and the state of the needToRegister flag is YES,
the gateway will not retransmit the RRQ.
If the gateway receives an RRJ message without the AltGKInfo field, it accepts the rejection and returns
to the GRQ phase. If the state of the altGKisPermanent flag is FALSE, the gateway sends the GRQ
message to the original gatekeeper that sent the first RRJ. If the state of the altGKisPermanent flag is
TRUE, the gateway sends the GRQ to the current gatekeeper.
If the current state of the altGKisPermanent flag is TRUE, then the next RAS message is sent to the new
gatekeeper. Otherwise, the next RAS message is sent to the original gatekeeper.
If the gateway exhausts the list of alternate gatekeepers without receiving any response from an alternate
gatekeeper, the gateway returns to the GRQ phase.
For more information regarding the Cisco ecosystem gatekeeper interoperability feature, see the
“Alternate Gatekeepers” section in the “Configuring H.323 Gateways and Proxies” chapter.
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H.323 Restrictions
H.323 Restrictions
The following sections contain the restrictions that apply to the Cisco H.323-compliant features:
•
H.323 Version 2 Feature Restrictions, page 235
•
H.323 Signaling Enhancement Feature Restrictions, page 235
•
Configurable Timers in H.225.0 Restriction, page 236
•
Source Call Signal Address and H.245 Empty Capabilities Set Restrictions, page 236
•
Ecosystem Gatekeeper Interoperability Restrictions, page 236
H.323 Version 2 Feature Restrictions
The following restrictions apply to the Cisco H.323 Version 2 features:
•
All systems must be running either Cisco IOS Release 11.3(9)NA and later releases or Cisco IOS
Release 12.0(3)T and later releases to interoperate with the Cisco H.323 Version 2 features. Earlier
releases contain H.323 Version 1 software that does not support protocol messages that have an
H.323 Version 2 protocol identifier. The earlier releases will not interoperate with Cisco H.323
Version 2 Phase 2 features.
•
To use H.450 services (call transfer or call deflection), use Cisco IOS Release 12.1(1)T of the
gatekeeper: H.450 on the gateways is incompatible with previous releases of the Cisco gatekeeper.
•
If a Cisco AS5300 universal access server is used, the software requires the appropriate version of
VCWare.
•
The H.323 Version 2 fast connect feature is not explicitly configurable. It is assumed that the
gateway is capable of sending and receiving fast connect procedures unless its corresponding dial
peer has been configured for RSVP (in other words, the req-qos is set to a value other than the
default of best-effort). If the dial peer has been configured for RSVP, traditional “slow” connect
procedures will be followed, and the endpoint will neither attempt to initiate fast connect nor
respond to a fast connect request from its peer.
H.323 Signaling Enhancement Feature Restrictions
The following restrictions apply to the H.323 signaling enhancement feature:
•
Supplementary voice services are not supported with ISDN and CAS over an H.323
network—except on the NET5 switch.
•
Progress messages require a PI value, and only ITU-T standards are supported.
•
Progress indicator 2 is not supported in progress messages for the DMS100 switch.
•
TCL 2.0 for IVR supports the interworking signaling enhancements only on the Cisco AS5300. For
IVR on other Cisco platforms, select TCL 1.0 as the session application. To use standard IVR
applications with TCL 1.0, configure the application name as “session.t.old” by using the call
application voice global configuration command. It is not necessary to do this if customized scripts
are used.
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•
The Cisco AS5300 universal access server sends a connect message to the originating gateway after
it receives a setup message only when it is configured for one of the following supported switch
types:
– 5ESS
– NET5
– NTT
– QSIG
– QSIGP
•
For the SS7 interconnect for voice gateways solution, the following behavior applies to suspend and
resume messages, which are supported on NET5 and NI2+ ISDN interfaces:
– If the ISDN interface is NET5, the Cisco AS5300 sends a notify message with the notification
indicator (NI) set to user-suspended or user-resumed.
– If the ISDN interface is NI2+, the Cisco AS5300 sends a suspend or resume message to the
Cisco SC2200.
– If the Cisco SC2200 receives an ISUP suspend or resume message, it sends an NI2+ suspend or
resume message to the Cisco AS5300.
– Both the Cisco AS5300 and SC2200 timers start when a suspend message is received. The
Cisco AS5300 timer, T307, is configurable from 30 to 300 seconds. The Cisco SC2200 timer,
T6, is not configurable and has a default of 120 seconds if the ISUP variant Q.761 is used.
When the Cisco AS5300 and the SC2200 receive a resume message, the timers are stopped. If
either of the the timers expires, the call is released with a cause code of normal clearing.
Configurable Timers in H.225.0 Restriction
This feature is limited to H.323 dial peers.
Source Call Signal Address and H.245 Empty Capabilities Set Restrictions
The following restrictions apply to source call signal address and H.245 empty capabilities set:
•
To use H.450 services (call transfer or call deflection), Cisco IOS Release 12.1(2)T of the gatekeeper
must be used. H.450 on the gateways is incompatible with previous releases of the Cisco gatekeeper.
•
If a Cisco AS5300 universal access server is used, the system requires the appropriate version of
VCWare.
Ecosystem Gatekeeper Interoperability Restrictions
The following restrictions apply to ecosystem gatekeeper interoperability:
•
The maximum number of alternate gatekeepers is eight (including static gatekeepers).
•
During the retransmission of the GRQ or RRQ messages, the gateway responds only to the current
gatekeeper (regardless of the state of the altGKisPermanent flag).
•
The process of retransmission to an alternate gatekeeper can be time-consuming.
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H.323 Prerequisite Tasks
To use the Cisco H.323 signaling enhancements, first do the following:
•
Establish a working IP network. For more information about configuring IP, refer to the Cisco IOS
IP Configuration Guide.
•
Install the appropriate voice network module and voice interface card for the Cisco router. For more
information about the physical characteristics of the voice network module or on how to install it,
refer to the Voice Network Module and Voice Interface Card Configuration Note that came with the
voice network module.
•
Configure your H.323 gateways, gatekeepers, and proxies. For more information about configuring
VoIP for your access platform, see the “Configuring H.323 Gateways,” “Configuring H.323
Gatekeepers and Proxies,” and “Voice over IP Overview” chapters in this configuration guide.
•
To ensure network security, configure a RADIUS authentication, authorization, and accounting
(AAA) server.
In addition to the configuration, make sure that the following information is configured in your
CiscoSecure AAA server:
In the /etc/raddb/clients file, ensure that the following information is provided:
#Client Name
#----------gk215.cisco.com
Key
------------------testing123
Where:
gk215.cisco.com is resolved to the IP address of the gatekeeper requesting authentication
In the /etc/raddb/users file, ensure that the following information is provided:
[email protected] Password = "thiswouldbethepassword"
User-Service-Type = Framed-User,
Login-Service = Telnet
Where:
[email protected] is the h323-id of the gateway authenticating to gatekeeper gk215.cisco.com.
•
Configure an NTP server for your network.
Additional requirements and tasks for the individual features follow:
•
The configurable timers in the H.225.0 feature require the Cisco H.323 VoIP Gateway for Cisco
Access Platforms feature.
•
Answer supervision reporting requires a Cisco H.323 gatekeeper.
•
Gateway-to-gatekeeper billing redundancy requires a Cisco H.323 gatekeeper and the Gateway
Support for Alternate Gatekeepers feature.
•
Ecosystem gatekeeper interoperability requires a Cisco H.323 gatekeeper.
•
For H.323 Version 2 features, configure an NTP server for the network.
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H.323 Configuration Task List
To configure the H.323 features in this chapter, perform the tasks described in the following sections:
•
Configuring Timers in H.225.0, page 238
•
Configuring H.245 Tunneling of DTMF Relay in Conjunction with Fast Connect, page 239
•
Configuring H.450, page 239
Configuring Timers in H.225.0
To use the configurable timers in H.225.0, first create an H.323 voice class and then specify the timeout
value associated with that class. To configure the H.225.0 TCP timeout value, use the following
commands beginning in global configuration mode:
Command
Purpose
Step 4
Router(config)# voice class h323 number
Enters voice class mode to create or modify an
H.323 voice class. The number argument identifies
the H.323 voice class. There is no default value.
Step 5
Router(voice-class)# h225 timeout tcp establish value
Sets the H.225.0 TCP timeout value for the
specified voice class. The value argument indicates
the timeout value, in seconds. There is no default
value.
Step 6
Router(voice-class)# exit
Exits voice class mode.
Next, associate the H.323 voice class with each VoIP dial peer that should use the specified timeout. To
associate the H.323 voice class with a dial peer, use the following commands beginning in global
configuration mode:
Step 1
Command
Purpose
Router(config)# dial-peer voice tag voip
Enters dial-peer configuration mode and defines a
remote VoIP dial peer.
The keywords and arguments are as follows:
Step 2
Router(config-dial-peer)# voice-class h323 number
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•
The tag argument is one or more digits
identifying the dial peer.Valid entries are from
1 to 2,147,483,647.
•
The voip keyword indicates a VoIP peer using
voice encapsulation on the IP network.
Associates the specified H.323 voice class (and all
of its related attributes) with the dial peer. The
number argument identifies the H.323 voice class.
There are no default values.
H.323 Applications
H.323 Configuration Task List
Verifying the H.225.0 TCP Timeout Value
To verify that the timeout value is defined for a dial peer, enter the show running-config command. The
output shows the current configuration of the voice class and the dial peer.
Router# show running-config
Building configuration...
Current configuration:
!
voice class h323 1
h225 timeout tcp etablish 10
dial-peer voice 919 voip
application session
destination-pattern 919555....
voice-class codec 1
voice-class h323 1
session target ras
Configuring H.245 Tunneling of DTMF Relay in Conjunction with Fast Connect
The dtmf-relay command configured on the outgoing VoIP dial peer initiates H.245 tunneling
procedures from a fast connect call. Note that H.245 tunneling will be activated only if the dtmf-relay,
h245-alphanumeric, or h245-signal (but not cisco-rtp) commands are configured on the VoIP dial peer.
Configuring H.450
A Cisco gateway for H.450 is configured in one of the following ways, depending on what the gateway
needs to do:
•
By redirecting an unanswered call (call deflection).
•
By transferring an answered call to a new DN (call transfer without consultation).
Although there are no new CLI commands for configuring H.450 services, the services are enabled only
when a TCL/IVR Session Application is configured. Therefore, to use H.450 services, you must
configure a TCL/IVR-based “application” on each applicable incoming dial peer for each Cisco gateway
that will be involved in call transfer or call deflection. If no special TCL/IVR behavior is required, you
can use the standard TCL/IVR application “session.” This is not to be confused with application
“SESSION,” which is not TCL/IVR-based and does not support H.450 services.
In addition, if call deflection is to be initiated from a QSIG PRI, you must configure the PRI using the
isdn switch-type primary-qsig and isdn alert end-to-end commands.
Note
For general information on configuring dial peer application and the meaning of incoming dial peer,
refer to Voice over IP for the Cisco AS5300.
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Configuring Call Deflection
A sample call deflection configuration is shown in Figure 53.
Figure 53
H.450 Configuration to Redirect Unanswered Calls
Cisco 3600
Gateway A
Party A
888-0000
Cisco 5300
Gateway B
Party B
IP
QSIG
PRI 555-3017
FXS
Party C
999-0000
30765
PRI
Cisco 5300
Gateway C
In this example, three gateways are configured to redirect unanswered calls, so that when Party A calls
Party B, Party B can invoke deflection to pass the call to Party C. For this to work, “application session”
or another TCL/IVR-based application must be configured on each applicable incoming dial peer as
follows:
•
On Gateway A, the POTS dial peer for destination pattern 8880000.
•
On Gateway B, the VoIP dial peer for destination pattern 8880000.
•
On Gateway C, the VoIP dial peer for destination pattern 8880000.
To configure the Gateway A POTS dial peer, use the following commands beginning in global
configuration mode:
Step 1
Command
Purpose
Router(config)# dial-peer voice tag pots
Enters dial-peer configuration mode.
The tag argument is a digit that defines a particular dial peer.
Valid entries are from 1 to 2,147,483,647.
Step 2
Router(dial-peer)# application name
Specifies the application that will be invoked for this dial peer.
Only TCL-based applications are able to support H.450
services.
The name argument indicates the name of the predefined
TCL/IVR application. Incoming calls using this POTS dial
peer will be handed off to this application.
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Step 3
Command
Purpose
Router(dial-peer)# destination-pattern
[+]string[T]
Specifies either the prefix or the full E.164 telephone number
(depending on the dial plan) to be used for a dial peer.
The keywords and arguments are as follows:
•
[+]—(Optional) Specifies a character indicating an E.164
standard number. The plus sign (+) is not supported on the
Cisco MC3810 multiservice concentrator.
•
string—Specifies a series of digits that specify the E.164
or private dialing plan telephone number. Valid entries are
the digits 0 through 9, the letters A through D, and the
following special characters:
– The asterisk (*) and pound sign (#) that appear on
standard touch-tone dial pads.
– Comma (,)—Inserts a pause between digits.
– Period (.)—Matches any entered digit (this character
is used as a wildcard).
– Percent sign (%)—Indicates that the previous
digit/pattern occurred zero or multiple times, similar
to the wildcard usage in the regular expression.
– Plus sign (+)—Matches a sequence of one or more
matches of the character/pattern.
Note
The plus sign used as part of the digit string is different
from the plus sign that can be used in front of the digit
string to indicate that the string is an E.164 standard
number.
– Circumflex (^)—Indicates a match to the beginning
of the string.
– Dollar sign ($)—Matches the null string at the end of
the input string.
– Backslash symbol (\)—Is followed by a single
character matching that character or used with a
single character having no other significance
(matching that character).
– Question mark (?)— Indicates that the previous digit
occurred zero or one time.
– Brackets ([])—Indicates a range of digits. A range is
a sequence of characters enclosed in the brackets, and
only numeric characters from “0” to “9” are allowed
in the range. This is similar to a regular expression
rule.
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Command
Purpose
– Parentheses (())—Indicate a pattern and is the same
as the regular expression rule—for example,
408(555). Parentheses are used in conjunction with
symbols ?, %, or +.
For more information on applying wildcard symbols
to destination patterns and the dial strings that result,
see the “Configuring Dial Plans, Dial Peers, and Digit
Manipulation” chapter.
•
T—(Optional) Control character indicating that the
destination-pattern value is a variable-length dial string.
Step 4
Router(dial-peer)# exit
Exits dial-peer configuration mode.
Step 5
2600 and 3600 Series Routers
Specifies the voice slot number, subunit number, and port
through which incoming VoIP calls will be received.
Router(config)# port
{slot-number/subunit-number/port} |
{slot/port:ds0-group-no}
The keywords and arguments are as follows:
•
slot-number—Specifies the slot number in the Cisco
router in which the voice interface card is installed. Valid
entries are from 0 to 3, depending on the slot in which it
has been installed.
•
subunit-number—Specifies the subunit on the voice
interface card in which the voice port is located. Valid
entries are 0 or 1.
•
port—Specifies the voice interface card location. Valid
entries are 0 or 3.
•
slot—Specifies the router location in which the voice port
adapter is installed. Valid entries are from 0 to 3.
•
port—Specifies the voice interface card location. Valid
entries are 0 or 3.
•
ds0-group-no—Indicates the defined DS0 group number.
Each defined DS0 group number is represented on a
separate voice port. This allows individual DS0s to be
defined on the digital T1/E1 card.
Note
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The slashes must be entered along with the arguments
shown within the braces in the Command column.
H.323 Applications
H.323 Configuration Task List
To configure the VoIP dial peers on Gateways B and C, use the following commands beginning in global
configuration mode:
Step 1
Command
Purpose
Router(config)# dial-peer voice number voip
Enters dial-peer configuration mode.
The number argument defines a particular dial peer. Valid
entries are 1 to 2,147,483,647.
Step 2
Router(dial-peer)# application name
Specifies the application that will be invoked for this dial peer.
Only Tool Command Language- (TCL-) based applications
are able to support H.450 services.
The name argument indicates the name of the predefined
TCL/IVR application. Incoming calls using this VoIP dial peer
will be handed off to this application.
Step 3
Router(dial-peer)# destination-pattern
[+]string[T]
Specifies either the prefix or the full E.164 telephone number
(depending on the dial plan) to be used for a dial peer.
For a description of the keywords and arguments for this
command, see Step 3 in the first configuration task table
(showing how to configure the Gateway A POTS dial peer) in
the “Configuring Call Deflection” section on page 240.
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Step 4
Command
Purpose
2600 and 3600 Series VoIP Dial Peers
Specifies the network-specific address for a specified dial
peer.
Router(dial-peer)#session target
{ipv4:destination-address | dns:[$s$. | $d$.
| $e$. | $u$.]host-name | loopback:rtp |
loopback:compressed | loopback:uncompressed}
The keywords and arguments are as follows:
•
ipv4:destination-address—Specifies the IP address of the
dial peer.
•
dns:host-name—Indicates that the Domain Name System
(DNS) will be used to resolve the name of the IP address.
Valid entries for this parameter are characters that
represent the name of the host device.
One of the following four optional wildcards can be used
with this keyword when defining the session target for
VoIP peers:
– $s$—Indicates that the source destination pattern
will be used as part of the domain name.
– $d$—Indicates that the destination number will be
used as part of the domain name.
– $e$—Indicates that the digits in the called number
will be reversed, that periods will be added in
between each digit of the called number, and that this
string will be used as part of the domain name.
– $u$—Indicates that the unmatched portion of the
destination pattern (such as a defined extension
number) will be used as part of the domain name.
•
loopback:rtp—Indicates that all voice data will be looped
back to the originating source. This is applicable for VoIP
peers.
•
loopback:compressed—Indicates that all voice data will
be looped back in compressed mode to the originating
source. This is applicable for POTS peers.
– loopback:uncompressed—Indicates that all voice
data will be looped back in uncompressed mode to the
originating source. This is applicable for POTS peers.
To configure the Gateway B PRI, use the following commands beginning in global configuration mode:
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Step 1
Command
Purpose
Cisco 4000 Series Access Servers
Configures the serial interface.
Router(config)# interface serial
number:timeslot
The keywords and arguments are as follows:
•
number—Channelized E1 or T1 controller number.
•
time-slot—For ISDN, the D channel time slot, which is
the :23 channel for channelized T1 and the :15 channel for
channelized E1. PRI time slots are in the range of 0 to 23
for channelized T1 and in the range of 0 to 30 for
channelized E1.
– For channel-associated signaling or robbed-bit
signaling, time-slot is the channel group number.
– The colon (:) is required.
– On a dual port card, it is possible to run channelized
on one port and PRI on the other port.
Step 2
Router(config-if)# isdn switch-type
switch-type
Configures the ISDN interface as a primary QSIG interface.
The switch-type argument is the service provider switch type.
PRI switch types vary by geographic area. (Refer to the
command reference master index, or search online for this
information.)
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Configuring Call Transfer Without Consultation
A sample configuration is shown in Figure 54.
Figure 54
H.450 Configuration for Calls Transfer Without Consultation
Party B
555-3017
Cisco 3600
Gateway A
Party A
888-0000
IP
FXS
H.450 Capable
Endpoint B
Party C
999-0000
Cisco 5300
Gateway C
30766
PRI
In this example, two gateways are configured to handle call transfers without consultation, so that when
Party A calls Party B at 555-3017 at Endpoint B, Endpoint B answers and then invokes call transfer to
Party C. To do this, configure the application session or another TCL/IVR-based application on each
applicable incoming dial peer as follows:
•
On Gateway A, the POTS dial peer for destination pattern 8880000.
•
On Gateway C, the VoIP dial peer for destination pattern 8880000.
To configure the Gateway A POTS dial peer, use the following commands beginning in global
configuration mode:
Step 1
Command
Purpose
Router(config)# dial-peer voice tag pots
Enters dial-peer configuration mode.
The tag argument is a digit that defines a particular dial peer.
Valid entries are from 1 to 2,147,483,647.
Step 2
Router(dial-peer)# application name
Specifies the application that will be invoked for this dial peer.
Only Tool Command Language- (TCL-) based applications
are able to support H.450 services.
The name argument indicates the name of the predefined
TCL/IVR application. Incoming calls using this POTS dial
peer will be handed off to this application.
Step 3
Router(dial-peer)# destination-pattern
[+]string[T]
Specifies either the prefix or the full E.164 telephone number
to be used for a dial peer (depending on the dial plan) .
For a description of the keywords and arguments for this
command, see Step 3 in the first configuration task table
(showing how to configure the Gateway A POTS dial peer) in
the “Configuring Call Deflection” section on page 240.
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Step 4
Command
Purpose
2600 and 3600 Series POTS Dial Peers
Specifies the IP address of the destination gateway for
“outbound” dial peers. Because this is an “incoming” dial
peer, the session target is not applicable, so the IP address is
ignored.
Router(dial-peer)# session target
Step 5
Router (dial-peer)# port
{slot-number/subunit-number/port} |
{slot/port:ds0-group-no}
Specifies the voice slot number, subunit number, and port
through which incoming VoIP calls will be received.
For a description of the keywords and arguments for this
command, see Step 5 in the first configuration task table
(showing how to configure the Gateway A POTS dial peer) in
the “Configuring Call Deflection” section on page 240.
Step 6
Router(dial-peer)# exit
Exits dial-peer configuration mode.
To configure the Gateway C VoIP dial peer, use the following commands beginning in global
configuration mode:
Step 1
Command
Purpose
Router(config)# dial-peer voice tag voip
Enters dial-peer configuration mode.
The tag argument is a digit that defines a particular dial peer.
Valid entries are from 1 to 2,147,483,647.
Step 2
Router(dial-peer)# application name
Specifies the application that will be invoked for this dial peer.
Only Tool Command Language- (TCL-) based applications
are able to support H.450 services.
The name argument indicates the name of the predefined
TCL/IVR application. Incoming calls using this VoIP dial peer
will be handed off to this application.
Step 3
Router(dial-peer)# destination-pattern
[+]string[T]
Specifies either the prefix or the full E.164 telephone number
to be used for a dial peer (depending on the dial plan) .
For a description of the keywords and arguments for this
command, see Step 3 in the first configuration task table
(showing how to configure the Gateway A POTS dial peer) in
the “Configuring Call Deflection” section on page 240.
Step 4
2600 and 3600 Series VoIP Dial Peers
Router(dial-peer)# session target
{ipv4:destination address | dns: [$s$. | $d$.
| $e$.| $u$.]host-name | loopback:rtp |
loopback:compressed | loopback:uncompressed}
Specifies the network-specific address for a specified dial
peer.
For a description of the keywords and arguments for this
command, see Step 4 in the second configuration task table
(showing how to configure the VoIP dial peers on Gateways B
and C) in the “Configuring Call Deflection” section on
page 240.
For more information about POTS dial peers, refer to the Cisco IOS Release 12.0 Voice, Video, and Home
Applications Configuration Guide or see the “Configuring Dial Plans, Dial Peers, and Digit
Manipulation” chapter in this configuration guide.
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For more information about any of the commands used to configure VoIP dial peers, refer to the
Cisco IOS Voice, Video, and Fax Command Reference; the Cisco IOS Voice, Video, and Home
Applications Command Reference; or see the “Configuring Voice Ports” or the “Configuring Voice over
IP” chapters in this configuration guide.
Configuring Voice-Port Descriptions
The voice-port description feature uses the existing description subcommand for the voice port. When
the voice-port description is being configured, the exact contents of the description field are included in
the ARQ message sent from the ingress gateway.
Note
Configuring the voice-port description has no effect for calls that are not configured to use RAS.
To configure the description on a voice port, use the following commands beginning in global
configuration mode:
Step 1
Command
Purpose
2600 and 3600 Series Routers
Enters voice-port configuration mode for the
specified voice port.
Router(config)# voice-port
{slot-number/subunit-number/port} |
{slot/port:ds0-group-no}
Step 2
Router(config-voiceport)# description string
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The arguments are as follows:
•
slot-number—Specifies the slot number in the
Cisco router in which the voice interface card
(VIC) is installed. Valid entries are from 0 to 3,
depending on the slot in which it has been
installed.
•
subunit-number—Specifies the subunit on the
VIC in which the voice port is located. Valid
entries are 0 or 1.
•
port—Specifies the voice port number. Valid
entries are 0 or 1.
•
slot—Specifies the router location in which the
voice port adapter is installed. Valid entries are
from 0 to 3.
•
port—Indicates the voice interface card location.
Valid entries are 0 or 3.
•
dso-group-no—Indicates the defines DS0 group
number. Each defined DS0 group number is
represented on a separate voice port. This allows
you to define individual DS0s on the digital
T1/E1 card.
Defines the description associated with the voice
port. The string argument is a character string from 1
to 80 characters.
Configuring H.323 Gateways
This chapter describes the configuration of H.323 gateways and contains the following sections:
•
H.323 Gateway Prerequisite Tasks, page 249
•
H.323 Gateway Configuration Task List, page 250
•
H.323 Gateway Configuration Examples, page 279
For a complete description of the gateway commands used in this chapter, refer to the Cisco IOS Voice,
Video, and Fax Command Reference. To locate documentation for other commands that appear in this
chapter, use the command reference master index or search online. For general information about H.323
gateways and their functions, see the “H.323 Applications” chapter in this configuration guide.
For more information on configuring Cisco mobile telephony products, see Appendix F, “Global System
for Mobile Communications Full Rate and Enhanced Full Rate Codecs.”
To identify the hardware platform or software image information associated with a feature in this
chapter, use the Feature Navigator on Cisco.com to search for information about the feature or refer to
the software release notes for a specific release. For more information, see the “Identifying Supported
Platforms”section in the “Using Cisco IOS Software” chapter.
H.323 Gateway Prerequisite Tasks
Before configuring the router as a gateway, perform the following tasks:
•
Establish a working IP network. For more information about configuring IP, refer to the Cisco IOS
IP Configuration Guide.
•
Develop a network plan that details the requirements and characteristics of your Voice over IP
(VoIP) network. For further information, see the “Voice over IP Overview” chapter of this
configuration guide and refer to the Voice over IP Implementation Guide.
•
Ensure that the routers you intend to configure as H.323 gateways are running a Cisco IOS software
image that contains gateway functionality. (Software images that support gateway features contain
-gw- in the code image name.)
To use the H.323 security and accounting features described in this document, keep the following in
mind:
•
These features use the H.235 standard. Because the standard is broad, ensure that the gatekeeper
provides H.235 functionality that specifically complements the gateway implementation described
in this document.
•
In addition, because the H.323 gateway sends the accounting information using a non-standard field
in the ClearToken message, ensure that the gatekeeper is able to handle this information.
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H.323 Gateway Configuration Task List
H.323 Gateway Configuration Task List
An H.323 gateway is an endpoint on a LAN that provides real-time, two-way communication between
H.323 terminals on the LAN and other International Telecommunication Union Telecommunication
Standardization Sector (ITU-T) terminals in the WAN. An H.323 gateway can also communicate with
another H.323 gateway. Gateways allow H.323 terminals to communicate with non-H.323 terminals by
converting protocols. The gateway is the point at which a circuit-switched call is encoded and
repackaged into IP packets. Because gateways function as H.323 endpoints, they provide admission
control, address lookup and translation, and accounting services. In an environment in which both
gatekeepers and gateways are used, only gateways are configured to send VoIP data.
To configure an H.323 gateway, perform the tasks described in the following sections. Except for the
first task, all tasks are optional.
•
Identifying a Router Interface As an H.323 Gateway, page 250
•
Configuring Gateway RAS, page 252
•
Configuring AAA Functionality on the Gateway, page 255
•
Configuring H.235 Gateway Security, page 261
•
Configuring Alternate Gatekeeper Support, page 268
•
Configuring Dual Tone Multifrequency Relay, page 270
•
Configuring FXS Hookflash Relay, page 274
•
Configuring Multiple Codecs, page 276
•
Configuring Rotary Calling Pattern, page 277
•
Configuring H.323 Support for Virtual Interfaces, page 278
Identifying a Router Interface As an H.323 Gateway
To configure a Cisco device as an H.323 gateway in a service provider environment, configure at least
one of its interfaces as a gateway interface. Use either an interface that is connected to the gatekeeper or
a loopback interface for the gateway interface. The interface that is connected to the gatekeeper is
usually a LAN interface—Fast Ethernet, Ethernet, FDDI, or Token Ring.
To configure a gateway interface, use the following commands beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# gateway
Enables the gateway and enters gateway configuration mode.
Step 2
Router(config-gateway)# exit
Exits gateway configuration mode.
Step 3
Router(config)# ip cef
(Optional) Enables Cisco Express Forwarding (CEF) routing.
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Step 4
Command
Purpose
Router(config)# interface type number
[name-tag]
Enters interface configuration mode for the interface that is
connected to the gatekeeper.
The keywords and arguments are as follow:
•
type—Specifies the type of interface to be configured.
•
number—Specifies the port, connector, or interface card
number. The number is assigned at the factory at the time
of installation or when added to a system and can be
displayed with the show interfaces command.
•
name-tag—(Optional) Specifies the logic name to
identify the server configuration so that multiple entries
of server configuration can be entered.
Step 5
Router(config-if)# h323-gateway voip
interface
Identifies this interface as a VoIP gateway interface.
Step 6
Router(config-if)# h323-gateway voip id
gatekeeper-id {ipaddr ip-address
[port-number]| multicast} [priority number]
(Optional) Defines the name and location of the gatekeeper for
this gateway.
The keywords and arguments are as follows:
•
gatekeeper-id—Indicates the H.323 identification of the
gatekeeper. This value must exactly match the gatekeeper
ID in the gatekeeper configuration. The recommended
format is name.domain-name.
•
ipaddr—Indicates that the gateway will use an IP address
to locate the gatekeeper.
•
ip-address—Defines the IP address to be used to identify
the gatekeeper.
•
port-number— (Optional) Defines the port number used.
•
multicast—Indicates that the gateway will use multicast
to locate the gatekeeper.
•
priority number—(Optional) The priority of this
gatekeeper. The range is 1 through 127, and the default
value is 127.
Step 7
Router(config-if)# h323-gateway voip h323-id
interface-id
(Optional) Defines the H.323 name of the gateway, identifying
this gateway to its associated gatekeeper. The interface-id
argument is the H.323 name (ID) used by this gateway when
this gateway communicates with its associated gatekeeper.
Usually, this ID is the name of the gateway, with the
gatekeeper domain name appended to the end:
[email protected]
Step 8
Router(config-if) h323-gateway voip
tech-prefix prefix
(Optional) Defines the technology prefix that the gateway will
register with the gatekeeper. The prefix argument defines the
numbers used as the technology prefixes. Each technology
prefix can contain up to 11 characters. Although not required,
a pound symbol (#) is frequently used as the last digit in a
technology prefix. Valid characters are 0 through 9, the pound
symbol (#), and the asterisk (*).
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Verifying Gateway Interface Configuration
To find the current registration information and status of the gateway, use the show gateway command.
Configuring Gateway RAS
The Registration, Admission, and Status (RAS) signaling function performs registration, admissions,
status, and disengage procedures between the H.323 VoIP gateway and the H.323 VoIP gatekeeper. RAS
tells the gatekeeper to translate the E.164 phone number of the session target into an IP address.
In the RAS exchange between a gateway and a gatekeeper, a technology prefix is used to identify the
specific gateway when the selected zone contains multiple gateways. The tech-prefix dial-peer
configuration command is used to define technology prefixes. See the “Configuring Dial Plans, Dial
Peers, and Digit Manipulation” chapter in this configuration guide for more information on the
tech-prefix command, or refer to the Cisco IOS Voice, Video, and Fax Command Reference.
In most cases there is a dynamic protocol exchange between the gateway and the gatekeeper that enables
the gateway to inform the gatekeeper about technology prefixes and where to forward calls. If, for some
reason, that dynamic registry feature is not in effect, statically configure the gatekeeper to query the
gateway for this information. To configure the gatekeeper to query for this information, see the
“Configuring H.323 Gatekeepers and Proxies” chapter. To configure RAS, define specific parameters for
the applicable plain old telephone service (POTS) and VoIP dial peers. The POTS dial peer informs the
system of which voice port to direct incoming VoIP calls to and (optionally) determines that
RAS-initiated calls will have a technology prefix prepended to the destination telephone number. The
VoIP dial peer determines how to direct calls that originate from a local voice port into the VoIP cloud
to the session target. The session target indicates the address of the remote gateway where the call is
terminated. There are several different ways to define the destination gateway address:
•
By statically configuring the IP address of the gateway.
•
By defining the Domain Name System (DNS) of the gateway.
•
By using RAS. If RAS is being used, the gateway determines the destination target by querying the
RAS gatekeeper.
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To configure RAS, use the following commands beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# dial-peer voice number pots
Enters dial-peer configuration mode to configure a POTS
peer. The number argument is a tag that identifies the dial
peer. (This number has local significance only.) Valid entries
are from 1 to 2,147,483,647.
Step 2
Router(config-dial-peer)# destination-pattern
[+]string[T]
Specifies the E.164 address associated with this dial peer.
The keywords and arguments are as follows:
•
+—(Optional) Specifies a character indicating an E.164
standard number. The plus sign (+) is not supported on
the Cisco MC3810 multiservice concentrator.
•
string—Indicates a series of digits that specify the E.164
or private dialing plan telephone number. Valid entries
are the digits 0 through 9, the letters A through D, and the
following special characters:
– The asterisk (*) and pound sign (#)—Indicates the
keys that appear on standard touch-tone dial pads.
– Comma (,)—Inserts a pause between digits.
– Period (.)—Matches any entered digit (this character
is used as a wildcard).
– Percent sign (%)—Indicates that the previous
digit/pattern occurred zero or multiple times, similar
to the wildcard usage in the regular expression.
– Plus sign (+)—Matches a sequence of one or more
matches of the character/pattern.
Note
The plus sign used as part of the digit string is
different from the plus sign that can be used in front
of the digit string to indicate that the string is an
E.164 standard number.
– Circumflex (^)—Indicates a match to the beginning
of the string.
– Dollar sign ($)—Matches the null string at the end of
the input string.
– Backslash symbol (\)—Is followed by a single
character matching that character or used with a
single character having no other significance
(matching that character).
– Question mark (?)— Indicates that the previous digit
occurred zero or one time.
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Command
Purpose
– Brackets ([])—Indicate a range of digits. A range is
a sequence of characters enclosed in the brackets,
and only numeric characters from “0” to “9” are
allowed in the range. This is similar to a regular
expression rule.
– Parentheses (())—Indicate a pattern and is the same
as the regular expression rule—for example,
408(555). Parentheses are used in conjunction with
symbols ?, %, or +.
For more information on applying wildcard symbols
to destination patterns and the dial strings that result,
see the “Configuring Dial Plans, Dial Peers, and
Digit Manipulation” chapter.
•
Step 3
Cisco AS5300 Universal Access Server
Router(config-dial-peer)# port controller:D
T—(Optional) Control character indicating that the
destination-pattern value is a variable-length dial
string.
Associates this POTS dial peer with a specific voice port.
The keywords and arguments are as follows:
•
controller—Specifies the T1 or E1 controller.
•
:D—Indicates the D channel associated with the ISDN
PRI.
Note
The syntax of the port command is platform specific.
For information on how to configure this command
for your specific device, see the port command
documentation in the “Configuring Voice Ports”
chapter.
Step 4
Router(config-dial-peer)# exit
Exits dial-peer configuration mode.
Step 5
Router(config)# dial-peer voice tag voip
Enters dial-peer configuration mode to configure a VoIP peer.
The tag argument identifies the dial peer. (This number has
local significance only.) Valid entries are from 1 to
2,147,483,647.
Step 6
Router(config-dial-peer)# destination-pattern
[+]string[T]
For an explanation of the command, keywords, and
arguments, see Step 2 of this configuration task table.
Step 7
Router (config-dial-peer)# tech-prefix number
The number argument defines the numbers used as the
technology prefix. Each technology prefix can contain up to
11 characters. Although not strictly necessary, a pound
symbol (#) is frequently used as the last digit in a technology
prefix. Valid characters are 0 though 9, the pound symbol (#),
and the asterisk (*).
Step 8
Router (config-dial-peer)# session target ras
Specifies that the RAS protocol is being used to determine
the IP address of the session target—meaning that a
gatekeeper will translate the E.164 address to an IP address.
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Verifying RAS Configuration
To verify the POTS and VoIP dial-peer configuration, use the show dial-peer voice command. The
following example shows output for a VoIP dial peer using RAS on a Cisco AS5300 universal access
server:
Router# show dial-peer voice 1234
VoiceOverIpPeer1234
tag = 1234, destination-pattern = 1234',
answer-address = ',
group = 1234, Admin state is up, Operation state is up,
incoming called-number = ', connections/maximum = 0/unlimited,
application associated:
type = voip, session-target = ras',
technology prefix: 8#
ip precedence = 0, UDP checksum = disabled,
session-protocol = cisco, req-qos = controlled-load,
acc-qos = best-effort,
fax-rate = voice, codec = g729r8,
Expect factor = 10, Icpif = 30,
VAD = enabled, Poor QOV Trap = disabled,
Troubleshooting Tips
To troubleshoot the dial-peer configuration, perform the following tasks:
•
To display the types and addressing of RAS messages sent and received, use the debug ras
command. The debug output lists the message type using mnemonics defined in ITU-T specification
H.225.
•
To display additional information about the actual contents of the H.225 RAS messages, use the
debug h225 asn1 command.
Configuring AAA Functionality on the Gateway
For the gateway to provide authentication and accounting services, enable and configure your gateway
to support authentication, authorization, and accounting (AAA) services. AAA enables the gateway to
interact with a RADIUS security server to authenticate users (typically incoming calls) and to perform
accounting services. For more information about RADIUS and AAA security services, refer to the
Cisco IOS Security Configuration Guide.
AAA Authentication
The gateway normally uses AAA in conjunction with interactive voice response (IVR) to check the
legitimacy of a prospective gateway user on the basis of an account number (collected by IVR) or
Automatic Number Identification (ANI). When the gateway uses AAA with IVR, the IVR application
collects the user account and personal identification number (PIN) information and then passes it to the
AAA interface. The AAA interface makes a RADIUS authentication request using the given information
and, based on the information received from the RADIUS server, forwards either a pass message or a
fail message to the IVR application.
For more information about configuring IVR, see the “Configuring Interactive Voice Response” chapter.
For more information about authentication services using AAA, refer to the “Configuring
Authentication” chapter in the Cisco IOS Security Configuration Guide.
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AAA Accounting
A call leg is a discrete segment of a call connection that lies between two points in the connection. Each
call made through the gateway consists of two call legs: incoming and outgoing. The RADIUS server
collects basic start-stop connection accounting data or syslog accounting information during the
accounting process for each call leg created on the gateway.
To collect basic start-stop connection accounting data, the gateway must be configured to support
gateway-specific H.323 accounting functionality. The gateway sends accounting data to the RADIUS
server in one of four ways, as is shown in the following sections:
•
Using RADIUS AV Pairs, page 256
•
Appendix , “Using RADIUS AV Pairs”Overloading the Acct-Session-Id Field, page 257
•
Using Vendor-Specific RADIUS Attributes, page 258
•
Using Syslog Records, page 259
Using RADIUS AV Pairs
Basic start-stop connection accounting data and standard RADIUS attributes are used where possible
using standard Internet Engineering Task Force (IETF) RADIUS attribute/value (AV) pairs. Table 20
shows the IETF RADIUS attributes that are supported.
Table 20
Supported IETF RADIUS Accounting Attributes
Number
Attribute
Description
30
Called-Station-Id
Allows the network access server to send the called telephone
number as part of the Access-Request packet (using Dialed
Number Identification Service [DNIS] or similar technology).
This attribute is only supported on ISDN and on modem calls
on the Cisco AS5200 and Cisco AS5300 routers if used with
ISDN PRI.
31
Calling-Station-Id
Allows the network access server to send the calling telephone
number as part of the Access-Request packet (using ANI or
similar technology). This attribute has the same value as the
remote-addr attribute from TACACS+. This attribute is
supported only on ISDN and on modem calls on the
Cisco AS5200 and Cisco AS5300 routers if used with ISDN
PRI.
42
Acct-Input-Octets
Indicates how many octets have been received from the port
over the course of the accounting service being provided.
43
Acct-Output-Octets
Indicates how many octets have been sent to the port over the
course of delivering the accounting service.
44
Acct-Session-Id
Indicates a unique accounting identifier that makes it easy to
match start and stop records in a log file. Acct-Session-Id
numbers restart at 1 each time the router is power-cycled or the
software is reloaded.
47
Acct-Input-Packets
Indicates how many packets have been received from the port
over the course of this service being provided to a framed user.
48
Acct-Output-Packets
Indicates how many packets have been sent to the port in the
course of delivering this service to a framed user.
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For more information about RADIUS and the use of IETF-defined attributes, refer to the Cisco IOS
Security Configuration Guide.
Overloading the Acct-Session-Id Field
Attributes that cannot be mapped to standard RADIUS attributes are packed into the Acct-Session-Id
attribute field as ASCII strings separated by the “/” character. The Acct-Session-Id attribute contains the
RADIUS account session ID, which is a unique identifier that links accounting records associated with
the same login session for a user. To support additional fields, the following string format has been
defined for this field:
<session id>/<call leg setup time>/<gateway id>/<connection id>/<call origin>/
<call type>/<connect time>/<disconnect time>/<disconnect cause>/<remote ip address>
Table 21 shows the field attributes to be used with the Overloaded Acct-Session-Id method and provides
a brief description of each.
Table 21
Field Attributes in Overloaded Acct-Session-Id
Field Attribute
Description
SESSION-ID
Specifies the standard RADIUS account session ID.
SETUP-TIME
Provides the Q.931 setup time for this connection in Network Time
Protocol (NTP) format. NTP time formats are displayed as
%H:%M:%S.%k %Z %tw %tn %td %Y where:
•
%H is hour (00 to 23).
•
%M is minutes (00 to 59).
•
%S is seconds (00 to 59).
•
%k is milliseconds (000 to 999).
•
%Z is time zone string.
•
%tw is day of week (Saturday through Sunday).
•
%tn is month name (January through December).
•
%td is day of month (01 to 31).
•
%Y is year including century (for example, 1998).
GATEWAY-ID
Indicates the name of the underlying gateway in the form of
“gateway.domain_name.”
CALL-ORIGIN
Indicates the origin of the call relative to the gateway. Possible values are
originate and answer.
CALL-TYPE
Indicates call leg type. Possible values are telephony and VoIP.
CONNECTION-ID
Specifies the unique global identifier used to correlate call legs that
belong to the same end-to-end call. The field consists of 4 long words
(128 bits). Each long word is displayed as a hexadecimal value and is
separated by a space character.
CONNECT-TIME
Provides the Q.931 connect time for this call leg, in NTP format.
DISCONNECT-TIME
Provides the Q.931 disconnect time for this call leg, in NTP format.
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Table 21
Field Attributes in Overloaded Acct-Session-Id
Field Attribute
Description
DISCONNECT-CAUSE
Specifies the reason a call was taken offline as defined in the Q.931
specification.
REMOTE-IP-ADDRESS
Indicates the address of the remote gateway port where the call is
connected.
Because of the limited size of the Acct-Session-Id string, it is not possible to embed many information
elements in it. Therefore, this feature supports only a limited set of accounting information elements.
Use the gw-accounting h323 command to configure the overloaded session ID method of applying
H.323 gateway-specific accounting.
Using Vendor-Specific RADIUS Attributes
The IETF draft standard specifies a method for communicating vendor-specific information between the
network access server (NAS) and the RADIUS server by using the vendor-specific attribute (Attribute
26). Vendor-specific attributes (VSAs) allow vendors to support their own extended attributes that are
not suitable for general use. The Cisco RADIUS implementation supports one vendor-specific option
using the format recommended in the specification. The Cisco vendor-ID is 9, and the supported option
has vendor-type 1, which is named “cisco-avpair.” The value is a string of the format:
protocol: attribute sep value *
“Protocol” is a value of the Cisco “protocol” attribute for a particular type of authorization. “Attribute”
and “value” are an appropriate AV pair defined in the Cisco TACACS+ specification, and “sep” is “=”
for mandatory attributes and “*” for optional attributes. The full set of features available for TACACS+
authorization can also be used for RADIUS.
For further information on vendor-specific RADIUS attributes, refer to the RADIUS Vendor-Specific
Attributes Voice Implementation Guide at the following URL:
http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/vapp_dev/vsaig3.htm
The VSA fields and their ASCII values are listed in Table 22.
Table 22
VSA Fields and Their ASCII Values
IETF
RADIUS
Attribute
VendorSpecific
Company
Code
Subtype
Number Attribute Name
26
9
23
h323-remote-address
Indicates the IP address of the remote
gateway.
26
9
24
h323-conf-id
Identifies the conference ID.
26
9
25
h323-setup-time
Indicates the setup time for this
connection in Coordinated Universal
Time (UTC), formerly known as
Greenwich Mean Time (GMT) and Zulu
time.
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Table 22
VSA Fields and Their ASCII Values (continued)
IETF
RADIUS
Attribute
VendorSpecific
Company
Code
Subtype
Number Attribute Name
26
9
26
h323-call-origin
Indicates the origin of the call relative to
the gateway. Possible values are
originating and terminating, which are
equivalent to originate and answer in
the Call-Origin field.
26
9
27
h323-call-type
Indicates call leg type. Possible values
are telephony and VoIP.
26
9
28
h323-connect-time
Indicates the connection time for this
call leg in UTC.
26
9
29
h323-disconnect-time
Indicates the time this call leg was
disconnected in UTC.
26
9
30
h323-disconnect-cause
Specifies the reason a connection was
taken offline per the Q.931 specification.
26
9
31
h323-voice-quality
Specifies the impairment/calculated
planning impairment factor (ICPIF)
affecting voice quality for a call.
26
9
33
h323-gw-id
Indicates the name of the underlying
gateway.
Description
Use the gw-accounting h323 vsa command to configure the VSA method of applying H.323
gateway-specific accounting.
Using Syslog Records
The syslog accounting option exports the information elements associated with each call leg through a
system log message, which can be captured by a syslog daemon on the network. The syslog output
consists of the following:
<server timestamp> <gateway id> <message number> : <message label> : <list of AV pairs>
The syslog message fields are listed in Table 23.
Table 23
Syslog Message Output Fields
Field
Description
server timestamp
The time stamp created by the server when it receives the message
to log.
gateway id
The name of the gateway that emits the message.
message number
The number assigned to the message by the gateway.
message label
A string that identifies the message category.
list of AV pairs
A string consisting of <attribute name> <attribute value> pairs
separated by commas.
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Use the gw-accounting h323 syslog command to configure the syslog record method of gathering H.323
accounting data.
To configure RADIUS authentication and accounting services (as facilitated through authentication,
authorization, and accounting [AAA]), use the following commands in global configuration mode:
Command
Purpose
Step 1
Router(config)# aaa new-model
Enables authentication, authorization, and accounting (AAA)
security services.
Step 2
Router(config)# gw-accounting {h323 [vsa]|
syslog}
Configures gateway-specific H.323 accounting, which may
be h323 or syslog.
The keywords are as follows:
•
h323—Enables standard H.323 accounting using
standard IETF RADIUS attributes.
•
vsa—(Optional) Enables H.323 accounting using
RADIUS vendor-specific attributes.
•
syslog—Enables the system logging facility to output
accounting information in the form of a system message.
Note
Step 3
Router(config)# aaa authentication login h323
radius
Because the Acct-Session-Id attribute is a standard
IETF RADIUS attribute, use the gw-accounting
h323 command to gather accounting data using the
overloaded Acct-Session-Id attribute.
Sets AAA authentication at login.
The keywords are as follows:
Step 4
Router(config)# aaa accounting connection
h323 start-stop radius
•
h323—Defines a method list called h323.
•
radius—Specifies that the RADIUS security protocol be
used.
Defines a method list called h323.
The keywords are as follows:
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•
start-stop—Sends a start accounting notice at the
beginning of a process and a stop accounting notice at the
end of a process. The start accounting record is sent in
the background. The requested user process begins
regardless of whether the start accounting notice was
received by the accounting server.
•
radius—Specifies that only the RADIUS security
protocol be used.
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Step 5
Command
Purpose
Router(config)# radius-server host ip-address
auth-port number acct-port number
Identifies the RADIUS server and the ports that will be used
for authentication and accounting services.
The keywords and arguments are as follows:
Step 6
Router(config)# radius-server key key
•
ip-address—Specifies the IP address of the RADIUS
server host.
•
auth-port—Specifies User Datagram Protocol (UDP)
for authentication requests.
•
number—Specifies the port number for authentication
requests; the host is not used for authentication if set to
0. The default authentication port number is 1645.
•
acct-port—Specifies the UDP destination port for
accounting requests.
•
number—Port number for accounting requests; the host
is not used for accounting if set to 0. The default
accounting port number is 1646.
Specifies the password used between the gateway and the
RADIUS server. The key argument specifies the
authentication and encryption key used between the router
and the RADIUS daemon running on this RADIUS server.
This key overrides the global setting of the radius-server key
command. If no key string is specified, the global value is
used.
The key is a text string that must match the encryption key
used on the RADIUS server. Always configure the key as the
last item in the radius-server host command syntax. This is
because the leading spaces are ignored, but spaces within and
at the end of the key are used. If the key includes spaces, do
not enclose the key in quotation marks unless the quotation
marks themselves are part of the key.
Verifying AAA and RADIUS Configuration
To view the configured RADIUS and AAA parameters for this gateway, use the show running-config
command.
Configuring H.235 Gateway Security
The Cisco H.235-based security and accounting features provide an alternative means for securing
H.323 calls. Before Cisco IOS Release 12.0(7)T, only RAS and AAA were used to configure the security
and accounting functions for H.323 calls. The H.235-based security and accounting features described
in this section can be used by a gatekeeper, which is considered a known and trusted entity, to
authenticate, authorize, and route H.323 calls.
The Cisco H.323 gateway supports the use of CryptoH323Tokens for authentication. The
CryptoH323Token is defined in the ITU-T H.225 Version 2 standard and is used in a
“password-with-hashing” security scheme as described in section 10.3.3 of the H.235 specification.
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A cryptoToken can be included in any RAS message to authenticate the sender of the message. A
separate database can be used for user ID and password verification.
Cisco H.323 gateways support three levels of authentication:
•
Endpoint—The RAS channel used for gateway-to-gatekeeper signaling is not a secure channel. To
ensure secure communication, H.235 allows gateways to include an authentication key in their RAS
messages. This key is used by the gatekeeper to authenticate the source of the messages. At the
endpoint level, validation is performed on all messages from the gateway. The cryptoTokens are
validated using the password configured for the gateway.
Note
To secure the RAS messages and calls, it is essential that the gatekeeper provides
authentication based on the secure key. The gatekeeper must support H.235 security
using the same security scheme as the Cisco gateway.
•
Per-Call—When the gateway receives a call over the telephony leg, it prompts the user for an
account number and PIN. These two numbers are included in certain RAS messages sent from the
endpoint to authenticate the originator of the call.
•
All—This option is a combination of the other two. With this option, the validation of cryptoTokens
in admission request (ARQ) messages is based on an the account number and PIN of the user who
is making a call. The validation of cryptoTokens sent in all the other RAS messages is based on the
password configured for the gateway.
CryptoTokens for registration requests (RRQs), unregistration requests (URQs), disengage requests
(DRQs), and the terminating side of ARQs contain information about the gateway that generated the
token. The cryptoTokens include the gateway identification (ID)—which is the H.323 ID configured on
the gateway—and the gateway password. The cryptoTokens for the originating-side ARQ messages
contain information about the user that is placing the call, including the user ID and PIN.
Although the scenarios in this document describe how to use the security and accounting features in a
prepaid call environment, these features may also be used to authorize IP calls that originate in another
domain (inter-service provider or inter-company calls).
The H.235-based security and accounting features can be used in conjunction with AAA. The gateway
can be configured to use the gatekeeper for call authentication or authorization, and AAA can be used
for call accounting.
In addition, the H.235-based security and accounting features include support for the following:
Note
•
Settlement with the gatekeeper, which allows the gateway to obtain, track, and return accounting
information.
•
Call metering, which allows the gateway to terminate a call if it exceeds the allotted time (in the case
of prepaid calls).
The H.235 security and accounting features described in this document are separate from, and should
not be confused with, the standard interactive voice response (IVR) and AAA features used to
authenticate inbound calls or with the settlement functions provided by the Open Settlement Protocol
(OSP).
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Settlement with the Gatekeeper
The H.235 security and accounting features are designed to support a variety of situations in which some
form of authentication or tracking is required. The security features allow control access through the use
of a userID-password database. The accounting enhancements allow call usage to be tracked at the origin
and at the destination.
Fields have been added to the RAS messages to enhance the accounting capabilities of the Cisco H.323
gateway. These fields allow the gateway to report call-usage information to the gatekeeper. The
call-usage information is included in the DRQ message that is sent when the call is terminated.
Call Tracking
With prepaid calling services, an account number and PIN must be entered and the duration of the call
must be tracked against the remaining credit of the customer. The Cisco H.323 gateway monitors prepaid
account balances and terminates a call if the account is exceeded.
Because the authentication information includes a time stamp, it is important that all the Cisco H.323
gateways and the gatekeepers (or other entity that is performing the authentication) be synchronized.
The Cisco H.323 gateways must be synchronized using the Network Time Protocol (NTP). illustrates
the flow of a possible call for which H.323 security and accounting features are used.Flow for a Call
That Requires H.323 Security and Accounting Features.
Figure 55 illustrates the flow of a possible call for which H.323 security and accounting features are
used.
Figure 55
Flow for a Call That Requires H.323 Security and Accounting Features
Billing sytem
Gatekeeper
A
Billing sytem
20
19
18
17
9
8
5
4
3
2
1
Gateway
A
V
6-7
Telephone A
14
Gateway
B
10
V
20
19
18
17
12
11
5
4
3
2
1
Gatekeeper
B
13
Telephone B
60098
Note
In this example, Telephone A is attempting to establish a phone call to Telephone B. The following
numbered explanations correspond to the action taking place at each numbered reference in Figure 1.
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Gateways Establish Secure Communication with the Gatekeepers
1.
Gateways A and B send gatekeeper request (GRQ) messages to their respective gatekeepers. The
GRQ message includes the authentication capability and the algorithm object ID.
2.
Gatekeepers A and B respond to their respective gateways with gatekeeper confirmation (GCF)
messages. The GCF message includes the authentication capability and the algorithm object ID.
3.
If the values for the H.323 security parameters do not match what is expected, the gatekeeper
responds with a gatekeeper rejection (GRJ) message that contains a reject reason of securityDenial.
This prompts the gateway to resend the GRQ.
4.
Gateways A and B send registration request (RRQ) messages to their respective gatekeepers. The
RRQ message includes authentication information in the cryptoToken field.
5.
Gatekeepers A and B respond to their respective gateways with registration confirmation (RCF)
messages.
If an authentication failure occurs, the gatekeeper responds with a registration rejection (RRJ)
message.
Secure Telephone Communications Are Initiated
6.
Telephone A establishes a connection with Gateway A.
7.
Gateway A initiates the interactive voice response (IVR) script to obtain the account number and
PIN of the user as well as the desired destination telephone number.
8.
Gateway A sends an admission request (ARQ) message to Gatekeeper A. The gateway must include
additional information in the ARQ message to enable the gatekeeper to authenticate the call. The
information included in the ARQ message varies depending on whether the ARQ message is being
sent by the source or the destination gateway. At this point in the scenario, it is the source gateway
that is requesting admission. Therefore, the ARQ message includes the account number and PIN of
the user. This information is encrypted using MD5 hashing and is included in the cryptoTokens field.
9.
Gatekeeper A validates the authentication information, resolves the destination telephone number,
and determines the appropriate destination gateway (which is Gateway B in this case). Then
Gatekeeper A sends an admission confirmation (ACF) message to Gateway A. The ACF message
includes the billing information of the user (such as a reference ID and current account balance for
prepaid call services) and an access token.
10. Gateway A sends a setup message to Gateway B. The setup message also includes the access token.
11. Gateway B sends an ARQ message to Gatekeeper B. The ARQ message includes the access token
received from Gateway A.
12. Gatekeeper B validates the authentication information in the access token and responds to Gateway
B with an ACF message.
If the authentication information is in error, Gatekeeper B sends an admission rejection (ARJ)
message to Gateway B with a reject reason of securityDenial.
13. Gateway B initiates a call to the destination telephone.
14. When the destination telephone is answered, Gateway B sends a connect message to Gateway A.
15. Gateways A and B start their timers to meter the call. If the caller is using prepaid call services, the
meter is constantly compared to the account balance of the user, which was included in the ACF
message sent in Step 9.
Telephone Communications Are Terminated
16. The call is terminated when one of the parties hangs up or, in the case of prepaid call services, when
either of the gateways determines that the account balance of the user has been exceeded.
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17. Gateways A and B send DRQ messages to the their respective gatekeepers. The DRQ message
contains the resulting billing information.
18. Gatekeepers A and B send disengage confirmation (DCF) messages to their respective gateways.
Communication Between the Gateways and the Gatekeepers Is Terminated
19. Gateways A and B send URQ messages to their respective gatekeepers.
20. Gatekeepers A and B send unregistration confirmation (UCF) messages to their respective gateways.
Downloading IVR Scripts
The Tool Command Language (TCL) IVR scripts are the default scripts for all Cisco voice features that
use IVR.
The H.323 security and accounting enhancements described in this document require the use of one of
the following IVR scripts:
Note
•
voip_auth_acct_pin_dest.tcl
•
voip_auth_acct_pin_dest_2.tcl
For more information on TCL IVR applications, see the “Configuring TCL IVR Applications”
chapter.
voip_auth_acct_pin_dest.tcl
The voip_auth_acct_pin_dest.tcl script does the following:
•
Prompts the caller to enter an account number, PIN, and destination number. This information is
provided to an H.323 gatekeeper, which authenticates and authorizes the call.
If the caller is using a debit card account number, the following will occur:
– The gatekeeper returns the remaining credit time amount.
– The TCL script monitors the time remaining and, based on a configured value, plays a “time
running out” message to the caller. The message (such as, “You have only 3 minutes remaining
on your credit.”) is played only to the calling party. The called party hears silence during this
time. For example, if the configured time-out value is 3 minutes, the message is played when
the caller has only 3 minutes of credit left.
– The TCL script plays a warning message when the credit of the user has been exhausted. The
message (such as, “Sorry, you have run out of credit.”) is played only to the calling party. The
called party hears silence during this time.
•
Allows the caller to make subsequent calls to different destinations without disconnecting from the
call leg. Thus, the caller is required to enter the account ID and PIN only once (during initial
authorization). For making subsequent calls, the caller needs to enter only the destination number.
After completing a call to one destination, the caller can disconnect the call by pressing the pound
(#) key on the keypad and holding it down from 1 to 2 seconds. If the # key is pressed down for more
than 1 second, it is treated as a long pound (#). The called party is disconnected, and the caller is
prompted to enter a new destination number. Once a new destination number is entered, the call is
authenticated and authorized using this number and the previously provided account number and
PIN.
This feature also allows the caller to continue making additional calls if the called party hangs up.
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•
Reauthenticates and authorizes each new call. Each time a caller enters a new destination number,
the TCL script reauthenticates or authorizes the call with the gatekeeper and, if the caller is using a
debit card account, obtains the remaining credit time information.
•
Allows the caller to enter the necessary information without having to hear all or any of the prompts.
The TCL script will stop playing (or will not begin playing) the prompt if it detects that the caller
wants to enter the information without listening to the prompt.
Note
The normal terminating character for the account number, PIN, and destination number
is the pound (#) key.
•
Allows the caller to interrupt announcements by pressing the touch tone key. This TCL script stops
playing announcements when the system detects that the caller has pressed any touch tone key.
•
Allows the caller to interrupt partially entered numbers and restart from the beginning by pressing
a designated key on the keypad. The asterisk (*) key is configured as the interrupt key in the TCL
script. The caller can use the asterisk key to cancel an entry and then reenter the account number,
PIN, or destination number. The caller is allowed to re-enter a field only a certain number of times.
The number of retries may be configured. The default is three times.
•
Can terminate a field by size instead of the terminating character (#). The TCL script allows a
specified number of digits to be entered in the account number and PIN fields. This means that the
caller can type all the digits (without the terminating character) and the script determines how to
extract different fields from the number strings. If the caller uses the terminating character, the
terminating character takes precedence and the fields are extracted accordingly.
•
Supports two languages. The IVR script supports two languages, which must be similar in syntax.
The languages must be similar in the manner in which numbers are constructed—especially for
currency, amount, and time. All the prompts are recorded and stored in both languages. The
language selection is made when the caller presses a predefined key in response to a prompt (such
as, “For English, press 1. For Spanish, press 2.”). The TCL script uses the selected language until
the caller disconnects.
voip_auth_acct_pin_dest_2.tcl
The voip_auth_acct_pin_dest_2.tcl script is a simplified version of the voip_auth_acct_pin_dest.tcl
script. It prompts the caller for an account number followed by a PIN. The caller is then prompted for a
destination number. This information is provided to the H.323 gatekeeper that authenticates and
authorizes the call. This script provides prompts only in English.
If the caller is using a debit account number, it plays a “time running out” message when the caller has
10 seconds of credit time remaining. It also plays a “time has expired” message when the credit of the
caller has been exhausted.
H.235 Gateway Security Configuration Tasks
To use the H.235 security features for routing H.323 calls as illustrated above, do the following:
•
Enable H.323 security on the gateway.
•
Download the appropriate TCL IVR scripts from the Cisco Connection Online Software Support
Center. The URL to this site is as follows:
http://www.cisco.com/cgi-bin/tablebuild.pl/tclware
•
Configure the IVR inbound dial peer on the gateway router.
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To enable security on the gateway, use the following commands beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# gateway
Enters gateway configuration mode.
Step 2
Router(config-gateway)# security password
password level {endpoint | per-call | all}
Enables security and specifies the level of validation to be
performed.
The keywords and arguments are as follows:
•
password—Specifies the gateway password.
•
endpoint—Specifies that validation be performed on all
RAS messages sent by the gateway using the
cryptoTokens that are generated based on the security
password configured for the gateway.
•
per-call—Specifies that validation be performed only on
the admission messages from the H.323 endpoints to the
gateway ARQ messages). The gateway prompts the user
for an account number and PIN. These two numbers are
sent from the endpoint and are used to authenticate the
originator of the call.
•
all—This option is a combination of the endpoint and
per-call options. Specifies that validation be performed
on all RAS messages sent by the gateway. The validation
of cryptoTokens in ARQ messages is based on the
account number and PIN of the user making the call, and
the validation of cryptoTokens sent in all other RAS
messages is based on the password configured for the
gateway.
Step 3
Router(config-gateway)# exit
Exits gateway configuration mode.
Step 4
Router(config)# dial-peer voice tag pots
Enters the dial-peer configuration mode to configure a POTS
dial peer. The tag value of the dial-peer voice POTS
command uniquely identifies the dial peer. Valid entries are
from 1 to 2,147,483,647.
Step 5
Router(config-dial-peer)# call application
voice application-name location {word}
Enters the command to initiate the IVR application and the
selected TCL application name. Enter the application name
and the location where the TCL IVR script is stored.
The arguments are as follows:
Step 6
Router(config-dial-peer)# destination-pattern
[+]string[T]
•
application-name—Specifies the character string that
defines the name of the application.
•
location—Specifies the location of the TCL file in URL
format. Valid storage locations are TFTP, FTP, and Flash.
•
word—Specifies the text string that defines an
attribute-value (AV) pair specified by the TCL script and
understood by the RADIUS server.
Specifies the E.164 address associated with this dial peer. For
an explanation of the keywords and arguments, see Step 2 of
the configuration table in the “Configuring Gateway RAS”
section on page 252.
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Step 7
Command
Purpose
Cisco AS5300 Universal Access Server
Configures the voice port associated with this dial peer.
Router(config-dial-peer)# port controller
number:D
•
controller number—Specifies the T1 or E1 controller.
•
:D—Indicates the D channel associated with the ISDN
PRI.
Note
The syntax of the port command is specific to Cisco
hardware platforms. For information on how to
configure this command for a specific device, refer to
the port command documentation in the Cisco IOS
Voice, Video, and Fax Command Reference.
Verifying H.235 Gateway Security Configuration
To display the security password and level when it is enabled, use the show running-config command.
By default, security is disabled.
Router# show running-config
security password 151E0A0E level all
Configuring Alternate Gatekeeper Support
An alternate gatekeeper provides redundancy for a gateway in a system in which gatekeepers are used.
A gateway may use up to two alternate gatekeepers as a backup in the case of a primary gatekeeper
failure.
A gatekeeper manages H.323 endpoints in a consistent manner, allowing them to register with the
gatekeeper and to locate another gatekeeper. The gatekeeper provides logic variables for proxies or
gateways in a call path to provide connectivity with the Public Switched Telephone Network (PSTN), to
improve quality of service (QoS), and to enforce security policies. Multiple gatekeepers may be
configured to communicate with one another, either by integrating their addressing into the DNS or by
using Cisco IOS configuration options.
Alternate gatekeeper support has the following restrictions:
•
This feature can be used only with a gatekeeper that supports the alternate gatekeeper functionality.
•
The timer/retry number of RAS messages remains internal to the gateway as currently implemented.
This feature does not include commands to allow tuning of these parameters.
•
The alternate gatekeeper list is volatile—when the gateway loses power or is reset or reloaded, the
alternate gatekeeper list that has been acquired from the gatekeeper is lost.
Gatekeeper Clustering
With gatekeeper clustering there is the potential that bandwidth may be overcommitted in a cluster. For
example, suppose that there are five gatekeepers in a cluster and that they share 10 Mbps of bandwidth.
Suppose that the endpoints registered to those alternates start placing calls quickly. It is possible that
within a few seconds, each gatekeeper could be allocating 3 Mbps of bandwidth if the endpoints on each
of the gatekeepers request that much bandwidth. The net result is that the bandwidth consumed in the
cluster is 15 Mbps.
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The alternate gatekeeper was purposely designed to restrict bandwidth because there is no clear way to
sync bandwidth information quickly and efficiently. To work around this problem, “announcement”
messages were restricted to intervals as small as 10 seconds. If the gatekeepers get into a situation in
which endpoints request bandwidth rapidly, the problem will be discovered and corrective action will
take place within at least 10 seconds. Assuming that the gatekeepers are probably not all synchronized
on their timers, the announcement messages from the various gatekeepers are likely to be heard more
quickly. Therefore, the problem will be less severe. The potential exists, however, for overcommitment
of the bandwidth between announcement messages if the call volume increases substantially in a short
amount of time (as small as 10 seconds).
Note
If you monitor your bandwidth, it is recommended that you consider lowering the maximum
bandwidth so that if “spikes” such as those described above do occur, some bandwidth will still be
available.
To configure alternate gatekeeper support on a gateway, use the following commands beginning in global
configuration mode:
Command
Purpose
Step 1
Router(config)# interface Ethernet 0/1
Enters interface configuration mode for the selected Ethernet
interface.
Step 2
Router(config-if)# ip address
Identifies the IP address of the Ethernet interface.
Step 3
Router(config-if)# h323-gateway voip
interface
Identifies this interface as a Voice over IP (VoIP) gateway
interface.
Step 4
Router(config-if)# h323-gateway voip id
gatekeeper-id {ipaddr ip-address
[port-number]| multicast} [priority number]
Identifies the gatekeeper for this gateway interface and sets
the attributes.
For an explanation of the keywords and arguments, see
Step 6 in the “Identifying a Router Interface As an H.323
Gateway” section on page 250.
Step 5
Router(config-if)# h323-gateway voip id
gatekeeper-id {ipaddr ip-address
[port-number] | multicast} [priority number]
To identify the alternate gatekeeper, use the following
keywords and arguments:
•
gatekeeper-id—Indicates the H.323 identification of the
gatekeeper. This value must exactly match the gatekeeper
identification (ID) in the gatekeeper configuration. The
recommended format is name.domain-name.
•
ipaddr—Indicates that the gateway will use an IP
address to locate the gatekeeper.
•
ip-address—Defines the IP address used to identify the
gatekeeper.
•
port-number—(Optional) Defines the port number used.
•
multicast—Indicates that the gateway will use multicast
to locate the gatekeeper.
•
priority number—(Optional) Specifies the priority of
this gatekeeper. The range is 1 through 127, and the
default value is 127.
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Step 6
Command
Purpose
Router(config-if)# h323-gateway voip h323-id
interface-id
Identifies the H.323 ID of a particular H.323 endpoint, which
in this case is the gateway. The interface-id argument is the
H.323 name (ID) used by this gateway when this gateway
communicates with its associated gatekeeper. Usually, this ID
is the name of the gateway, with the gatekeeper domain name
appended to the end: [email protected]
Verifying Configuration of the Alternate Gatekeeper
To see that there is an alternate gatekeeper configured, enter the show gate command
Alternate Gatekeeper List
priority 126 id GK1 ipaddr 172.18.193.61 1719
priority 127 id GK2 ipaddr 172.18.193.63 1719
Configuring Dual Tone Multifrequency Relay
Dual tone multifrequency (DTMF) is the tone generated on a touch-tone phone when the keypad digits
are pressed. During a call, DTMF may be entered to access interactive voice response (IVR) systems,
such as voice mail and automated banking services.
Although DTMF is usually transported accurately when using high-bit-rate voice codecs such as G.711,
low-bit-rate codecs such as G.729 and G.723.1 are highly optimized for voice patterns and tend to distort
DTMF tones. As a result, IVR systems may not correctly recognize the tones.
DTMF relay solves the problem of DTMF distortion by transporting DTMF tones “out of band,” or
separate from the encoded voice stream.
For a more thorough explanation of DTMF relay, see the “H.323 Applications” chapter.
To configure DTMF relay on a gateway, use the following commands beginning in global configuration
mode:
Step 1
Command
Purpose
Router(config)# dial-peer voice tag voip
Defines a Voice over IP (VoIP) dial peer and enters dial-peer
configuration mode.
The keywords and argument are as follows:
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•
tag—Indicates the digit that defines a particular dial
peer. Valid entries are from 1 to 2,147,483,647.
•
voip—Indicates that this is a VoIP peer using voice
encapsulation on the POTS network. Use this keyword to
configure DTMF relay.
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H.323 Gateway Configuration Task List
Step 2
Command
Purpose
Router(config-dial-peer)# dtmf-relay
[cisco-rtp] [h245-alphanumeric] [h245-signal]
Forwards DTMF tones.
The keywords are as follows:
Step 3
Router(config-dial-peer)# codec codec
{clear-channel | g711alaw | g711ulaw | g723ar53 |
g723ar63 | g723r53 | g723r63 | g726r16 |
g726r24 |g726r32 | g726r53 | g726r63 |g728 |
g729abr8 | g729ar8 | g729br8 | g729r8 |
gsmefr | gsmfr} [bytes payload_size]
•
cisco-rtp—(Optional) Forwards DTMF tones by using
RTP with a Cisco proprietary payload type.
•
h245-alphanumeric—(Optional) Forwards DTMF
tones by using the H.245 “alphanumeric” User Input
Indication (UII) method. It supports tones 0 through 9, *,
#, and A through D. Use this keyword to configure
DTMF relay.
•
h245-signal—(Optional) Forwards DTMF tones by
using the H.245 “signal” UII method. It supports tones 0
through 9, *, #, and A through D.
Specifies the voice coder rate of speech for a dial peer.
•
The codec keywords are as follows:
– clear-channel—Clear channel at 64,000 bits per
second (bps).
– g711ala—G.711 a-Law at 64,000 bits per second.
– g711ula—G.711 u-Law at 64,000 bps.
– g723ar53—G.723.1 Annex A at 5300 bps.
– g723ar63—G.723.1 Annex A at 6300 bps.
– g723r53—G.723.1 at 5300 bps.
– g723r63—G.723.1 at 6300 bps.
– g726r16—G.726 at 16,000 bps.
– g726r24—G.726 at 24,000 bps.
– g726r32—G.726 at 32,000 bps.
– g728—G.728 at 16,000 bps.
– g729abr8—G.729 Annex A and B at 8000 bps.
– g729ar8—G729 Annex A at 8000 bps.
– g729br8—G.729 Annex B at 8000 bps.
– g729r8—G.729 at 8000 bps. This is the default
codec.
– gsmefr—Global System for Mobile
Communications Enhanced Full Rate (GSMEFR) at
12,200 bps.
– gsmfr—Global System for Mobile Communications
Full Rate (GSMFR) at 13,200 bps.
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Step 2
Command
Purpose
Router(config-dial-peer)# dtmf-relay
[cisco-rtp] [h245-alphanumeric] [h245-signal]
Forwards DTMF tones.
The keywords are as follows:
Step 3
Router(config-dial-peer)# codec codec
{clear-channel | g711alaw | g711ulaw | g723ar53 |
g723ar63 | g723r53 | g723r63 | g726r16 |
g726r24 |g726r32 | g726r53 | g726r63 |g728 |
g729abr8 | g729ar8 | g729br8 | g729r8 |
gsmefr | gsmfr} [bytes payload_size]
•
cisco-rtp—(Optional) Forwards DTMF tones by using
RTP with a Cisco proprietary payload type.
•
h245-alphanumeric—(Optional) Forwards DTMF
tones by using the H.245 “alphanumeric” User Input
Indication (UII) method. It supports tones 0 through 9, *,
#, and A through D. Use this keyword to configure
DTMF relay.
•
h245-signal—(Optional) Forwards DTMF tones by
using the H.245 “signal” UII method. It supports tones 0
through 9, *, #, and A through D.
Specifies the voice coder rate of speech for a dial peer.
•
The codec keywords are as follows:
– clear-channel—Clear channel at 64,000 bits per
second (bps).
– g711ala—G.711 a-Law at 64,000 bits per second.
– g711ula—G.711 u-Law at 64,000 bps.
– g723ar53—G.723.1 Annex A at 5300 bps.
– g723ar63—G.723.1 Annex A at 6300 bps.
– g723r53—G.723.1 at 5300 bps.
– g723r63—G.723.1 at 6300 bps.
– g726r16—G.726 at 16,000 bps.
– g726r24—G.726 at 24,000 bps.
– g726r32—G.726 at 32,000 bps.
– g728—G.728 at 16,000 bps.
– g729abr8—G.729 Annex A and B at 8000 bps.
– g729ar8—G729 Annex A at 8000 bps.
– g729br8—G.729 Annex B at 8000 bps.
– g729r8—G.729 at 8000 bps. This is the default
codec.
– gsmefr—Global System for Mobile
Communications Enhanced Full Rate (GSMEFR) at
12,200 bps.
– gsmfr—Global System for Mobile Communications
Full Rate (GSMFR) at 13,200 bps.
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Command
Step 4
Router(config-dial-peer)# destination-pattern
[+]string[T]
Purpose
•
bytes—(Optional) Specifies the number of bytes in the
voice payload of each frame.
•
payload-size—(Optional) The number of bytes in the
voice payload of each frame. Refer to the codec
(dial-peer) command table titled “Voice
Payload-Per-Frame Options and Defaults” in the
Cisco IOS Voice, Video, and Fax Command Reference for
valid entries and default values.
Specifies the prefix, the full E.164 telephone number, or an
ISDN directory number to be used for a dial peer (depending
on the dial plan).
For an explanation of the keywords and arguments, see
Step 2 of the configuration task table in the “Configuring
Gateway RAS” section on page 252.
Step 5
Cisco 2600 and 3600 Series Routers
Router(config-dial-peer)# session target
{ipv4:destination-address | dns:[$s$. | $d$.
| $e$. | $u$.] host-name |
loopback:rtp | loopback:compressed |
loopback:uncompressed}
Cisco AS5300 Universal Access Server
Specifies a network-specific address for a specified dial peer
or destination gatekeeper.
Keywords and arguments are as follows:
Cisco 2600 and 3600 Series Routers
•
ipv4:destination-address—IP address of the dial peer.
•
dns:host-name—Indicates that the DNS will be used to
resolve the name of the IP address. Valid entries for this
parameter are characters representing the name of the
host device. (Optional) You can use one of the following
four wildcards with this keyword when defining the
session target for VoIP peers:
Route(config-dial-peer)# session target
{ipv4:destination-address | dns:[$s$. | $d$.
| $e$. | $u$.] host-name |
loopback:rtp | loopback:compressed |
loopback:uncompressed | mailto:{name |
$d$.}@domain-name}
– $s$.—Indicates that the source destination pattern
will be used as part of the domain name.
– $d$.—Indicates that the destination number will be
used as part of the domain name.
– $e$.—Indicates that the destination pattern is used
as part of the domain name in reverse dotted format
for tpc.int DNS format. For example, if the
destination number is 310 555-1234 and the session
target is configured as $e$.cisco.com, the translated
DNS name will be 4.3.2.1.5.5.5.0.1.3.cisco.com.
– $u$.—Indicates that the unmatched portion of the
destination pattern (such as a defined extension
number) will be used as part of the domain name.
•
loopback:rtp—Indicates that all voice data will be
looped back to the originating source. This is applicable
for VoIP peers.
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Command
Purpose
•
loopback:compressed—Indicates that all voice data
will be looped back in compressed mode to the
originating source. This is applicable for POTS peers.
•
loopback:uncompressed—Indicates that all voice data
will be looped back in uncompressed mode to the
originating source. This is applicable for POTS peers.
Cisco AS5300 Universal Access Server
In addition to the above, the following keywords and
arguments apply to the Cisco AS5300 universal access
server:
•
mailto:name—Specific recipient e-mail address, name,
or mailing list alias.
•
@domain-name—Specifies
the appropriate domain name
associated with the e-mail address.
Configuring FXS Hookflash Relay
A “hookflash” indication is a brief on-hook condition during a call. The indication is not long enough in
duration to be interpreted as a signal to disconnect the call.
PBXs and telephone switches are frequently programmed to intercept hookflash indications and use
them as a way to allow a user to invoke supplemental services. In a traditional telephone network, a
hookflash results in a voltage change on the telephone line. Because there is no equivalent of this voltage
change in an IP network, a message may be sent over the IP network that represents a hookflash. To send
a hookflash indication using this message, an H.323 endpoint sends an H.245 user input indication
message that contains an “H.245-signal” or “H.245-alpha” structure.
Hookflash relay is supported on the Cisco 2600, 3600, and 7200 series routers and on the MC3810
multiservice concentrator.
For a further explanation of configuring hookflash relay, see the “H.323 Applications” chapter.
To configure hookflash relay on a gateway, use the following commands beginning in global
configuration mode:
Note
Hookflash relay is enabled only when the dtmf-relay h245-signal command is configured on the
applicable VoIP dial peers. Hookflash is relayed using an h245-signal indication and can be sent only
when an h245-signal is available.
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Step 1
Command
Purpose
Cisco 2600 and 3600 Series Routers
Enters voice-port configuration mode.
Router(config)# voice-port
{slot-number/subunit-number/port} |
{slot/port:ds0-group-no}
The keywords and arguments are as follows:
Cisco 2600 and 3600 Series Routers
Cisco 7200 Series Routers
•
slot-number—Specifies the slot number in the Cisco
router in which the voice interface card is installed. Valid
entries are from 0 to 3, depending on the slot in which it
has been installed.
•
subunit-number—Specifies the subunit on the voice
interface card in which the voice port is located. Valid
entries are 0 or 1.
•
port—Indicates the voice port. Valid entries are 0 or 1.
•
slot—Specifies the router location in which the voice
port adapter is installed. Valid entries are from 0 to 3.
Router(config)# voice-port
{slot/port:ds0-group-no} |
{slot-number/subunit-number/port}
Cisco MC3810 Multiservice Concentrator
Router(config)# voice-port slot/port
Cisco 7200 Series Routers
•
slot—Specifies the router location in which the voice
port adapter is installed. Valid entries are from 0 to 3.
•
port—Indicates the voice interface card location. Valid
entries are 0 or 1.ds0-group-no—Defines DS0 group
number. Each defined DS0 group number is represented
on a separate voice port. This allows you to define
individual DS0s on the digital T1/E1 card.
•
slot-number—Indicates the slot number in the Cisco
router in which the voice interface card is installed. Valid
entries are from 0 to 3, depending on the slot in which it
has been installed.
• subunit-number—Indicates the subunit on the voice
interface card in which the voice port is located. Valid
entries are 0 or 1.
•
port—Indicates the voice port number. Valid entries are
0 or 1.
Cisco MC3810 Multiservice Concentrator
•
slot—Specifies the slot number in the Cisco router in
which the voice interface card is installed. The only valid
entry is 1.
•
port—Specifies the voice port number. Valid ranges are
as follows:
– Analog voice ports: from 1 to 6.
– Digital T1: from 1 to 24.
– Digital E1: from 1 to 15 and from 17 to 31.
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Command
Purpose
Step 2
Router(config-voice-port)# timing
hookflash-input duration
Specifies the maximum duration of a hookflash indication. If
the hookflash lasts longer than the specified limit, the
Foreign Exchange Station (FXS) interface processes the
indication as an on-hook. The duration is shown in
milliseconds. Possible values are 50 through 1550. The
default value is 600 milliseconds.
Step 3
Router(config-voice-port)# timing
hookflash-out duration
Specifies the duration, in milliseconds, of the hookflash
indications that the gateway generates on a Foreign Exchange
Office (FXO) interface. Valid entries are from 50 through
1550 milliseconds. The default is 400 milliseconds.
Configuring Multiple Codecs
Normally only one codec is specified when a dial peer is configured on a gateway. However, a prioritized
list of codecs can be configured to increase the probability of establishing a connection between
endpoints during the H.245 exchange phase. For more information about configuring multiple codecs,
see the “Codec Negotiation” section in the “H.323 Applications” chapter.
To configure multiple codecs for a dial peer, use the following commands beginning in global
configuration mode:
Command
Purpose
Step 1
Router(config)# voice class codec tag
Enters voice class configuration mode and assigns an
identification tag number for a codec voice class. The tag
argument is the unique number assigned to the voice class.
The valid range is from 1 to 10,000. Each tag number must be
unique on the router.
Step 2
Router(config-class)# codec preference value
codec-type [bytes payload-size]
Adds codecs to the prioritized list of codecs.
The keywords and arguments are as follows:
•
value—Specifies the order of preference, with 1 being
the most preferred and 12 being the least preferred.
•
codec-type—Specifies the type of codec preferred. Types
are as follows:
– clear-channel—Clear channel at 64,000 bps.
– g711alaw—G.711a-Law at 64,000 bps.
– g711ulaw—G.711 u-Law at 64,000 bps.
– g723ar53—G.723.1 Annex-A at 5300 bps.
– g723ar63—G.723.1 Annex-A at 6300 bps.
– g723r53—G.723.1 at 5300 bps.
– g723r63—G.723.1 at 6300 bps.
– g726r16—G.726 at 16,000 bps.
– g726r24—G.726 at 24,000 bps.
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– g726r32—G.726 at 32,000 bps.
– g728—G.728 at 16,000 bps.g729abr8—G.729
Annex-A and B at 8000 bps.
– g729br8—G.729 Annex-B at 8000 bps.
– g729r8—G.729 at 8000 bps. This is the default
codec.
– gsmefr—Global System for Mobile
Communications Enhanced Full Rate (GSMEFR) at
12,200 bps.
– gsmfr—Global System for Mobile Communications
Full Rate (GSMFR) at 13,200 bps.
•
bytes—(Optional) Specifies that the size of the voice
frame is in bytes.
– payload-size—(Optional) Number of bytes that you
specify as the voice payload of each frame. Values
depend on the codec type and the packet voice
protocol.
Step 3
Router(config-class)# exit
Exits voice class configuration mode.
Step 4
Router(config)# dial-peer voice tag voip
Enters dial-peer configuration mode to configure a VoIP peer.
The tag value of the dial-peer voice VoIP command uniquely
identifies the dial peer. (This number has local significance
only.)
Step 5
Router(config-dial-peer)# voice-class codec
tag
The tag is the unique number assigned to the voice class. The
valid range for this tag is from 1 to 10,000. The tag number
maps to the tag number created using the voice class codec
global configuration command.
Verifying Multiple Codecs Configuration
To show the codecs defined for a particular prioritized list of codecs, use the show running-config
command.
Configuring Rotary Calling Pattern
Rotary calling pattern routes an incoming call that arrives over a telephony interface back out through
another telephony interface under certain circumstances. Rotary calling pattern primarily provides
reliable service during network failures. Call establishment using rotary calling pattern is supported by
rotary group support of dial peers, where multiple dial peers may match a given destination phone
number and be selected in sequence. In addition, if the destinations need to be tried in a certain order,
preference may be assigned. Use the preference command when configuring the dial peers to reflect the
preferred order (0 being the highest preference and 10 the lowest).
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If several dial peers match a particular destination pattern, the system attempts to place a call to the dial
peer configured with the highest preference. If the call cannot be completed because of a system outage
(for example, the gatekeeper or gateway cannot be contacted), the rotary call pattern performs the
following tasks:
•
Lists all the conditions under which this instance occurs.
•
Retries the call to the next highest preference dial peer.
•
Continues until no more matching dial peers are found.
If there are equal priority dial peers, the order is determined randomly.
Note
The hunting algorithm precedence may be configured. See the preference command discussed in the
“Configuring Dial Plans, Dial Peers, and Digit Manipulation” chapter.
Configuring H.323 Support for Virtual Interfaces
H.323 support for virtual interfaces allows the IP address of the gateway to be configured so that the IP
address included in the H.323 packet is always the source IP address of the gateway, regardless of the
physical interface and protocol used. This single-address feature allows firewall applications to be easily
configured to work with H.323 messages.
To configure a source IP address for a gateway, use the following commands beginning in global
configuration mode:
Step 1
Command
Purpose
Router(config)# interface type slot/port
Enters interface configuration mode to configure parameters
for the specified interface.
The keywords and arguments are as follows:
Step 2
Router(config-if)# h323-gateway voip bind
srcaddr ip-address
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•
type—Indicates the type of interface.
•
slot—Specifies the number of the slot being configured.
•
port—Specifies the number of the port being configured.
Note
This syntax will vary, depending on the platform.
Sets the source IP address to be used for this gateway. The
ip-address argument specifies the IP address to be used for
outgoing H.323 traffic, which includes H.225, H.245, and
RAS messages. Typically, this is the IP address assigned to
the Ethernet interface.
Configuring H.323 Gateways
H.323 Gateway Configuration Examples
Verifying the Source IP Address of the Gateway
To verify the source IP address of the gateway, enter the show running-config command. The output
shows the source IP address that is bound to the interface.
router# show running-config
interface Loopback0
ip address 10.0.0.0 255.255.255.0
no ip directed-broadcast
h323-gateway voip bind srcaddr 10.0.0.0
!
interface Ethernet0/0
ip address 172.18.194.50 255.255.255.0
no ip directed-broadcast
h323-gateway voip interface
h323-gateway voip id j70f_2600_gk2 ipaddr 172.18.194.53 1719
h323-gateway voip h323-id j70f_3640_gw1
h323-gateway voip tech-prefix 3#
.
.
.
In the following example, Ethernet interface 0/0 is used as the gateway interface. For convenience, the
h323-gateway voip bind srcaddr command has been specified on the same interface. The designated
source IP address is the same as the IP address assigned to the interface.
interface Ethernet0/0
ip address 172.18.194.50 255.255.255.0
no ip directed-broadcast
h323-gateway voip interface
h323-gateway voip id j70f_2600_gk2 ipaddr 172.18.194.53 1719
h323-gateway voip h323-id j70f_3640_gw1
h323-gateway voip tech-prefix 3#
h323-gateway voip bind srcaddr 172.18.194.50
H.323 Gateway Configuration Examples
This section includes the following gateway configuration examples:
•
H.323 Gateway RAS Configuration Example, page 280
•
AAA Functionality on the Gateway Configuration Example, page 281
•
H.323 Gateway Security Configuration Example, page 284
•
H.235 Security Example, page 286
•
Alternate Gatekeeper Configuration Example, page 286
•
DTMF Relay Configuration Example, page 287
•
FXS Hookflash Relay Configuration Example, page 287
•
Multiple Codec Configuration Example, page 287
•
Rotary Calling Pattern Configuration Example, page 287
•
H.323 Support for Virtual Interfaces Configuration Example, page 288
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H.323 Gateway RAS Configuration Example
Figure 56 shows a Cisco 2600 router and a Cisco AS5800 universal access server as gateways and a
Cisco 3640 router as a gatekeeper.
Figure 56
VoIP for the Cisco AS5800
AS5800 VoIP
H.323 gateway
100BASE-T
Cisco 2600
10BASE-T
10BASE-T
10BASE-T
Catalyst
5000
NT Server
Cisco CallManager
10BASE-T
Cisco 3640
gatekeeper
30460
Cisco 2600
5000
The following example shows a Cisco AS5800 universal access server configured as a gateway using
RAS:
! Configure the T1 controller. (This configuration is for a T3 card.)
controller T1 1/0/0:1
framing esf
linecode b8zs
pri-group timeslots 1-24
!
! Configure POTS and VoIP dial peers.
dial-peer voice 11111 pots
incoming called-number 12345
destination-pattern 9#11111
direct-inward-dial
port 1/0/0:1:D
prefix 11111
!
dial-peer voice 12345 voip
destination-pattern 12345
tech-prefix 6#
session target ras
!
! Enable gateway functionality.
gateway
!
! Enable Cisco Express Forwarding.
ip cef
!
! Configure and enable the gateway interface.
interface FastEthernet0/3/0
ip address 172.16.0.0.255.255.255.0
no ip directed-broadcast
no keepalive
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full-duplex
no cdp enable
h323-gateway voip interface
h323-gateway voip id gk3.gg-dn1 ipaddr 172.18.0.0 1719
h323-gateway voip h323-id [email protected]
h323-gateway voip tech-prefix 9#
!
! Configure the serial interface.(This configuration is for a T3 serial interface.)
interface Serial1/0/0:1:23
no ip address
no ip directed-broadcast
ip mroute-cache
isdn switch-type primary-5ess
isdn incoming-voice modem
no cdp enable
AAA Functionality on the Gateway Configuration Example
The following example shows AAA functionality configured on a Cisco AS5300 universal access server.
version 12.2
no service single-slot-reload-enable
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
service password-encryption
service tcp-small-servers
!
hostname doc-rtr53-05
aaa new-model
aaa authentication login default local
aaa authentication login NO_AUTHENT none
aaa authentication login h323 group radius
aaa authentication ppp default if-needed local
aaa authorization exec default local if-authenticated
aaa authorization exec NO_AUTHOR none
aaa authorization commands 15 default local if-authenticated
aaa authorization commands 15 NO_AUTHOR none
aaa accounting exec default start-stop group tacacs+
aaa accounting exec NO_ACCOUNT none
aaa accounting commands 15 default stop-only group tacacs+
aaa accounting commands 15 NO_ACCOUNT none
aaa accounting connection h323 start-stop group radius
enable secret 5 $1$l545$V4haZZN8AOKem8T1DhF5i/
enable password 7 060506324F41
!
resource-pool disable
!
call rsvp-sync
ip subnet-zero
no ip source-route
no ip finger
ip domain-name the.net
ip name-server 172.22.53.210
ip name-server 172.19.23.12
!
controller T1 0
framing esf
clock source line primary
linecode b8zs
pri-group timeslots 1-24
!
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controller T1 1
framing esf
linecode b8zs
pri-group timeslots 1-24
!
controller T1 2
framing esf
linecode b8zs
pri-group timeslots 1-24
!
controller T1 3
framing esf
clock source line secondary 3
linecode b8zs
pri-group timeslots 1-24
!
gw-accounting h323
gw-accounting voip
translation-rule 1
Rule 1 408555.... 5
Rule 2 650 5
!
interface Loopback0
ip address 172.21.10.10 255.255.255.255
!
interface Loopback1
ip address 172.21.104.254 255.255.255.0
!
interface Ethernet0
no ip address
shutdown
!
interface Virtual-Template1
description Template for Multilink Users
ip unnumbered Loopback1
no logging event link-status
no snmp trap link-status
peer default ip address pool addr-pool
ppp authentication chap
ppp multilink
!
interface Serial0:23
description description Headquarters 324-1937 active PRI line
no ip address
no logging event link-status
no snmp trap link-status
isdn switch-type primary-5ess
isdn incoming-voice modem
fair-queue 64 256 0
no cdp enable
!
interface Serial1:23
no ip address
no logging event link-status
no snmp trap link-status
isdn switch-type primary-5ess
isdn incoming-voice modem
fair-queue 64 256 0
no cdp enable
!
interface Serial2:23
no ip address
no logging event link-status
no snmp trap link-status
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isdn switch-type primary-5ess
isdn incoming-voice modem
fair-queue 64 256 0
no cdp enable
!
interface Serial3:23
no ip address
no logging event link-status
no snmp trap link-status
isdn switch-type primary-5ess
isdn incoming-voice modem
fair-queue 64 256 0
no cdp enable
!
interface FastEthernet0
ip address 172.21.101.23 255.255.255.0
duplex auto
speed auto
!
radius-server host 10.10.10.10 auth-port 1612 acct-port 1616
radius-server retransmit 3
radius-server key 7 00071C080A520E
!
dial-peer voice 1 pots
!
dial-peer voice 2 voip
destination-pattern +1234...
session target ipv4:10.1.1.1
!
dial-peer voice 3 voip
destination-pattern 555*
session target ipv4:10.1.1.2
!
dial-peer voice 4 voip
destination-pattern 555
session target ipv4:10.1.2.2
!
dial-peer voice 90 voip
destination-pattern 555.%
session target ipv4:10.1.2.2
!
dial-peer voice 50 voip
destination-pattern 408555%
session target ipv4:10.1.1.2
!
dial-peer voice 55 voip
destination-pattern 408555.%
session target ipv4:10.2.2.2
!
line con 0
exec-timeout 0 0
authorization commands 15 NO_AUTHOR
authorization exec NO_AUTHOR
login authentication NO_AUTHENT
transport input none
line 1 48
autoselect during-login
autoselect ppp
modem InOut
transport preferred none
transport output telnet
line aux 0
line vty 0 4
password 7 03470A1C140635495C
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transport preferred none
transport input telnet
transport output telnet
!
!
!ntp clock-period 17180261
ntp update-calendar
ntp server 172.22.255.1 prefer
H.323 Gateway Security Configuration Example
The following example illustrates H.323 security configuration on a Cisco AS5300 gateway.
hostname um5300
!
enable password xyz
!
resource-pool disable
!
clock timezone EST -5
clock summer-time EDT recurring
ip subnet-zero
no ip domain-lookup
!
isdn switch-type primary-5ess
isdn voice-call-failure 0
call application voice xyz tftp://172.18.16.2/samp/xyz.tcl
call application voice load xys
mta receive maximum-recipients 1024
!
xgcp snmp sgcp
!
controller T1 0
framing esf
clock source line primary
linecode b8zs
pri-group timeslots 1-24
!
controller T1 1
framing esf
clock source line secondary 1
linecode b8zs
pri-group timeslots 1-24
!
controller T1 2
!
controller T1 3
!
voice-port 0:D
!
voice-port 1:D
!
dial-peer voice 4001 pots
application xyz
destination-pattern 4003
port 0:D
prefix 4001
!
dial-peer voice 513 voip
destination-pattern 1513200....
session target ras
!
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dial-peer voice 9002 voip
destination-pattern 9002
session target ras
!
dial-peer voice 4191024 pots
destination-pattern 4192001024
port 0:D
prefix 4001
!
dial-peer voice 1513 voip
destination-pattern 1513.......
session target ras
!
dial-peer voice 1001 pots
destination-pattern 14192001001
port 0:D
!
gateway
security password 151E0A0E level all
!
interface Ethernet0
ip address 10.99.99.7 255.255.255.0
no ip directed-broadcast
shutdown
!
interface Serial0:23
no ip address
no ip directed-broadcast
isdn switch-type primary-5ess
isdn protocol-emulate user
isdn incoming-voice modem
fair-queue 64 256 0
no cdp enable
!
interface Serial1:23
no ip address
no ip directed-broadcast
isdn switch-type primary-5ess
isdn protocol-emulate user
isdn incoming-voice modem
isdn guard-timer 3000
isdn T203 10000
fair-queue 64 256 0
no cdp enable
!
interface FastEthernet0
ip address 172.18.72.121 255.255.255.192
no ip directed-broadcast
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip id [email protected] ipaddr 172.18.72.58 1719
h323-gateway voip h323-id um5300
h323-gateway voip tech-prefix 1#
!
no ip http server
ip classless
ip route 10.0.0.0 172.18.72.65
!
!
line con 0
exec-timeout 0 0
length 0
transport input none
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line aux 0
line vty 0 4
password xyz
login
!
ntp clock-period 17179974
ntp server 172.18.72.124
H.235 Security Example
The following example shows output from configuring secure registrations from the gatekeeper and
identifying which RAS messages the gatekeeper will check to find authentication tokens:
dial-peer voice 10 voip
destination-pattern 4088000
session target ras
dtmf-relay h245-alphanumeric
!
gateway
security password 09404F0B level endpoint
The following example shows output from configuring which RAS messages will contain gateway
generated tokens:
dialer-list 1 protocol ip permit
dialer-list 1 protocol ipx permit
radius-server host 10.25.0.0 auth-port 1645 acct-port 1646
radius-server retransmit 3
radius-server deadtime 5
radius-server key lab
radius-server vsa send accounting
!
gatekeeper
zone local GK1 test.com 10.0.0.3
zone remote GK2 test2.com 10.0.2.2 1719
accounting
security token required-for registration
no use-proxy GK1 remote-zone GK2 inbound-to terminal
no use-proxy GK1 remote-zone GK2 inbound-to gateway
no shutdown
Alternate Gatekeeper Configuration Example
In the following example, the gateway is configured to have alternate gatekeepers. The primary and
secondary gatekeepers are configured with the priority option. The priority range is 1 through 127. The
first alternate gatekeeper has been configured as priority 120; the second alternate gatekeeper has not
been configured, so it remains at the default setting of 127.
interface Ethernet 0/1
ip address 172.18.193.59 255.255.255.0
h323-gateway voip interface
h323-gateway voip id GK1 ipaddr 172.18.193.65 1719 priority 120
h323-gateway voip id GK2 ipaddr 172.18.193.66 1719
h323-gateway voip h323-id cisco2
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DTMF Relay Configuration Example
The following example shows DTMF relay configured on a gateway.
dial-peer voice 1 voip
dtmf-relay h245-alphanumeric
codec g723r53
destination-pattern 5....
session target ipv4:192.168.100.2
FXS Hookflash Relay Configuration Example
The following example shows how to implement hookflash-in and hookflash-out timing for the
hookflash after voice port 1/0/0 has been configured.
voice-port 1/0/0
timing hookflash-in 200
timing hookflash-out 200
Multiple Codec Configuration Example
The following configuration shows how to create a list of prioritized codecs and apply that list to a
specific VoIP dial peer:
voice class codec
codec preference
codec preference
codec preference
codec preference
codec preference
codec preference
codec preference
codec preference
codec preference
codec preference
codec preference
codec preference
99
1 g711alaw
2 g711ulaw bytes 80
3 g723ar53
4 g723ar63 bytes 144
5 g723r53
6 g723r63 bytes 120
7 g726r16
8 g726r24
9 g726r32 bytes 80
10 g728
11 g729br8
12 g729r8 bytes 50
!
dial-peer voice 1919 voip
voice-class codec 99
Rotary Calling Pattern Configuration Example
The following example configures POTS dial peer 10 for a preference of 1, POTS dial peer 20 for a
preference of 2, and Voice over Frame Relay dial peer 30 for a preference of 3:
dial-peer voice 10 pots
destination pattern 5552150
preference 1
dial-peer voice 20 pots
destination pattern 5552150
preference 2
dial-peer voice 30 vofr
destination pattern 5552150
preference 3
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H.323 Support for Virtual Interfaces Configuration Example
In the following example, Ethernet interface 0/0 is used as the gateway interface. For convenience, the
h323-gateway voip bind srcaddr command has been specified on the same interface. The designated
source IP address is the same as the IP address assigned to the interface.
interface Ethernet0/0
ip address 172.18.194.50 255.255.255.0
no ip directed-broadcast
h323-gateway voip interface
h323-gateway voip id j70f_2600_gk2 ipaddr 172.18.194.53 1719
h323-gateway voip h323-id j70f_3640_gw1
h323-gateway voip tech-prefix 3#
h323-gateway voip bind srcaddr 172.18.194.50
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Configuring H.323 Gatekeepers and Proxies
This chapter describes how to configure the Cisco Multimedia Conference Manager. The Multimedia
Conference Manager provides gatekeeper and proxy capabilities required for service provisioning and
management of H.323-compliant networks.
This chapter includes the following sections:
•
Multimedia Conference Manager Overview, page 289
•
H.323 Gatekeeper Features, page 290
•
H.323 Proxy Features, page 297
•
H.323 Prerequisite Tasks and Restrictions, page 302
•
H.323 Gatekeeper Configuration Task List, page 303
•
H.323 Gatekeeper Configuration Examples, page 345
For a complete description of the H.323 gatekeeper commands used in this chapter, refer to the
Cisco IOS Voice, Video, and Fax Command Reference. To locate documentation for other commands that
appear in this chapter, use the command reference master index or search online.
For more information regarding Resource Reservation Protocol (RSVP), synchronous reservation
timers, and slow connect, refer to the Cisco IOS Release 12.1(5)T VoIP Call Admission Control Using
RSVP or the Cisco IOS Quality of Service Solutions Configuration Guide.
To identify the hardware platform or software image information associated with a feature in this
chapter, use the Feature Navigator on Cisco.com to search for information about the feature or refer to
the software release notes for a specific release. For more information, see the “Identifying Supported
Platforms” section in the “Using Cisco IOS Software” chapter.
Multimedia Conference Manager Overview
Deploying H.323 applications and services requires careful design and planning for the network
infrastructure and for the H.323 devices. The Cisco H.323-compliant Multimedia Conference Manager
provides gatekeeper and proxy capabilities, which are required for service provisioning and management
of H.323 networks. With the Cisco Multimedia Conference Manager, your current internetwork can be
configured to route bit-intensive data, such as audio, telephony, video and audio telephony, and data
conferencing using existing telephone and ISDN links without degrading the current level of service of
the network. In addition, H.323-compliant applications can be implemented on existing networks in an
incremental fashion without upgrades.
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H.323 Gatekeeper Features
Multimedia Conference Manager provides a rich list of networking capabilities, including the following:
•
A means to implement quality of service (QoS), which is required for the successful deployment of
H.323 applications.
•
Interzone routing in the E.164 address space. When using H.323-identification (H.323-ID) format
addresses, interzone routing is accomplished by using domain names.
Multimedia Conference Manager allows you to do the following:
•
Identify H.323 traffic and apply appropriate policies.
•
Limit H.323 traffic on the LAN and WAN.
•
Provide user accounting for records based on service use.
•
Insert QoS for the H.323 traffic generated by applications such as Voice over IP (VoIP), data
conferencing, and video conferencing.
•
Implement security for H.323 communications.
Principal Multimedia Conference Manager Functions
The H.323-compliant Multimedia Conference Manager has two principal functions: gatekeeper and
proxy. Gatekeeper subsystems provide the following features:
•
User authorization in which authentication, authorization, and accounting (AAA) account holders
are permitted to register and use the services of the gatekeeper application.
•
Accounting using AAA call detail records.
•
Zone bandwidth management to limit the number of active sessions.
•
H.323 call routing.
•
Address resolution.
Cisco Multimedia Conference Managers can be configured to use the Cisco Hot Standby Router Protocol
(HSRP) so that when one gatekeeper fails, the standby gatekeeper assumes its role.
Proxy subsystems provide the following features:
•
H.323 traffic consolidation.
•
Tight bandwidth controls.
•
QoS mechanisms such as IP Precedence and RSVP.
•
Secure communication over extranets.
H.323 Gatekeeper Features
The following sections describe the main features of a gatekeeper in an H.323 network:
•
Zone and Subnet Configuration, page 291
•
Redundant H.323 Zone Support, page 291
•
Gatekeeper-to-Gatekeeper Redundancy and Load-Sharing Mechanism, page 292
•
Interzone Communication, page 293
•
RADIUS and TACACS+, page 293
•
Accounting via RADIUS and TACACS+, page 293
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•
Interzone Routing Using E.164 Addresses, page 294
•
HSRP Support, page 296
Zone and Subnet Configuration
A zone is defined as the set of H.323 nodes controlled by a single gatekeeper. Gatekeepers that coexist
on a network may be configured so that they register endpoints from different subnets.
Endpoints attempt to discover a gatekeeper and consequently the zone of which they are members by
using the Registration, Admission, and Status (RAS) message protocol. The protocol supports a
discovery message that may be sent multicast or unicast.
If the message is sent multicast, the endpoint registers nondeterministically with the first gatekeeper that
responds to the message. To enforce predictable behavior, where endpoints on certain subnets are
assigned to specific gatekeepers, the zone subnet command can be used to define the subnets that
constitute a given gatekeeper zone. Any endpoint on a subnet that is not enabled for the gatekeeper will
not be accepted as a member of that gatekeeper zone. If the gatekeeper receives a discovery message
from such an endpoint, it will send an explicit reject message.
Redundant H.323 Zone Support
Redundant H.323 zone support allows for the following:
•
Gatekeeper Multiple Zone Support, page 291
•
Gateway Support for Alternate Gatekeepers, page 291
•
Zone Prefixes, page 291
•
Technology Prefixes, page 292
Gatekeeper Multiple Zone Support
Redundant H.323 zone support allows users to configure multiple remote zones to service the same zone
or technology prefix. A user is able to configure more than one remote gatekeeper to which the local
gatekeeper can send location requests (LRQs). This allows for more reliable call completion.
Redundant H.323 zone support is supported on all gatekeeper-enabled IOS images.
Gateway Support for Alternate Gatekeepers
Redundant H.323 zone support in the gateway allows a user to configure two gatekeepers in the gateway
(one as the primary and the other as the alternate). All gatekeepers are active. The gateway can choose
to register with any one (but not both) at a given time. If that gatekeeper becomes unavailable, the
gateway registers with the other.
Redundant H.323 zone support is supported on all gateway-enabled images.
Zone Prefixes
The zone prefixes (typically area codes) serve the same purpose as the domain names in the H.323-ID
address space.
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For example, the local gatekeeper can be configured with the knowledge that zone prefix “212......” (that
is, any address beginning “212” and followed by 7 arbitrary digits) is handled by the gatekeeper
gatekeeper_2. Then, when the local gatekeeper is asked to admit a call to destination address
2125551111, it knows to send the LRQ to gatekeeper_2.
When gatekeeper_2 receives the request, the gatekeeper must resolve the address so that the call can be
sent to its final destination. There may be an H.323 endpoint with that E.164 address that has registered
with gatekeeper_2, in which case gatekeeper_2 returns the IP address for that endpoint. However, it is
possible that the E.164 address belongs to a non-H.323 device (for example, a telephone or an H.320
terminal). Because non-H.323 devices do not register with gatekeepers, gatekeeper_2 cannot resolve the
address. The gatekeeper must be able to select a gateway that can be used to reach the non-H.323 device.
This is where the technology prefixes (or “gateway-type”) become useful.
Technology Prefixes
The network administrator selects technology prefixes (tech-prefixes) to denote different types or classes
of gateways. The gateways are then configured to register with their gatekeepers with these prefixes. For
example, voice gateways can register with tech-prefix 1#, H.320 gateways with tech-prefix 2#, and
voicemail gateways with tech-prefix 3#. More than one gateway can register with the same type prefix.
When this happens, the gatekeeper makes a random selection among gateways of the same type.
If the callers know the type of device that they are trying to reach, they can include the technology prefix
in the destination address to indicate the type of gateway to use to get to the destination. For example, if
a caller knows that address 2125551111 belongs to a regular telephone, the destination address of
1#2125551111 can be used, where 1# indicates that the address should be resolved by a voice gateway.
When the voice gateway receives the call for 1#2125551111, it strips off the technology prefix and
bridges the next leg of the call to the telephone at 2125551111.
Gatekeeper-to-Gatekeeper Redundancy and Load-Sharing Mechanism
The gatekeeper-to-gatekeeper redundancy and load-sharing mechanism expands the capability that is
provided by the redundant H.323 zone support feature. Redundant H.323 zone support, which was
introduced in Cisco IOS Release 12.1(1)T, allows you to configure multiple gatekeepers to service the
same zone or technology prefix by sending LRQs to two or more gatekeepers.
With the redundant H.323 zone support feature, the LRQs are sent simultaneously (in a “blast” fashion)
to all of the gatekeepers in the list. The gateway registers with the gatekeeper that responds first. Then,
if that gatekeeper becomes unavailable, the gateway registers with another gatekeeper from the list.
The gatekeeper-to-gatekeeper redundancy and load-sharing mechanism allows you to configure
gatekeeper support and to give preference to specific gatekeepers. You may choose whether the LRQs
are sent simultaneously or sequentially (one at a time) to the remote gatekeepers in the list. If the LRQs
are sent sequentially, a delay is inserted after the first LRQ and before the next LRQ is sent. This delay
allows the first gatekeeper to respond before the LRQ is sent to the next gatekeeper. The order in which
LRQs are sent to the gatekeepers is based on the order in which the gatekeepers are listed (using either
the zone prefix command or the gw-type-prefix command).
Once the local gatekeeper has sent LRQs to all the remote gatekeepers in the list (either simultaneously
or sequentially), if it has not yet received a location confirmation (LCF), it opens a “window.” During
this window, the local gatekeeper waits to see whether a LCF is subsequently received from any of the
remote gatekeepers. If no LCF is received from any of the remote gatekeepers while the window is open,
the call is rejected.
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Terminal Name Registration
Gatekeepers recognize one of two types of terminal aliases, or terminal names:
•
H.323 IDs, which are arbitrary, case-sensitive text strings
•
E.164 addresses, which are telephone numbers
If an H.323 network deploys interzone communication, each terminal should at least have a fully
qualified e-mail name as its H.323 identification (ID), for example, [email protected] The domain name
of the e-mail ID should be the same as the configured domain name for the gatekeeper of which it is to
be a member. As in the previous example, the domain name would be cisco.com.
Interzone Communication
To allow endpoints to communicate between zones, gatekeepers must be able to determine which zone
an endpoint is in and be able to locate the gatekeeper responsible for that zone. If the Domain Name
System (DNS) mechanism is available, a DNS domain name can be associated with each gatekeeper. See
the DNS configuration task in the “Configuring Intergatekeeper Communication” section to understand
how to configure DNS.
RADIUS and TACACS+
Version 1 of the H.323 specification does not provide a mechanism for authenticating registered
endpoints. Credential information is not passed between gateways and gatekeepers. However, by
enabling AAA on the gatekeeper and configuring for RADIUS and TACACS+, a rudimentary form of
identification can be achieved.
If the AAA feature is enabled, the gatekeeper attempts to use the registered aliases along with a password
and completes an authentication transaction to a RADIUS and TACACS+ server. The registration will
be accepted only if RADIUS and TACACS+ successfully authenticates the name.
The gatekeeper can be configured so that a default password can be used for all users. The gatekeeper
can also be configured so that it recognizes a password separator character that allows users to piggyback
their passwords onto H.323-ID registrations. In this case, the separator character separates the ID and
password fields.
Note
The names loaded into RADIUS and TACACS+ are probably not the same names provided for dial
access because they may all have the same password.
Accounting via RADIUS and TACACS+
If AAA is enabled on the gatekeeper, the gatekeeper will emit an accounting record each time a call is
admitted or disconnected.
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Interzone Routing Using E.164 Addresses
Interzone routing may be configured using E.164 addresses.
Two types of address destinations are used in H.323 calls. The destination can be specified using either
an H.323-ID address (a character string) or an E.164 address (a string that contains telephone keypad
characters). The way interzone calls are routed depends on the type of address being used.
When using H.323-ID addresses, interzone routing is handled through the use of domain names. For
example, to resolve the domain name [email protected], the source endpoint gatekeeper finds the
gatekeeper for cisco.com and sends it the location request for the target address [email protected] The
destination gatekeeper looks in its registration database, sees bob registered, and returns the appropriate
IP address to get to bob.
When using E.164 addresses, call routing is handled through zone prefixes and gateway-type prefixes,
also referred to as technology prefixes. The zone prefixes, which are typically area codes, serve the same
purpose as domain names in H.323-ID address routing. Unlike domain names, however, more than one
zone prefix can be assigned to one gatekeeper, but the same prefix cannot be shared by more than one
gatekeeper.
Use the zone prefix command to define gatekeeper responsibilities for area codes. The command can
also be used to tell the gatekeeper which prefixes are in its own zones and which remote gatekeepers are
responsible for other prefixes.
Note
Area codes are used as an example in this section, but a zone prefix need not be an area code. It can
be a country code, an area code plus local exchange (NPA-NXX), or any other logical hierarchical
partition.
The following sample command shows how to configure a gatekeeper with the knowledge that zone
prefix 212....... (that is, any address beginning with area code 212 and followed by seven arbitrary digits)
is handled by gatekeeper gk-ny:
my-gatekeeper(config-gk)# zone prefix gk-ny 212.......
When my-gatekeeper is asked to admit a call to destination address 2125551111, it knows to send the
location request to gk-ny.
However, once the query gets to gk-ny, gk-ny still needs to resolve the address so that the call can be
sent to its final destination. There could be an H.323 endpoint that has registered with gk-ny with that
E.164 address, in which case gk-ny would return the IP address for that endpoint. However, it is more
likely that the E.164 address belongs to a non-H.323 device, such as a telephone or an H.320 terminal.
Because non-H.323 devices do not register with gatekeepers, gk-ny has no knowledge of which device
the address belongs to or which type of device it is, so the gatekeeper cannot decide which gateway
should be used for the hop off to the non-H.323 device. (The term hop off refers to the point at which the
call leaves the H.323 network and is destined for a non-H.323 device.)
Note
The number of zone prefixes defined for a directory gatekeeper that is dedicated to forwarding LRQs,
and not for handling local registrations and calls, should not exceed 10,000; 4 MB of memory must
be dedicated to describing zones and zone prefixes to support this maximum number of zone prefixes.
The number of zone prefixes defined for a gatekeeper that handles local registrations and calls should
not exceed 2000.
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To enable the gatekeeper to select the appropriate hop-off gateway, use the gw-type-prefix command to
configure technology or gateway-type prefixes. Select technology prefixes to denote different types or
classes of gateways. The gateways are then configured to register with their gatekeepers using these
technology prefixes.
For example, voice gateways might register with technology prefix 1#, and H.320 gateways might
register with technology prefix 2#. If there are several gateways of the same type, configure them to
register with the same prefix type. By having them register with the same prefix type, the gatekeeper
treats the gateways as a pool out of which a random selection is made whenever a call for that prefix type
arrives. If a gateway can serve more than one type of hop-off technology, it can register more than one
prefix type with the gatekeeper.
Callers will need to know the technology prefixes that are defined. The callers will need to know the type
of device they are trying to reach and will need to prepend the appropriate technology prefix to the
destination address to indicate the type of gateway needed to reach the destination.
For example, callers might request 1#2125551111 if they know that address 2125551111 is for a
telephone and that the technology prefix for voice gateways is 1#. The voice gateway is configured with
a dial peer (using the dial-peer command) so that when the gateway receives the call for 1#2125551111,
it strips off the technology prefix 1# and bridges the next leg of the call to the telephone at 2125551111.
In cases in which the call scenario is as shown in Figure 57, voice-gw1 can be configured to prepend the
voice technology prefix 1# so that the use of technology prefixes is completely transparent to the caller.
PSTN
Telephone
Call Scenario
H.323 network
voice-gw1
PSTN
voice-gw2
Telephone
13098
Figure 57
Additionally, in using the gw-type-prefix command, a particular gateway-type prefix can be defined as
the default gateway type to be used for addresses that cannot be resolved. It also forces a technology
prefix to always hop off in a particular zone.
If the majority of calls hop off on a particular type of gateway, the gatekeeper can be configured to use
that type of gateway as the default type so that callers no longer have to prepend a technology prefix on
the address. For example, if voice gateways are mostly used in a network, and all voice gateways have
been configured to register with technology prefix 1#, the gatekeeper can be configured to use 1#
gateways as the default technology if the following command is entered:
my-gatekeeper(config-gk)# gw-type-prefix 1# default-technology
Now a caller no longer needs to prepend 1# to use a voice gateway. Any address that does not contain
an explicit technology prefix will be routed to one of the voice gateways that registered with 1#.
With this default technology definition, a caller could ask the gatekeeper for admission to 2125551111.
If the local gatekeeper does not recognize the zone prefix as belonging to any remote zone, it will route
the call to one of its local (1#) voice gateways so that the call hops off locally. However, if it knows that
gk-ny handles the 212 area code, it can send a location request for 2125551111 to gk-ny. This requires
that gk-ny also be configured with some default gateway type prefix and that its voice gateways be
registered with that prefix type.
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Note
For ease of maintenance, the same prefix type should be used to denote the same gateway type in all
zones under your administration. No more than 50 different technology prefixes should be registered
per zone.
Also, with the gw-type-prefix command, a hop off can be forced to a particular zone. When an endpoint
or gateway makes a call-admission request to its gatekeeper, the gatekeeper determines the destination
address by first looking for the technology prefix. When that is matched, the remaining string is
compared against known zone prefixes. If the address is determined to be a remote zone, the entire
address, including technology and zone prefixes, is sent to the remote gatekeeper in a location request.
That remote gatekeeper then uses the technology prefix to decide on which of its gateways to hop off. In
other words, the zone prefix (defined using the zone prefix command) determines the routing to a zone,
and once there, the technology prefix (defined using the gw-type-prefix command) determines the
gateway to be used in that zone. The zone prefix takes precedence over the technology prefix.
This behavior can be overridden by associating a forced hop-off zone with a particular technology prefix.
Associating a forced hop-off zone with a particular technology prefix forces the call to the specified
zone, regardless of what the zone prefix in the address is. As an example, you are in the 408 area code
and want callers to the 212 area code in New York to use H.323-over-IP and hop off there because it
saves on costs. However, the only H.320 gateway is in Denver. In this example, calls to H.320 endpoints
must be forced to hop off in Denver, even if the destination H.320 endpoint is in the 212 area code. The
forced hop-off zone can be either a local zone (that is, one that is managed by the local gatekeeper) or a
remote zone.
HSRP Support
Cisco routers support Hot Standby Router Protocol (HSRP), which allows one router to serve as a backup
to another router. Cisco gatekeepers can be configured to use HSRP so that when one gatekeeper fails,
the standby gatekeeper assumes its role.
To configure a gatekeeper to use HSRP, perform the following tasks:
•
Select one interface on each gatekeeper to serve as the HSRP interface and configure these two
interfaces so that they belong to the same HSRP group but have different priorities. The one with
the higher priority will be the active gatekeeper; the other assumes the standby role. Make a note of
the virtual HSRP IP address shared by both of these interfaces. (For details on HSRP and HSRP
configuration, refer to the Cisco IOS IP Configuration Guide.)
•
Configure the gatekeepers so that the HSRP virtual IP address is the RAS address for all local zones.
•
Make sure that the gatekeeper-mode configurations on both routers are identical.
•
If the endpoints and gateways are configured so that they use a specific gatekeeper address (rather
than multicasting), use the HSRP virtual IP address as the gatekeeper address. You can also let the
endpoints and gateways find the gatekeeper by multicasting. As long as it is on standby status, the
secondary gatekeeper neither receives nor responds to multicast or unicast requests.
As long as both gatekeepers are up, the one with the higher priority on its HSRP interface will be the
active gatekeeper. If this active gatekeeper fails, or if its HSRP interface fails, the standby HSRP
interface assumes the virtual HSRP address and, with it, the active gatekeeper role. When the gatekeeper
with the higher HSRP priority comes back online, it reclaims the HSRP virtual address and the
gatekeeper function, while the secondary gatekeeper goes back to standby status.
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Note
Gatekeeper failover will not be completely transparent to endpoints and gatekeepers. When the
standby gatekeeper takes over, it does not have the state of the failed gatekeeper. If an endpoint that
had registered with the failed gatekeeper now makes a request to the new gatekeeper, the gatekeeper
responds with a reject, indicating that it does not recognize the endpoint. The endpoint must
reregister with the new gatekeeper before it can continue H.323 operations.
For an example of configuring gatekeeper HSRP support, see the “H.323 Gatekeeper and Proxy
Configuration Examples” section.
H.323 Proxy Features
Each of the following sections describes how the proxy feature can be used in an H.323 network:
•
Security, page 297
•
Quality of Service, page 301
•
Application-Specific Routing, page 301
Security
When terminals signal each other directly, they must have direct access to each other’s addresses. This
exposes an attacker to key information about a network. When a proxy is used, the only addressing
information that is exposed to the network is the address of the proxy; all other terminal and gateway
addresses are hidden.
There are several ways to use a proxy with a firewall to enhance network security. The configuration to
be used depends on how capable the firewall is of handling the complex H.323 protocol suite. Each of
the following sections describes a common configuration for using a proxy with a firewall:
•
Proxy Inside the Firewall, page 298
•
Proxy in Co-Edge Mode, page 299
•
Proxy Outside the Firewall, page 300
•
Proxies and NAT, page 300
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Proxy Inside the Firewall
H.323 is a complex, dynamic protocol that consists of several interrelated subprotocols. During H.323
call setup, the ports and addresses released with this protocol require a detailed inspection as the setup
progresses. If the firewall does not support this dynamic access control based on the inspection, a proxy
can be used just inside the firewall. The proxy provides a simple access control scheme, as illustrated in
Figure 58.
Figure 58
Proxy Inside the Firewall
Terminals
Firewall
Edge router
Outside
devices
S6913
Gatekeeper
Proxy
Because the gatekeeper (using RAS) and the proxy (using call setup protocols) are the only endpoints
that communicate with other devices outside the firewall, it is simple to set up a tunnel through the
firewall to allow traffic destined for either of these two endpoints to pass through.
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Proxy in Co-Edge Mode
If H.323 terminals exist in an area with local interior addresses that must be translated to valid exterior
addresses, the firewall must be capable of decoding and translating all addresses passed in the various
H.323 protocols. If the firewall is not capable of this translation task, a proxy may be placed next to the
firewall in a co-edge mode. In this configuration, interfaces lead to both inside and outside networks.
(See Figure 59.)
Figure 59
Proxy in Co-Edge Mode
Edge router
Firewall
Outside
devices
Gatekeeper
Proxy
S6914
Terminals
In co-edge mode, the proxy can present a security risk. To avoid exposing a network to unsolicited
traffic, configure the proxy to route only proxied traffic. In other words, the proxy routes only H.323
protocol traffic that is terminated on the inside and then repeated to the outside. Traffic that moves in the
opposite direction can be configured this way as well.
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Proxy Outside the Firewall
To place the proxy and gatekeeper outside the firewall, two conditions must exist. First, the firewall must
support H.323 dynamic access control. Second, Network Address Translation (NAT) must not be in use.
If NAT is in use, each endpoint must register with the gatekeeper for the duration of the time it is online.
This will quickly overwhelm the firewall because a large number of relatively static, internal-to-external
address mappings will need to be maintained.
If the firewall does not support H.323 dynamic access control, the firewall can be configured with static
access lists that allow traffic from the proxy or gatekeeper through the firewall. This can present a
security risk if an attacker can spoof, or simulate, the IP addresses of the gatekeeper or proxy and use
them to attack the network. Figure 60 illustrates proxy outside the firewall.
Figure 60
Proxy Outside the Firewall
Edge router
Gatekeeper
Terminals
Firewall
Outside
devices
S6915
Proxy
Proxies and NAT
When a firewall is providing NAT between an internal and an external network, proxies may allow H.323
traffic to be handled properly, even in the absence of a firewall that can translate addresses for H.323
traffic. Table 24 and Table 25 provide guidelines for proxy deployment for networks that use NAT.
Table 24
Guidelines for Networks That Use NAT
For Networks Using NAT
Firewall with H.323 NAT
Firewall Without H.323 NAT
Firewall with dynamic access
control
Gatekeeper and proxy inside the
firewall
Co-edge gatekeeper and proxy
Firewall without dynamic access Gatekeeper and proxy inside the
control
firewall, with static access lists
on the firewall
Co-edge gatekeeper and proxy
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Table 25
Guidelines for Networks That Do Not Use NAT
For Networks Not Using NAT
Firewall with H.323. NAT
Firewall with Dynamic Access
Control
Gatekeeper and proxy inside the Gatekeeper and proxy inside the
firewall
firewall
Gatekeeper and proxy outside
the firewall
Firewall Without Dynamic
Access Control
Firewall Without H.323 NAT
Gatekeeper and proxy outside the
firewall
Gatekeeper and proxy inside the Gatekeeper and proxy inside the
firewall, with static access lists firewall, with static access lists
on the firewall
on the firewall
Quality of Service
Quality of service (QoS) enables complex networks to control and predictably service a variety of
applications. QoS expedites the handling of mission-critical applications while sharing network
resources with noncritical applications. QoS also ensures available bandwidth and minimum delays
required by time-sensitive multimedia and voice applications. In addition, QoS gives network managers
control over network applications, improves cost-efficiency of WAN connections, and enables advanced
differentiated services. QoS technologies are elemental building blocks for other Cisco IOS-enabling
services such as its H.323-compliant gatekeeper. Overall call quality can be improved dramatically in
the multimedia network by using pairs of proxies between regions of the network where QoS can be
requested.
When two H.323 terminals communicate directly, the resulting call quality can range from good (for
high-bandwidth intranets) to poor (for most calls over the public network). As a result, deployment of
H.323 is almost always predicated on the availability of some high-bandwidth, low-delay,
low-packet-loss network that is separate from the public network or that runs overlaid with the network
as a premium service and adequate QoS.
Adequate QoS usually requires terminals that are capable of signaling such premium services. There are
two major ways to achieve such signaling:
•
RSVP to reserve flows having adequate QoS based on the media codecs of H.323 traffic
•
IP precedence bits to signal that the H.323 traffic is special and that it deserves higher priority
Unfortunately, the vast majority of H.323 terminals cannot achieve signaling in either of these ways.
The proxy can be configured to use any combination of RSVP and IP precedence bits.
The proxy is not capable of modifying the QoS between the terminal and itself. To achieve the best
overall QoS, ensure that terminals are connected to the proxy using a network that intrinsically has good
QoS. In other words, configure a path between a terminal and proxy that provides good bandwidth, delay,
and packet-loss characteristics without the terminal needing to request special QoS. A high-bandwidth
LAN works well for this.
Application-Specific Routing
To achieve adequate QoS, a separate network may be deployed that is partitioned away from the standard
data network. The proxy can take advantage of such a partitioned network using a feature known as
application-specific routing (ASR).
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H.323 Prerequisite Tasks and Restrictions
Application-specific routing is simple. When the proxy receives outbound traffic, it directs traffic to an
interface that is connected directly to the QoS network. The proxy does not send the traffic using an
interface that is specified for the regular routing protocol. Similarly, inbound traffic from other proxies
is received on the interface that is connected to the QoS network. This is true if all these other proxies
around the QoS network use ASR in a consistent fashion. ASR then ensures that ordinary traffic is not
routed into the QoS network by mistake.
Implementation of ASR ensures the following:
Note
•
Each time a connection is established with another proxy, the proxy automatically installs a host
route pointing at the interface designated for ASR.
•
The proxy is configured to use a loopback interface address. The proxy address is visible to both the
ASR interface and all regular interfaces, but there are no routes established between the loopback
interface and the ASR interface. This ensures that no non-H.323 traffic is routed through the ASR
interface.
ASR is not supported on Frame Relay or ATM interfaces for the Cisco MC3810 platform.
H.323 Prerequisite Tasks and Restrictions
This section contains prerequisite tasks and restrictions for configuring H.323 gatekeepers and proxies.
Redundant H.323 Zone Support
Redundant H.323 zone support has the following restrictions and limitations:
•
The gateway can register with only one gatekeeper at any given time.
•
Only E.164 address resolution is supported.
•
Because the gateway can register with only one gatekeeper at a time, redundant H.323 zone support
provides only redundancy and does not provide any load balancing.
•
Although redundant H.323 zone support allows you to configure alternate gatekeepers, it will not
insert information in the alternate gatekeeper field of some RAS messages.
Gatekeeper-to-Gatekeeper Redundancy and Load-Sharing Mechanism
The gatekeeper-to-gatekeeper redundancy and load-sharing mechanism has the following restrictions
and limitations:
•
The gatekeeper-to-gatekeeper redundancy and load-sharing mechanism requires the Cisco H.323
VoIP Gatekeeper for Cisco Access Platforms feature.
•
The order in which LRQs are sent to the gatekeepers is based on the order in which the gatekeepers
are listed. You cannot specify a priority number for a gatekeeper.
•
Regardless of the order in which the LRQs are sent, the gateway will still use the first gatekeeper
that sends an LCF.
•
The settings for delay between LRQs and the LRQ window are global and cannot be set on a
per-zone or technology-prefix basis.
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H.323 Gatekeeper Configuration Task List
•
The number of remote gatekeepers multiplied by the delay per LRQ cannot exceed the Routing
Information Protocol (RIP) timeout. Therefore, we recommend that you limit your list of remote
gatekeepers to two or three.
•
If LRQ forwarding is enabled on the directory gatekeeper, the sequential setting for LRQs is
ignored.
•
Only E.164 address resolution is supported.
•
Using redundant H.323 zone support in the “directory gatekeeper” can generate extra RAS
messages. Therefore, the number of “directory gatekeeper” levels should be kept to a minimum (two
or three at the maximum).
H.323 Gatekeeper Configuration Task List
To configure Cisco gatekeepers, perform the tasks in the following sections. The tasks in these two
sections are required.
•
Configuring the Gatekeeper, page 303 (Required)
•
Configuring the Proxy, page 332 (Required)
Configuring the Gatekeeper
To configure gatekeepers, perform the tasks in the following sections. All of the tasks listed are required.
•
Starting a Gatekeeper, page 304
– Configuring Intergatekeeper Communication, page 307
•
Configuring Redundant H.323 Zone Support, page 308
•
Configuring Local and Remote Gatekeepers, page 309
•
Configuring Redundant Gatekeepers for a Zone Prefix, page 310
•
Configuring Redundant Gatekeepers for a Technology Prefix, page 311
•
Configuring Static Nodes, page 313
•
Configuring H.323 Users via RADIUS, page 314
•
Configuring a RADIUS/AAA Server, page 318
•
Configuring User Accounting Activity for RADIUS, page 320
•
Configuring E.164 Interzone Routing, page 321
•
Configuring H.323 Version 2 Features, page 322
– Configuring a Dialing Prefix for Each Gateway, page 323
– Configuring a Prefix to a Gatekeeper Zone List, page 326
– Configuring a Gatekeeper for Interaction with External Applications, page 325
– Configuring Gatekeeper Triggers for Interaction with External Applications, page 327
– Configuring Redundant H.323 Zone Support, page 308
– Configuring a Forced Disconnect on a Gatekeeper, page 332
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Starting a Gatekeeper
To enter gatekeeper configuration mode and to start the gatekeeper, use the following commands
beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# gatekeeper
Enters gatekeeper configuration mode.
Step 2
Router(config-gk)# zone local gatekeeper-name
domain-name [ras-IP-address]
Specifies a zone controlled by a gatekeeper.
The arguments are as follows:
•
gatekeeper-name—Specifies the gatekeeper
name or zone name. This is usually the fully
domain-qualified host name of the gatekeeper.
For example, if the domain name is cisco.com,
the gatekeeper name might be gk1.cisco.com.
However, if the gatekeeper is controlling
multiple zones, the gatekeeper name for each
zone should be some unique string that has a
mnemonic value.
•
domain-name—Specifies the domain name
served by this gatekeeper.
•
ras-IP-address—(Optional) Specifies the IP
address of one of the interfaces on the
gatekeeper. When the gatekeeper responds to
gatekeeper discovery messages, it signals the
endpoint or gateway to use this address in future
communications.
Note
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Setting this address for one local zone makes
it the address used for all local zones.
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Step 3
Command
Purpose
Router(config-gk)# zone prefix gatekeeper-name
e164-prefix [blast | seq] [gw-priority priority
gw-alias [gw-alias, ...]]
Adds a prefix to the gatekeeper zone list.
The keywords and arguments are as follows:
•
gatekeeper-name—Specifies the name of a local
or remote gatekeeper, which must have been
defined by using the zone local or zone remote
command.
•
e164-prefix—Specifies an E.164 prefix in
standard form followed by dots (.). Each dot
represents a number in the E.164 address. For
example, 212....... is matched by 212 and any 7
numbers.
Note
Although a dot to represent each digit in an
E.164 address is the preferred configuration
method, you can also enter an asterisk (*) to
match any number of digits.
•
blast—(Optional) If you list multiple hopoffs,
indicates that the location requests (LRQs)
should be sent simultaneously to the gatekeepers
based on the order in which they were listed. The
default is seq.
•
seq—(Optional) If you list multiple hopoffs,
indicates that the LRQs should be sent
sequentially to the gatekeepers based on the
order in which they were listed. The default is
seq.
•
gw-priority priority gw-alias—(Optional) Use
the gw-priority option to define how the
gatekeeper selects gateways in its local zone for
calls to numbers that begin with prefix
e164-prefix. Do not use this option to set priority
levels for a prefix assigned to a remote
gatekeeper.
Use values from 0 to 10. A 0 value prevents the
gatekeeper from using the gateway gw-alias for
that prefix. Value 10 places the highest priority
on gateway gw-alias. If you do not specify a
priority value for a gateway, the value 5 is
assigned.
To assign the same priority value for one prefix
to multiple gateways, list all the gateway names
after the pri-0-to-10 value.
The gw-alias name is the H.323 ID of a gateway
that is registered or will register with the
gatekeeper. This name is set on the gateway with
the h323-gateway voip h.323-id command.
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Step 4
Command
Purpose
Router(config-gk)# zone subnet local-gatekeeper-name
[default | subnet-address {/bits-in-mask |
mask-address} enable]
Defines a set of subnets that constitute the gatekeeper
zone. Enables the gatekeeper for each of these
subnets and disables it for all other subnets. (Repeat
for all subnets.)
The keywords and arguments are as follows:
•
local-gatekeeper-name—Specifies the name of
the local gatekeeper.
•
default—(Optional) Applies to all other subnets
that are not specifically defined by the
zone subnet command.
•
subnet-address—(Optional) Specifies the
address of the subnet that is being defined.
•
bits-in-mask—(Optional) Specifies the number
of bits of the mask to be applied to the subnet
address.
Note
Step 5
Router(config-gk)# no shutdown
The slash must be entered before this
argument.
•
mask-address—(Optional) Specifies the mask (in
dotted string format) to be applied to the subnet
address.
•
enable—(Optional) Specifies that the gatekeeper
accepts discovery and registration from the
specified subnets.
Note
To define the zone as being all but one set of
subnets by disabling that set and enabling all
other subnets, use the no form of the
command as follows: Configure no zone
subnet local-gatekeeper-name
subnet-address {/bits-in-mask |
mask-address} enable.
Note
To accept the default behavior, which is that
all subnets are enabled, use the no form of the
command as follows: no zone subnet
local-gatekeeper-name default enable.
Brings the gatekeeper online.
The local-gatekeeper-name argument should be a Domain Name System (DNS) host name if DNS is to
be used to locate remote zones.
The zone subnet command may be used more than once to create a list of subnets controlled by a
gatekeeper. The subnet masks need not match actual subnets in use at your site. For example, to specify
a particular endpoint, show its address as a 32-bit netmask.
If a local gatekeeper name is contained in the message, it must match the local-gatekeeper-name
argument.
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Note
To explicitly enable or disable a particular endpoint, specify its host address using a 32-bit subnet
mask.
Configuring Intergatekeeper Communication
This section describes two ways to configure intergatekeeper communication:
•
Via DNS, page 307
•
Manual Configuration, page 308
Via DNS
To configure intergatekeeper communication using DNS, use the following commands in global
configuration mode:
Step 1
Command
Purpose
Router(config)# ip name-server dns-server-name
[server-address2...server-address6]
Specifies the DNS server address.
The arguments are as follows:
Step 2
Router(config)# ip domain-name name
•
dns-server-name— Specifies the IP address of
the name server.
•
server-address2...server-address6—(Optional)
IP addresses of additional name servers (a
maximum of six name servers).
Defines a default domain name that the Cisco IOS
software uses to complete unqualified host names
(names without a dotted-decimal domain name). The
name argument specifies the default domain name
used to complete unqualified host names. Do not
include the initial period that separates an unqualified
name from the domain name.
For all gatekeepers in the system, enter a text record of the form into DNS:
ras [[email protected]] host [:port] [priority]
The gk-id argument is an optional gatekeeper ID. If the optional gatekeeper ID is not specified, host is
used as the gatekeeper ID.
The host argument is either an IP address or the actual host name of the gatekeeper in the form
host.some_domain.com.
The port argument, if specified, should be some port number other than RAS port 1719.
The priority argument specifies the order in which the listed gatekeepers should be searched for
endpoints. Gatekeepers with lower priorities are searched before those with higher numbers.
How you enter the text record for a particular domain depends on the DNS implementation. The
following examples are for the Berkeley Internet Name Domain (BIND). These records are typically
entered into the “hosts” database:
zone1.comintxt“ras gk.zone1.com”
zone2.comintxt“ras [email protected]”
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zone3.comintxt“ras [email protected]:1725”
zone4.comintxt“ras [email protected]:1725 123”
zone5.comintxt“ras [email protected]:1725”
Manual Configuration
If you choose not to use DNS or if DNS is not available, configure intergatekeeper communication
manually. To configure intergatekeeper manual communication, use the following command in
gatekeeper configuration mode for every other gatekeeper in the network:
Command
Purpose
Router(config-gk)# zone remote other-gatekeeper-name
other-domain-name other-gatekeeper-ip-address
[port-number]
Statically specifies a remote zone if Domain Name System
(DNS) is unavailable or undesirable. Enter this command
for each gatekeeper.
The arguments are as follows:
•
other-gatekeeper-name—Specifies the name of the
remote gatekeeper.
•
other-domain-name—Specifies the domain name of
the remote gatekeeper.
•
other-gatekeeper-ip-address—Specifies the IP
address of the remote gatekeeper.
•
port-number—(Optional) Specifies the RAS signaling
port number for the remote zone. Value ranges are
from 1 to 65,535. If this option is not set, the default is
the well-known RAS port number 1719.
Configuring Redundant H.323 Zone Support
Regardless of whether you specify sequential or blast, there is an order to how the LRQs are sent. With
sequential, the LRQs are sent one at a time with a delay between each. With blast, the LRQs are sent
back-to-back in a rapid sequence without any delay between them. The order in which zone and
technology prefixes are configured determines the order in which the LRQs are sent to the remote
gatekeepers. Using zone prefixes as an example, the local gatekeeper routes the call to the first zone that
responds with an LCF. If the local gatekeeper is configured for a zone prefix that already has remote
gatekeepers configured, the local gatekeeper will automatically put that zone prefix at the top of the list.
For example:
gatekeeper
zone local gnet-2503-2-gk cisco.com
zone remote gnet-2600-1-gk cisco.com 172.18.194.131 1719
zone remote gnet-2503-3-gk cisco.com 172.18.194.134 1719
zone prefix gnet-2600-1-gk 919.......
zone prefix gnet-2503-6-gk 919.......
With this configuration, LRQs are first sent to gnet-2600-1-gk (which is the first zone prefix because it
has a remote gatekeeper configured for it) and then to gnet-2503-6-gk (which is the second zone prefix).
If you add the local gatekeeper to that zone prefix, it automatically goes to the top of the list, as shown
below:
gatekeeper
zone local gnet-2503-2-gk cisco.com
zone remote gnet-2600-1-gk cisco.com 172.18.194.131 1719
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zone
zone
zone
zone
remote
prefix
prefix
prefix
gnet-2503-3-gk
gnet-2503-2-gk
gnet-2600-1-gk
gnet-2503-6-gk
cisco.com 172.18.194.134 1719
919.......
919.......
919.......
As you can see, the zone prefix for the local gatekeeper (gnet-2503-2-gk) has been inserted at the top of
the zone prefix list. If the local gatekeeper can resolve the address, it will not send LRQs to the remote
zones.
If you are configuring technology prefixes, the zone prefix for the local gatekeeper should be inserted at
the top of the zone prefix list. If the local gatekeeper can resolve the address, it will not send LRQs to
the remote zones.
Configuring Local and Remote Gatekeepers
To configure local and remote gatekeepers, use the following commands beginning in global
configuration mode:
Command
Purpose
Step 1 Router(config)# gatekeeper
Enters gatekeeper configuration mode.
Step 2 Router(config-gk)# zone local gatekeeper-name
Specifies a zone controlled by a gatekeeper.
domain-name [ras-IP-address]
The arguments are as follows:
•
gatekeeper-name—Specifies the gatekeeper name or
zone name. This is usually the fully
domain-qualified host name of the gatekeeper. For
example, if the domain name is cisco.com, the
gatekeeper name might be gk1.cisco.com. However,
if the gatekeeper is controlling multiple zones, the
gatekeeper name for each zone should be some
unique string that has a mnemonic value.
•
domain-name—Specifies the domain name served
by this gatekeeper.
•
ras-IP-address—(Optional) Specifies the IP address
of one of the interfaces on the gatekeeper. When the
gatekeeper responds to gatekeeper discovery
messages, it signals the endpoint or gateway to use
this address in future communications.
Note
Setting this address for one local zone makes it
the address used for all local zones.
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Command
Purpose
Step 3 Router(config-gk)# zone remote
other-gatekeeper-name other-domain-name
other-gatekeeper-ip-address
[port-number]
Configures the remote gatekeeper.
The arguments are as follows:
•
other-gatekeeper-name—Name of the remote
gatekeeper.
•
other-domain-name—Domain name of the remote
gatekeeper.
•
other-gatekeeper-ip-address—IP address of the
remote gatekeeper.
•
port-number—(Optional) RAS signaling port
number for the remote zone. Value ranges from 1 to
65,535. If this option is not set, the default is the
well-known RAS port number 1719.
Configuring Redundant Gatekeepers for a Zone Prefix
To configure redundant gatekeepers for a zone prefix, use the following commands beginning in global
configuration mode:
Command
Purpose
Step 1
Router(config)# gatekeeper
Enters gatekeeper configuration mode.
Step 2
Router(config-gk)# zone prefix gatekeeper-name
e164-prefix [blast | seq] [gw-priority priority
gw-alias [gw-alias, ...]]
Adds a prefix to the gatekeeper zone list.
For an explanation of the keywords and arguments,
see Step 3 of the configuration task table in the
“Starting a Gatekeeper” section on page 304.
You can configure multiple remote gatekeepers for the same prefix, but only one of the gatekeepers
defined for any given zone prefix can be local. It is recommended that you limit the number of remote
gatekeepers that service the same zone prefix to two.
By default, LRQs are sent sequentially to the remote gatekeepers. If you would like the LRQs to be sent
simultaneously (blast), you need only specify the blast keyword on one zone prefix command per E.164
prefix.
Verifying Zone Prefix Redundancy
To verify the order in which LRQs will be sent to the gatekeepers defined for a zone prefix, enter the
show gatekeeper zone prefix command. The following output lists all the gatekeepers, in order, and the
zone prefixes serviced by each.
router# show gatekeeper zone prefix
ZONE PREFIX TABLE
=================
GK-NAME
E164-PREFIX
----------------c3620-1-gk
917300....
c2514-2-gk
917300....
c2600-1-gk
919.......
c2514-1-gk
919.......
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To verify whether the LRQs will be sent sequentially or simultaneously to the gatekeepers, enter the
show running-config command. If the LRQs will be sent simultaneously, blast will appear beside the
first entry for a particular zone (as shown in the following output for zone 919).
Router# show running-config
Building configuration...
Current configuration:
!
gatekeeper
zone remote c3620-1-gk cisco.com 172.18.194.79 1719
zone remote c2514-2-gk cisco.com 172.18.194.89 1719
zone remote gk-cisco-paul cisco.com 172.18.193.155 1719
zone prefix c3620-1-gk 917300....
zone prefix c2514-2-gk 917300....
zone prefix c2514-2-gk 919....... blast
zone prefix c3620-1-gk 919.......
Configuring Redundant Gatekeepers for a Technology Prefix
To configure redundant gatekeepers for a technology prefix, use the following commands beginning in
global configuration mode:
Command
Purpose
Step 1
Router(config)# gatekeeper
Enters gatekeeper configuration mode.
Step 2
Router(config-gk)# gw-type-prefix type-prefix
[[hopoff gkid1] [hopoff gkid2] [hopoff gkidn] [seq |
blast]] [default-technology] [[gw ipaddr ipaddr
[port]]...]
Configures the gatekeepers to service a technology
zone and specifies whether LRQs should be sent in
blast or sequential fashion. The default is
sequential.
The keywords and arguments are as follows:
•
type-prefix—Specifies that a technology prefix
is recognized and stripped before checking for
the zone prefix. It is strongly recommended
that you select technology prefixes that do not
lead to ambiguity with zone prefixes. Do this
by using the # character to terminate
technology prefixes, for example, 3#.
•
hopoff gkid—(Optional) Specifies the
gatekeeper where the call is to hop off,
regardless of the zone prefix in the destination
address. The gkid argument refers to a
gatekeeper previously configured using the
zone local or zone remote command. You can
enter this keyword and argument multiple
times to configure redundant gatekeepers for a
given technology prefix.
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Command
Purpose
•
seq | blast—(Optional) If multiple hopoffs are
listed, indicates that the location requests
(LRQs) should be sent sequentially or
simultaneously (blast) to the gatekeepers
based on the order in which they were listed.
The default is to send them sequentially.
•
default-technology—(Optional) Specifies
that gateways that register with this prefix
option are used as the default for routing any
addresses that are otherwise unresolved.
•
gw ipaddr ipaddr [port]—(Optional)
Indicates that the gateway is incapable of
registering technology prefixes. When it
registers, it adds the gateway to the group for
this type-prefix, just as if it had sent the
technology prefix in its registration. This
parameter can be repeated to associate more
than one gateway with a technology prefix.
You can enter the hopoff keyword and gkid argument multiple times in the same command to define a
group of gatekeepers that will service a given technology prefix. After you have listed all of the
gatekeepers that will service that technology zone, you can specify whether the LRQs should be sent in
blast or sequential fashion.
Note
Only one of the gatekeepers in the hopoff list can be local. We recommend that you limit the number
of remote gatekeepers that service the same technology prefix to two.
Verifying Technology Prefix Redundancy
To verify that multiple gatekeepers are defined for a technology prefix, enter the show gatekeeper
gw-type-prefix command. The following output displays the gateway technology prefix table.
router# show gatekeeper gw-type-prefix
(GATEWAYS-TYPE PREFIX TABLE
================================
Prefix:3#*
(Hopoff zone c2600-1-gk c2514-1-gk)
To verify whether the LRQs will be sent sequentially or simultaneously to the gatekeepers, enter the
show running-config command. If the LRQs will be sent simultaneously, blast will appear at the end of
the gw-type-prefix line (as shown below).
Router# show running-config
Building configuration...
Current configuration:
!
gatekeeper
zone remote c2600-1-gk cisco.com 172.18.194.70 1719
zone remote c2514-1-gk cisco.com 172.18.194.71 1719
gw-type-prefix 3#* hopoff c2600-1-gk hopoff c2514-1-gk blast
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Configuring Static Nodes
In some cases, the registration information is not accessible for a terminal or endpoint from any
gatekeeper. This inaccessible registration information may be because the endpoint does not use RAS,
is in an area where no gatekeeper exists, or is in a zone where the gatekeeper addressing is unavailable
either through DNS or through configuration.
These endpoints can still be accessed via a gatekeeper by entering them as static nodes. To enter the
endpoints as static nodes, obtain the address of the endpoint and then use the following commands
beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# gatekeeper
Enters gatekeeper configuration mode.
Step 2
Router(config-gk)# zone local gatekeeper-name
domain-name [ras-IP-address]
Specifies a zone controlled by a gatekeeper.
For an explanation of the arguments, see Step 2 of the
configuration task table in the “Starting a
Gatekeeper” section on page 304.
Step 3
Router(config-gk)# alias static ip-signalling-addr
[port] gkid gatekeeper-name [ras ip-ras-addr port]
[terminal | mcu | gateway {h320 |h323-proxy | voip}]
[e164 e164-address] [h323id h323-id]
Creates a static entry in the local alias table for each
E.164 address. Repeat this step for each E.164
address you want to add for the endpoint.
The keywords and arguments are as follows:
•
ip-signalling-addr—Specifies the IP address of
the H.323 node, used as the address to signal
when establishing a call.
•
port—(Optional) Specifies the port number other
than the endpoint call-signaling well-known port
number (1720).
•
gkid gatekeeper-name—Specifies the name of
the local gatekeeper of whose zone this node is a
member.
•
ras ip-ras-addr—(Optional) Specifies the node
remote access server (RAS) signaling address. If
omitted, the ip-signalling-addr parameter is used
in conjunction with the RAS well-known port.
•
port—(Optional) Specifies a port number other
than the RAS well-known port number (1719).
•
terminal—(Optional) Indicates that the alias
refers to a terminal.
•
mcu—(Optional) Indicates that the alias refers to
a multiple control unit (MCU).
•
gateway—(Optional) Indicates that the alias
refers to a gateway.
•
h320—(Optional) Indicates that the alias refers
to an H.320 node.h320—(Optional) Indicates
that the alias refers to an H.320 node.
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Command
Purpose
•
h-323 proxy—(Optional) Indicates that the alias
refers to an H.323 proxy.
•
voip—(Optional) Indicates that the alias refers to
VoIP.
•
e164 e164-address—(Optional) Specifies the
node E.164 address. This keyword and argument
can be used more than once to specify as many
E.164 addresses as needed. Note that there is a
maximum number of 128 characters that can be
entered for this address. To avoid exceeding this
limit, you can enter multiple alias static
commands with the same call-signaling address
and different aliases.
•
h323-id h323-id—(Optional) Specifies the node
H.323 alias. This keyword and argument can be
used more than once to specify as many H.323
identification (ID) aliases as needed. Note that
there is a maximum number of 256 characters
that can be entered for this address. To avoid
exceeding this limit, you can enter multiple
commands with the same call signaling address
and different aliases.
Configuring H.323 Users via RADIUS
To authenticate H.323 users via RADIUS, use the following commands beginning in global
configuration mode:
Command
Purpose
Step 1
Router(config)# aaa new-model
Enables the authentication, authorization, and
accounting (AAA) access model.
Step 2
Router(config)# aaa authentication login {default |
list-name} method1 [method2...]
Sets AAA authentication at login.
The keywords and arguments are as follows:
•
default—Uses the listed authentication methods
that follow this keyword as the default list of
methods when a user logs in.
•
list-name—Specifies the character string used to
name the list of authentication methods activated
when a user logs in.
•
method1 [method2...]—Specifies that at least one
of the keywords described below be used:
– enable—Uses the enable password for
authentication.
– krb5—Uses Kerberos 5 for authentication..
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Command
Purpose
– krb5-telnet—Uses the Kerberos 5 Telnet
authentication protocol when using Telnet to
connect to the router.
– line—Uses the line password for
authentication.
– local—Uses the local username database for
authentication
– local-case—Uses case-sensitive local
username authentication.
– none—Uses no authentication.
– group radius—Uses the list of all RADIUS
servers for authentication.
– group tacacs+—Uses the list of all
TACACS+ servers for authentication.
– group group-name—Uses a subset of
RADIUS or TACACS+ servers for
authentication as defined by the group
server radius or aaa group server tacacs+
command.
Step 3
Router(config)# radius-server host {hostname |
ip-address} [auth-port port-number] [acct-port
port-number] [timeout seconds] [retransmit retries]
[key string]
Specifies the RADIUS server host.
The keywords and arguments are as follows:
•
hostname—Specifies the Domain Name System
(DNS) name of the RADIUS server host.
•
ip-address—Specifies the IP address of the
RADIUS server host.
•
auth-port—(Optional) Specifies the User
Datagram Protocol (UDP) destination port for
authentication requests.
•
port-number—(Optional) Specifies the port
number for authentication requests; the host is
not used for authentication if set to 0. If
unspecified, the port number defaults to 1645.
•
acct-port—(Optional) Specifies the UDP
destination port for accounting requests.
•
port-number—(Optional) Specifies the port
number for accounting requests; the host is not
used for accounting if set to 0. If unspecified, the
port number defaults to 1646.
•
acct-port—(Optional) Specifies the UDP
destination port for accounting requests.
•
port-number—(Optional) Specifies the port
number for accounting requests; the host is not
used for accounting if set to 0. If unspecified, the
port number defaults to 1646.
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Command
Purpose
•
timeout—(Optional) Specifies the time interval
(in seconds) for which the router waits for the
RADIUS server to reply before retransmitting.
This setting overrides the global value of the
radius-server timeout command. If no timeout
value is specified, the global value is used. Enter
a value in the range of from 1 to 1000.
•
seconds—(Optional) Specifies the timeout value.
Enter a value in the range of from 1 to 1000. If no
timeout value is specified, the global value is
used.
•
retransmit—(Optional) Specifies the number of
times a RADIUS request is resent to a server if
that server is not responding or responding
slowly. This setting overrides the global setting
of the radius-server retransmit command.
•
retries—(Optional) Specifies the retransmit
value. Enter a value in the range of from 1 to 100.
If no retransmit value is specified, the global
value is used.
•
key—(Optional) Specifies the authentication and
encryption key used between the router and the
RADIUS daemon running on this RADIUS
server. This key overrides the global setting of
the radius-server key command. If no key string
is specified, the global value is used.
The key is a text string that must match the
encryption key used on the RADIUS server.
Always configure the key as the last item in the
radius-server host command syntax. This is
because the leading spaces are ignored, but
spaces within and at the end of the key are used.
If you use spaces in the key, do not enclose the
key in quotation marks unless the quotation
marks themselves are part of the key.
•
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string—(Optional) Specifies the authentication
and encryption key for all RADIUS
communications between the router and the
RADIUS server. This key must match the
encryption used on the RADIUS daemon. All
leading spaces are ignored, but spaces within and
at the end of the key are used. If you use spaces
in your key, do not enclose the key in quotation
marks unless the quotation marks themselves are
part of the key.
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Step 4
Command
Purpose
Router(config)# radius-server key {0 string | 7
string | string}
Sets the authentication and encryption key for all
RADIUS communications between the router and the
RADIUS daemon.
The arguments are as follows:
•
0—Specifies that an unencrypted key will follow.
•
string—Specifies the unencrypted (cleartext)
shared key.
•
7—Specifies that a hidden key will follow.
•
string—Specifies the hidden shared key.
•
string—Specifies the unencrypted (cleartext)
shared key.
Step 5
Router(config)# gatekeeper
Enters gatekeeper configuration mode.
Step 6
Router(config-gk)# security {any | h323-id | e164}
{password default password | password separator
character}
Enables authentication and authorization on a
gatekeeper.
The keywords and arguments are as follows:
•
any—Uses the first alias of an incoming
Registration, Admission, and Status (RAS)
registration, regardless of its type, as the means
of identifying the user to RADIUS/TACACS+.
•
h323-id—Uses the first H.323 ID type alias as
the means of identifying the user to
RADIUS/TACACS+.
•
e164—Uses the first E.164 address type alias as
the means of identifying the user to
RADIUS/TACACS+.
•
password default password—Specifies the
default password that the gatekeeper associates
with endpoints when authenticating them with an
authentication server. The password must be
identical to the password on the authentication
server.
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Command
Purpose
•
password separator character—Specifies the
character that endpoints use to separate the
H.323-ID from the piggybacked password in the
registration. This allows each endpoint to supply
a user-specific password. The separator character
and password will be stripped from the string
before it is treated as an H.323-ID alias to be
registered.
Note that passwords may be piggybacked only in
the H.323-ID, not the E.164 address. This is
because the E.164 address allows a limited set of
mostly numeric characters. If the endpoint does
not wish to register an H.323-ID, it can still
supply an H.323-ID that consists of just the
separator character and password. This will be
understood to be a password mechanism, and no
H.323-ID will be registered.
After the previous steps have been completed, enter each user into the RADIUS database using either
the default password if using the security password default command or the actual passwords if using
the piggybacked password mechanism as the RADIUS authentication for that user. Enter either the user
H.323-ID or the E.164 address, depending on how the gatekeeper was configured.
For more information about configuring AAA services or RADIUS, refer to the Cisco IOS Security
Configuration Guide.
Configuring a RADIUS/AAA Server
To configure the RADIUS/AAA server with information about the gatekeeper for your network
installation, use the following commands beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# aaa new-model
Enables the authentication, authorization, and
accounting (AAA) model.
Step 2
Router(config)# aaa authentication login {default |
list-name} method1 [method2...]
Sets AAA authorization at login.
For an explanation of the keywords and arguments,
see Step 2 in the configuration task table in the
“Configuring H.323 Users via RADIUS” section on
page 314.
Step 3
Router(config)# radius-server deadtime minutes
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Improves the server response time when some servers
might be unavailable. The minutes argument
specifies the length of time, in minutes, for which a
RADIUS server is skipped over by transaction
requests, up to a maximum of 1440 minutes (24
hours).
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Step 4
Step 5
Command
Purpose
Router(config)# radius-server host {host-name |
ip-address} [auth-port port-number] [acct-port
port-number] [timeout seconds] [retransmit retries]
[key string]
Specifies the RADIUS server host.
Router(config)# radius-server key {0 string | 7
string | string}
Sets the authentication and encryption key for all
RADIUS communications between the router and the
RADIUS daemon.
For an explanation of the keywords and arguments,
see Step 3 in the configuration task table in the
“Configuring H.323 Users via RADIUS” section on
page 314.
For an explanation of the arguments, see Step 4 in the
configuration task table in the “Configuring H.323
Users via RADIUS” section on page 314.
In addition to the above configuration, make sure that the following information is configured in your
CiscoSecure AAA server:
•
In the /etc/raddb/clients file, ensure that the following information is provided.
#Client Name
#----------gk215.cisco.com
Key
------------------testing123
Where:
gk215.cisco.com is resolved to the IP address of the gatekeeper requesting authentication.
•
In the /etc/raddb/users file, ensure that the following information is provided:
[email protected] Password = "thiswouldbethepassword"
User-Service-Type = Framed-User,
Login-Service = Telnet
Where:
[email protected] is the h323-id of the gateway authenticating to gatekeeper gk215.cisco.com.
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Configuring User Accounting Activity for RADIUS
After AAA has been enabled and the gateway has been configured to recognize RADIUS as the remote
security server providing authentication services, the next step is to configure the gateway to report user
activity to the RADIUS server in the form of connection accounting records. To send connection
accounting records to the RADIUS server, use the following commands beginning in global
configuration mode:
Step 1
Command
Purpose
Router(config)# aaa accounting connection h323
{stop-only | start-stop | wait-start | none}
[broadcast] group group-name
Defines the accounting method list H.323 with
RADIUS as a method.
The keywords and arguments are as follows:
•
stop-only—Sends a “stop” accounting notice at
the end of the requested user process.
•
start-stop—Sends a “start” accounting notice at
the beginning of a process and a “stop”
accounting notice at the end of a process. The
“start” accounting record is sent in the
background. The requested user process begins
regardless of whether the “start” accounting
notice was received by the accounting server.
•
wait-start—Sends a “start” accounting notice at
the beginning of a process and a “stop”
accounting notice at the end of a process. The
“start” accounting record is sent in the
background. The requested user process does not
begin until the “start” accounting notice is
received by the server.
•
none—Disables accounting services on this line
or interface.
•
broadcast—(Optional) Enables sending
accounting records to multiple AAA servers.
Simultaneously sends accounting records to the
first server in each group. If the first server is
unavailable, failover occurs using the backup
servers defined within that group.
•
group group-name—Specifies the server group
to be used for accounting services. The following
are valid server group names:
– string—Specifies the character string used to
name a server group.
– radius—Uses list of all RADIUS hosts.
– tacacs+—Uses list of all TACACS+ hosts.
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Command
Purpose
Step 2
Router(config)# gatekeeper
Enters gatekeeper configuration mode.
Step 3
Router(config-gk)# aaa accounting
Enables authentication, authorization, and
accounting (AAA) of requested services for billing or
security purposes when you use RADIUS or
TACACS+.
For more information about AAA connection accounting services, refer to the Cisco IOS Security
Configuration Guide.
Configuring E.164 Interzone Routing
With Cisco IOS Release 12.0(3)T and later releases, interzone routing may be configured using E.164
addresses. To configure interzone routing in the E.164 address space, use the following commands
beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# gatekeeper
Enters gatekeeper configuration mode.
Step 2
Router(config-gk)# zone local gatekeeper-name
domain-name [ras-IP-address]
Specifies a zone controlled by a gatekeeper.
For an explanation of the arguments, see Step 2 of the
configuration task table in the “Starting a
Gatekeeper” section on page 304.
Step 3
Router(config-gk)# zone remote other-gatekeeper-name
other-domain-name other-gatekeeper-ip-address
[port-number]
Statically specifies a remote zone if Domain Name
System (DNS) is unavailable or undesirable. Enter
this command for each gatekeeper.
The arguments are as follows:
•
other-gatekeeper-name—Specifies the name of
the remote gatekeeper.
•
other-domain-name—Specifies the domain name
of the remote gatekeeper.
•
other-gatekeeper-ip-address—Specifies the IP
address of the remote gatekeeper.
•
port-number—(Optional) Specifies the
Registration, Admission, and Status (RAS)
signaling port number for the remote zone. Value
ranges are from 1 to 65,535. If this option is not
set, the default is the well-known RAS port
number 1719.
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Step 4
Step 5
Command
Purpose
Router(config-gk)# zone prefix gatekeeper-name
e164-prefix [blast | seq] [gw-priority priority
gw-alias [gw-alias, ...]]
Adds a prefix to the gatekeeper zone list.
Router(config-gk)# gw-type-prefix type-prefix
[[hopoff gkid1] [hopoff gkid2] [hopoff gkidn] [seq |
blast]] [default-technology] [[gw ipaddr ipaddr
[port]]...]
Configures the gatekeepers to service a technology
zone and specifies whether location requests (LRQs)
should be sent in blast or sequential fashion. The
default is sequential.
For an explanation of the keywords and arguments,
see Step 3 of the configuration task table in the
“Starting a Gatekeeper” section on page 304.
For an explanation of the keywords and arguments,
see Step 2 of the configuration task table in the
“Configuring Redundant Gatekeepers for a
Technology Prefix” section on page 311.
Configuring H.323 Version 2 Features
To configure H.323 Version 2 features using the Cisco gatekeeper, perform the following configuration
tasks. The first two tasks are required; the others are optional. Make sure that you include a priority value
for selecting between multiple gateways when you configure the gatekeeper.
•
Configuring a Dialing Prefix for Each Gateway, page 323 (Required)
•
Configuring a Gatekeeper for Interaction with External Applications, page 325 (Required)
•
Configuring a Prefix to a Gatekeeper Zone List, page 326 (Optional)
•
Configuring Gatekeeper Triggers for Interaction with External Applications, page 327 (Optional)
•
Configuring Inbound or Outbound Gatekeeper Proxied Access, page 330 (Optional)
•
Configuring a Forced Disconnect on a Gatekeeper, page 332 (Optional)
See the “H.323 Applications” chapter for further information on H.323 Version 2 features supported by
Cisco IOS software.
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Configuring a Dialing Prefix for Each Gateway
To configure a dialing prefix for each gateway, use the following commands beginning in global
configuration mode:
Command
Purpose
Step 1
Router(config)# gatekeeper
Enters gatekeeper configuration mode.
Step 2
Router(config-gk)# zone local
gatekeeper-name domain-name
[ras-IP-address]
Specifies a zone controlled by a gatekeeper.
The arguments are as follows:
•
gatekeeper-name—Specifies the gatekeeper name or zone name. This is
usually the fully domain-qualified host name of the gatekeeper. For
example, if the domain-name is cisco.com, the gatekeeper-name might be
gk1.cisco.com. However, if the gatekeeper is controlling multiple zones,
the gatekeeper-name for each zone should be some unique string that has
a mnemonic value.
•
domain-name—Specifies the domain name served by this gatekeeper.
•
ras-IP-address—(Optional) The IP address of one of the interfaces on the
gatekeeper. When the gatekeeper responds to gatekeeper discovery
messages, it signals the endpoint or gateway to use this address in future
communications.
Note
Setting this address for one local zone makes it the address used for all
local zones.
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Step 3
Command
Purpose
Router(config-gk)# zone prefix
gatekeeper-name e164-prefix
[gw-priority pri-0-to-10
gw-alias [gw-alias, ...]]
Adds a prefix to the gatekeeper zone list. To remove knowledge of a zone
prefix, use the no form of this command with the gatekeeper name and prefix.
To remove the priority assignment for a specific gateway, use the no form of
this command with the gw-priority option.
The keywords and arguments are as follows:
•
gatekeeper-name—Specifies the name of a local or remote gatekeeper,
which must have been defined by using the zone local or zone remote
command.
•
e164-prefix—Specifies an E.164 prefix in standard form followed by dots
(.). Each dot represents a number in the E.164 address. For example,
212....... is matched by 212 and any seven numbers.
Note
•
Although a dot representing each digit in an E.164 address is the
preferred configuration method, you can also enter an asterisk (*) to
match any number of digits.
gw-priority pri-0-to-10 gw-alias—(Optional) Use the gw-priority
option to define how the gatekeeper selects gateways in its local zone for
calls to numbers that begin with prefix e164-prefix. Do not use this option
to set priority levels for a prefix assigned to a remote gatekeeper.
Use values from 0 to 10. A 0 value prevents the gatekeeper from using the
gateway gw-alias for that prefix. Value 10 places the highest priority on
gateway gw-alias. If you do not specify a priority value for a gateway, the
value 5 is assigned.
To assign the same priority value for one prefix to multiple gateways, list
all the gateway names after the pri-0-to-10 value.
The gw-alias name is the H.323 identification (ID) of a gateway that is
registered or that will register with the gatekeeper. This name is set on the
gateway with the h323-gateway voip h.323-id command.
To put all your gateways in the same zone, use the gw-priority option and specify which gateways are
used for calling different area codes. For example:
zone
zone
zone
zone
local localgk xyz.com
prefix localgk 408.......
prefix localgk 415....... gw-priority 10 gw1 gw2
prefix localgk 650....... gw-priority 0 gw1
The above commands accomplish the following:
•
Domain xyz.com is assigned to gatekeeper localgk.
•
Prefix 408 is assigned to gatekeeper localgk, and no gateway priorities are defined for it; therefore,
all gateways registering to localgk can be used equally for calls to the 408 area code. No special
gateway lists are built for the 408 prefix; a selection is made from the master list for the zone.
•
The prefix 415 is added to gatekeeper localgk, and priority 10 is assigned to gateways gw1 and gw2.
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•
Prefix 650 is added to gatekeeper localgk, and priority 0 is assigned to gateway gw1.
•
A priority 0 is assigned to gateway gw1 to exclude it from the gateway pool for prefix 650. When
gw2 registers with gatekeeper localgk, it is added to the gateway pool for each prefix as follows:
– For gateway pool for 415, gateway gw2 is set to priority 10.
– For gateway pool for 650, gateway gw2 is set to priority 5.
Configuring a Gatekeeper for Interaction with External Applications
There are two ways of configuring the gatekeeper for interaction with an external application. You can
configure a port number where the gatekeeper listens for dynamic registrations from applications. Using
this method, the application connects to the gatekeeper and specifies the trigger conditions in which it
is interested.
The second method involves using the command-line interface to statically configure the information
about the application and its trigger conditions, in which case the gatekeeper initiates a connection to
the external application.
To configure a gatekeeper (sj.xyz.com) that uses port 20000 for a specific connection with an external
server (Server-123), use the following commands beginning in global configuration mode. Server-123
has a number of triggers that are used to maintain a database of active gateways, which are used for active
call resolution.
Command
Purpose
Step 1
Router(config)# gatekeeper
Enters gatekeeper configuration mode.
Step 2
Router(config)# server registration-port port-number
Establishes the server registration port that is used for
communication between the server and the
gatekeeper. The port-number argument specifies a
single range of values from 1 through 65,535 for the
port number on which the gatekeeper listens for
external server connections.
Server-123 establishes a connection with gatekeeper sj.xyz.com on port 20000 and sends a REGISTER
RRQ message to gatekeeper sj.xyz.com to express interest in all RRQs from voice gateways that support
a technology prefix of 1# or 2#.
The following is an example of a registration message:
REGISTER RRQ
Version-id:1
From:Server-123
To:sj.xyz.com
Priority:2
Notification-Only:
Content-Length:29
t=voice-gateway
p=1#
p=2#
When gatekeeper sj.xyz.com receives this message, the information supplied in the message is added to
the trigger list. Then, when an endpoint registers with this gatekeeper by using an RRQ that matches the
specified trigger condition in the message, the gatekeeper sends a notification to Server-123.
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The following is an example of an RRQ notification sent from the gatekeeper to the server when the
above trigger condition matches:
REQUEST RRQ
Version-id:1
From:sj.xyz.com
To:Server-123
Notification-Only:
Content-Length:89
c=I:172.18.00.00:1720
r=I:172.20.01.40:16523
a=H:gw3-sj
t=voice-gateway
p=1# 2#
Configuring a Prefix to a Gatekeeper Zone List
To add a prefix to a gatekeeper zone list, use the following commands beginning in global configuration
mode:
Command
Purpose
Step 1
Router(config)# gatekeeper
Enters gatekeeper configuration mode.
Step 2
Router(config-gk)# zone prefix gatekeeper-name
e164-prefix [blast | seq] [gw-priority priority
gw-alias [gw-alias, ...]]
Adds a prefix to the gatekeeper zone list.
Note
For an explanation of the keywords and arguments,
see Step 3 of the configuration task table in the
“Starting a Gatekeeper” section on page 304.
Note that the zone prefix command matches a prefix to a gateway. It does not register the gateway.
The gateway must register with the gatekeeper before calls can be completed through that gateway.
Verifying an Added Prefix
To view the prefixes added to the gatekeeper zone list, use the show gatekeeper zone prefix command.
To see gatekeeper zone information, use the show gatekeeper zone status command.
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Configuring Gatekeeper Triggers for Interaction with External Applications
To establish statically configured triggers on a router, use the following commands beginning in global
configuration mode:
Command
Purpose
Step 1
Router(config)# gatekeeper
Enters gatekeeper configuration mode.
Step 2
Router(config)# server trigger
{arq | lcf | lrj | lrq | rrq | urq} gkid priority
server-id server-ip-address server-port
Configures a static server trigger for external
applications. Enter the all form of the no server trigger
all command to remove every static trigger that you
configured if you want to delete them all.
The keywords and arguments are as follows:
Step 3
Router(config)# info-only
•
all—Deletes all command-line interface- (CLI-)
configured triggers.
•
arq, lcf, lrj, lrq, rrq, urq—Specifies Registration,
Admission, and Status (RAS) message types. Use
these message types to specify a submode in the
gatekeeper configuration mode where you configure
a trigger for the gatekeeper to act upon. Specify only
one message type per server trigger command. There
is a different trigger submode for each message type.
Each trigger submode has its own set of applicable
commands.
•
gkid—Specifies the local gatekeeper identifier.
•
priority—Specifies the priority for each trigger. The
range is from 1 through 20, with 1 being the highest
priority.
•
server-id—Specifies the identification (ID) number
of the external application.
•
server-ip-address—Specifies the IP address of the
server.
•
server-port—Specifies the port on which the
Cisco IOS gatekeeper listens for messages from the
external server connection.
Indicates to the Cisco IOS gatekeeper that messages that
meet the specified trigger parameters should be sent as
notifications only and that the Cisco IOS gatekeeper
should not wait for a response from the external
application.
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Step 4
Command
Purpose
Router(config)# destination-info {e164 | email-id
| h323-id} value
Configures a trigger that is based on a particular
destination. Repeat this command for more destinations.
The keywords and arguments are as follows:
•
e164—Indicates that the destination address is an
E.164 address.
•
email-id—Indicates that the destination address is an
e-mail ID.
•
h323-id—Indicates that the destination address is an
H.323 ID.
•
value—Specifies the value against which to compare
the destination address in the RAS messages. For
E.164 addresses, the following wildcards can be
used:
– A trailing series of periods, each of which
represents a single character.
– A trailing asterisk, which represents one or more
characters.
Step 5
Router(config)# redirect-reason value
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Configures a trigger that is based on a specific redirect
reason. Repeat this command for more destinations. The
value argument specifies the value against which to
compare the redirect reason in the RAS messages.
Possible values are from 0 to 65,535. Currently used
redirect reasons are as follows:
•
0—Unknown reason.
•
1—Call forwarding is busy or called DTE is busy.
•
2—Call forwarded; no reply.
•
4—Call deflection.
•
9—Called DTE out of order.
•
10—Call forwarding by the call DTE 15—Call
forwarding unconditionally.
•
15—Call forwarding unconditionally.
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Step 6
Command
Purpose
Router(config)# remote-ext-address [e164] value
Limits the qualifying messages based on the remote
extension address. Repeat this command for more
destinations.
The keywords and arguments are as follows:
•
e164—(Optional) Indicates that the remote extension
address is an E.164 address.
•
value—Specifies the value against which to compare
the destination address in the RAS messages. The
following wildcards can be used:
– A trailing series of periods, each of which
represents a single character.
– A trailing asterisk, which represents one or more
characters.
Step 7
Step 8
Router(config)# endpoint-type value
Router(config)# supported-prefix value
Note
Configures a trigger that is based on a specific endpoint.
Repeat this command for more destinations. The value
argument specifies the value against which to compare
the endpoint type in the RAS messages. The possible
values are as follows:
•
gatekeeper—Specifies that the endpoint is an H.323
gatekeeper.
•
h320-gateway—Specifies that the endpoint is an
H.320 gateway.
•
mcu—Specifies that the endpoint is a multipoint
control unit (MCU).
•
other-gateway—Specifies that the endpoint is a type
of gateway not specified on this list.
•
proxy—Specifies that the endpoint is an H.323
proxy.
•
terminal—Specifies that the endpoint is an H.323
terminal.
•
voice-gateway—Specifies that the endpoint is a
voice type gateway.
Configures a trigger that is based on a specific supported
prefix. Repeat this command for more destinations. The
value argument specifies the value against which to
compare the supported prefix in the RAS messages. The
possible values are any E.164 pattern used as a gateway
technology prefix. The value string may contain any of
the following: 0123456789#*,
Repeat Steps 2 through 8 in the above configuration task table for each trigger that you want to
define.
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Note
To remove a trigger, enter the no server trigger command. To temporarily suspend a trigger, enter
the trigger configuration mode, as described in Step 2, and enter the shutdown subcommand.
Configuring Inbound or Outbound Gatekeeper Proxied Access
By default, a gatekeeper will offer the IP address of the local proxy when queried by a remote gatekeeper
(synonymous with remote zone). This is considered proxied access. Before Cisco IOS Release 12.0(5)T,
the local gatekeeper was configured using the zone access command to offer the address of the local
endpoint instead of the address of the local proxy (considered direct access).
Note
The use-proxy command replaces the zone access command. The use-proxy command, configured
on a local gatekeeper, affects only the use of proxies for incoming calls (that is, it does not affect the
use of local proxies for outbound calls). When originating a call, a gatekeeper will use a proxy only
if the remote gatekeeper offers a proxy at the remote end. A call between two endpoints in the same
zone will always be a direct (nonproxied) call.
To configure a proxy for inbound calls from remote zones to gateways in its local zone and to configure
a proxy for outbound calls from gateways in its local zone to remote zones, use the following commands
beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# gatekeeper
Enters gatekeeper configuration mode.
Step 2
Router(config-gk)# use-proxy local-zone-name
{default | remote-zone remote-zone-name} {inbound-to
| outbound-from} {gateway | terminal}
Enables proxy communications for calls between
local and remote zones.
The keywords and arguments are as follows:
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•
local-zone-name—Specifies the name or zone
name of the gatekeeper, which is usually the fully
domain-qualified host name of the gatekeeper.
For example, if the domain name is cisco.com,
the gatekeeper name might be gk1.cisco.com.
However, if the gatekeeper is controlling
multiple zones, the name of the gatekeeper for
each zone should be a unique string that has a
mnemonic value.
•
default—Defines the default proxy policy for all
calls that are not defined by a use-proxy
command that includes the remote-zone
keyword.
•
remote-zone remote-zone-name—Defines a
proxy policy for calls to or from a specific remote
gatekeeper or zone.
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Command
Purpose
•
inbound-to—Applies the proxy policy to calls
that are inbound to the local zone from a remote
zone. Each use-proxy command defines the
policy for only one direction.
•
outbound-from—Applies the proxy policy to
calls that are outbound from the local zone to a
remote zone. Each use-proxy command defines
the policy for only one direction.
•
gateway—Defines the type of local device to
which the policy applies. The gateway option
applies the policy only to local gateways.
•
terminal—Defines the type of local device to
which the policy applies. The terminal option
applies the policy only to local terminals.
Verifying Gatekeeper Proxied Access Configuration
Use the show gatekeeper zone status command to see information about the configured gatekeeper
proxies and gatekeeper zone information (as shown in the following output).
Router# show gatekeeper zone status
GK name
GATEKEEPER ZONES
================
Domain Name
RAS Address
PORT
FLAGS MAX-BW
(kbps)
----- -----0
CUR-BW
(kbps)
------
-----------------------------sj.xyz.com
xyz.com
10.0.0.9 1719 LS
SUBNET ATTRIBUTES :
All Other Subnets :(Enabled)
PROXY USAGE CONFIGURATION :
inbound calls from germany.xyz.com :
to terminals in local zone sj.xyz.com :use proxy
to gateways in local zone sj.xyz.com :do not use proxy
outbound calls to germany.xyz.com
from terminals in local zone germany.xyz.com :use proxy
from gateways in local zone germany.xyz.com :do not use proxy
inbound calls from all other zones :
to terminals in local zone sj.xyz.com :use proxy
to gateways in local zone sj.xyz.com :do not use proxy
outbound calls to all other zones :
from terminals in local zone sj.xyz.com :do not use proxy
from gateways in local zone sj.xyz.com :do not use proxy
tokyo.xyz.co xyz.com
172.21.139.89
1719 RS
0
milan.xyz.co xyz.com
172.16.00.00
1719 RS
0
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H.323 Gatekeeper Configuration Task List
Configuring a Forced Disconnect on a Gatekeeper
To force a disconnect on a gatekeeper, use the following command in privileged EXEC mode:
Command
Purpose
Router# clear h323 gatekeeper call
{all | local-callID local-callID}
Forces a disconnect on a specific call or on all calls currently
active on this gatekeeper.
The keywords and arguments are as follows:
•
all—Forces all active calls currently associated with this
gatekeeper to be disconnected.
•
local-callID—Forces a single active call associated with this
gatekeeper to be disconnected.
•
local-callID—Specifies the local call identification number
(CallID) that identifies the call to be disconnected.
To force a particular call to be disconnected (as opposed to all active calls on the H.323 gateway), use
the local call identification number (CallID) to identify that specific call. Find the local CallID number
for a specific call by using the show gatekeeper calls command; the ID number is displayed in the
LocalCallID column.
Verifying a Forced Disconnect
To show the status of each ongoing call that a gatekeeper is aware of, use the show gatekeeper calls
command. If you have forced a disconnect either for a particular call or for all calls associated with a
particular H.323 gatekeeper, the system will not display information about those calls.
The following is sample output from the show gatekeeper calls command:
router# show gatekeeper calls
Total number of active calls =1
Gatekeeper Call Info
====================
LocalCallID
Age (secs)
BW
12-3339
94
768 (Kbps)
Endpt(s): Alias
E.164Addr
CallSignalAddr
Port
RASSignalAddr
src EP: epA
10.0.0.11
1720
10.0.0.11
dst EP: epB2zoneB.com
src PX: pxA
10.0.0.1
1720
10.0.0.11
dst PX: pxB
172.21.139.90
1720
172.21.139.90
Port
1700
24999
24999
Configuring the Proxy
This section describes the following configuration tasks for configuring the proxy. Depending on your
specific network design, either the first task or the second task is required.
•
Configuring a Proxy Without ASR, page 333
•
Configuring a Proxy with ASR, page 337
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H.323 Gatekeeper Configuration Task List
Configuring a Proxy Without ASR
To start the proxy without application-specific routing (ASR), start the proxy and then define the H.323
name, zone, and QoS parameters on the interface whose IP address the proxy will use. To start the proxy
without ASR, use the following commands beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# proxy h323
Starts the proxy feature.
Step 2
Router(config)# interface type number [name-tag]
Configures an interface type and enters interface
configuration mode.
Cisco 4000 Series with Channelized T1 or E1 and the Cisco MC3810
Router(config)# interface serial
number:channel-group
To configure a subinterface, use these forms of the interface
global configuration command:
The keywords and arguments are as follows:
•
type—Specifies the type of interface to be
configured. (See Table 26 that follows this
configuration task table.)
•
number—Specifies the port, connector, or
interface card number. On a Cisco 4000 series
router, specifies the network process monitor
(NPM) number. The numbers are assigned at the
factory at the time of installation or when added
to a system, and they can be displayed with the
show interfaces command.
•
name-tag—(Optional) Specifies the logic name
to identify the server configuration so that
multiple entries of server configuration can be
entered. This optional argument is for use with
the Redundant Link Manager (RLM) feature.
•
slot—Specifies the number of the slot being
configured. Refer to the appropriate hardware
manual for slot and port information.
•
port—Specifies the number of the port being
configured. Refer to the appropriate hardware
manual for slot and port information.
•
port-adapter—Specifies the number of the port
adapter being configured. Refer to the
appropriate hardware manual for information
about port adapter compatibility.
•
ethernet—(Optional) Specifies an Ethernet
IEEE 802.3 interface.
•
serial—(Optional) Specifies a serial interface.
Cisco 7200 Series
Router(config)# interface type
slot/port-adapter/port.subinterface-number
[multipoint | point-to-point]
Cisco 7200 Series and Cisco 7500 Series with a Packet over SONET
Interface Processor
Router(config)# interface type slot/port
Cisco 7500 Series
Router(config)# interface type
slot/port-adapter.subinterface-number [multipoint |
point-to-point][ethernet | serial]
Cisco 7500 Series with Channelized T1 or E1
Router(config)# interface serial
slot/port:channel-group
Cisco 7500 Series with Ports on VIP Cards
Router(config)# interface type
slot/port-adapter/port [ethernet | serial]
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Command
Purpose
•
:channel-group—Specifies a T1 channel group
number in the range 0 to 23 defined with the
channel-group controller configuration
command. On a dual port card, it is possible to
run channelized on one port and primary rate on
the other port.
Cisco MC3810 specifies the T1/E1 channel
group number in the range 0 to 23 defined with
the channel-group controller configuration
command.
Step 3
Step 4
Router(config-if)# h323 interface [port-number]
Router(config-if)# h323 h323-id h323-id
•
.subinterface-number—Specifies a subinterface
number in the range of 1 to 4,294,967,293. The
number that precedes the period (.) must match
the number to which this subinterface belongs.
•
multipoint | point-to-point—(Optional)
Specifies a multipoint or point-to-point
subinterface. There is no default.
Selects an interface whose IP address will be used by
the proxy to register with the gatekeeper. The
port-number argument specifies the port number on
which the proxy will listen for incoming call setup
requests:
•
The port-number range is from 1 to 65,356. The
default port number for the proxy is 11,720 in
-isx- or -jsx- Cisco IOS images.
•
The default port number for the proxy is 1720 in
-ix- Cisco IOS images, which do not contain the
Voice over IP (VoIP) gateway.
Configures the proxy name. (More than one name
may be configured if necessary.)
The h323-id argument specifies the name of the
proxy. It is recommended that this be a fully qualified
e-mail identification (ID), with the domain name
being the same as that of its gatekeeper.
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Step 5
Command
Purpose
Router(config-if)# h323 gatekeeper [id
gatekeeper-id] {ipaddr ipaddr [port] | multicast}
Specifies the gatekeeper associated with a proxy and
controls how the gatekeeper is discovered.
The keywords and arguments are as follows:
Step 6
Router(config-if)# h323 qos {ip-precedence value |
rsvp {controlled-load | guaranteed-qos}}
•
id gatekeeper-id—(Optional) Specifies the
gatekeeper name. Typically, this is a Domain
Name System (DNS) name, but it can also be a
raw IP address in dotted form. If this parameter
is specified, gatekeepers that have either the
default or the explicit flags set for the subnet of
the proxy will respond. If this parameter is not
specified, only those gatekeepers with the default
subnet flag will respond.
•
ipaddr ipaddr [port]—If this parameter is
specified, the gatekeeper discovery message will
be unicast to this address and, optionally, to the
port specified.
•
multicast—If this parameter is specified, the
gatekeeper discovery message will be multicast
to the well-known Registration, Admission, and
Status (RAS) multicast address and port.
Enables quality of service (QoS) on the proxy.
The keywords and arguments are as follows:
Step 7
Router(config-if)# ip route-cache [cbus]
same-interface [flow] distributed
•
ip-precedence value—Specifies that Realtime
Transport Protocol (RTP) streams should set
their IP precedence bits to the specified value.
•
rsvp [controlled-load]—Specifies controlled
load class of service.
•
rsvp [guaranteed-qos]—Specifies guaranteed
QoS class of service.
Controls the use of high-speed switching caches for
IP routing.
The keywords are as follows:
•
cbus—(Optional) Enables both autonomous
switching and fast switching.
•
same-interface—Enables fast-switching packets
to back out through the interface on which they
arrived.
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Command
Purpose
•
flow—(Optional) Enables the Route Switch
Processor (RSP) to perform flow switching on
the interface.
•
distributed—Enables Versatile Interface
Processor (VIP) distributed switching on the
interface. This feature can be enabled on
Cisco 7500 series routers with RSP and VIP
controllers. If both the ip route-cache flow
command and the ip route-cache distributed
command are configured, the VIP does
distributed flow switching. If only the ip
route-cache distributed command is
configured, the VIP does distributed switching.
Table 26 lists interface types that may be used for the type argument in Step 2 of the configuration task
table in the “Configuring a Proxy Without ASR” section on page 333.
Table 26
Interface Type Keywords
Keyword
Interface Type
async
Port line used as an asynchronous interface.
atm
ATM interface.
bri
ISDN BRI. This interface configuration is propagated to each of the B channels.
B channels cannot be individually configured. The interface must be configured
with dial-on-demand commands for calls to be placed on that interface.
dialer
Dialer interface.
ethernet
Ethernet IEEE 802.3 interface.
fastethernet
100-Mbps Ethernet interface on the Cisco 4500, Cisco 4700, Cisco 7000, and
Cisco 7500 series routers.
fddi
FDDI interface.
group-async
Master asynchronous interface.
hssi
High-Speed Serial Interface (HSSI).
lex
LAN Extender (LEX) interface.
loopback
Software-only loopback interface that emulates an interface that is always up. It
is a virtual interface supported on all platforms. The interface-number is the
number of the loopback interface that you want to create or configure. There is
no limit on the number of loopback interfaces you can create.
null
Null interface.
port-channel
Port channel interface.
pos
Packet OC-3 interface on the Packet over SONET Interface Processor.
serial
Serial interface.
switch
Switch interface.
tokenring
Token Ring interface.
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Table 26
Interface Type Keywords (continued)
Keyword
Interface Type
tunnel
Tunnel interface; a virtual interface. The number is the number of the tunnel
interface that you want to create or configure. There is no limit on the number
of tunnel interfaces you can create.
vg-anylan
100VG-AnyLAN port adapter.
Configuring a Proxy with ASR
To enable ASR on the proxy, start the proxy and then define the H.323 name, zone, and QoS parameters
on the loopback interface. Next, determine which interface will be used to route the H.323 traffic and
configure ASR on it. The ASR interface and all other interfaces must be separated so that routing
information never travels from one to the other. There are two different ways to separate the ASR
interface and all other interfaces:
•
Use one type of routing protocol on the ASR interface and another on all the non-ASR interfaces.
Include the loopback subnet in both routing domains.
•
Set up two different autonomous systems, one that contains the ASR network and the loopback
network and another that contains the other non-ASR networks and loopback network.
To ensure that the ASR interface and all other interfaces never route packets between each other,
configure an access control list. (The proxy traffic will be routed specially because it is always addressed
to the loopback interface first and then translated by the proxy subsystem.)
To start the proxy with ASR enabled on the proxy using one type of routing protocol on the ASR
interface and another on all of the non-ASR interfaces, and with the loopback subnet included in both
routing domains, use the following commands beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# proxy h323
Starts the proxy.
Step 2
Router(config)# interface type number [name-tag]
Enters loopback interface configuration mode.
For an explanation of the arguments, see Step 2 in the
“Configuring a Proxy Without ASR” configuration
task table.
To configure a proxy with ASR enabled on the proxy
using one type of routing protocol, the type argument
is loopback. The loopback type specifies the
software-only loopback interface that emulates an
interface that is always up. It is a virtual interface
supported on all platforms. The number argument is
the number of the loopback interface that you want to
create or configure. There is no limit on the number
of loopback interfaces that you can create.
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Step 3
Command
Purpose
Router(config-if)# ip address ip-address mask
[secondary]
Sets a primary or secondary IP address for an
interface.
The keywords and arguments are as follows:
Step 4
Router(config-if)# h323 interface [port-number]
•
ip-address—Specifies the IP address.
•
mask—Specifies the mask for the associated IP
subnet.
•
secondary—(Optional) Specifies that the
configured address is a secondary IP address. If
this keyword is omitted, the configured address is
the primary IP address.
Signals the proxy that this interface IP address is the
one to use.
For an explanation of the arguments, see Step 3 in the
configuration task table in the “Configuring a Proxy
Without ASR” section on page 333.
Step 5
Router(config-if)# h323 h323-id h323-id
Configures the proxy name. (More than one name can
be configured if necessary.)
The h323-id argument specifies the name of the
proxy. It is recommended that this be a fully qualified
e-mail identification (ID), with the domain name
being the same as that of its gatekeeper.
Step 6
Router(config-if)# h323 gatekeeper [id
gatekeeper-id] {ipaddr ipaddr [port] | multicast}
Specifies the gatekeeper associated with a proxy and
controls how the gatekeeper is discovered.
For an explanation of the keywords and arguments,
see Step 5 in the configuration task table in the
“Configuring a Proxy Without ASR” section on
page 333.
Step 7
Router(config-if)# h323 qos {ip-precedence value |
rsvp {controlled-load | guaranteed-qos}}
Enables quality of service (QoS) on the proxy.
For an explanation of the keywords and arguments,
see Step 6 in the configuration task table in the
“Configuring a Proxy Without ASR” section on
page 333.
Step 8
Router(config-if)# interface type number [name-tag]
If ASR is to be used, enters the interface through
which outbound H.323 traffic should be routed.
For an explanation of the keywords and arguments,
see Step 2 in the configuration task table in the
“Configuring a Proxy Without ASR” section on
page 333.
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Step 9
Command
Purpose
Router(config-if)# h323 asr [bandwidth
max-bandwidth]
Enables ASR and specifies the maximum bandwidth
for a proxy.
The keywords and arguments are as follows:
•
Step 10
Router(config-if)# ip address ip-address mask
[secondary]
bandwidth max-bandwidth—Specifies the
maximum bandwidth on the interface. Value
ranges are from 1 to 10,000,000 kbps. If you do
not specify a value for the max-bandwidth
argument, the value defaults to the bandwidth on
the interface. If you specify the max-bandwidth
value as a value greater than the interface
bandwidth, the bandwidth will default to the
interface bandwidth.
Sets up the ASR interface network number.
For an explanation of the keywords and arguments,
see Step 3 in this configuration task table.
Step 11
Router(config-if)# exit
Exits interface configuration mode and returns to
global configuration mode.
Step 12
Router(config)# interface type number [name-tag]
Enters interface configuration mode for a non-ASR
interface.
For an explanation of the keywords and arguments,
see Step 2 in the configuration task table in the
“Configuring a Proxy Without ASR” section on
page 333.
Step 13
Router(config-if)# ip address ip-address mask
[secondary]
Sets up a non-ASR interface network number.
For an explanation of the keywords and arguments,
see Step 3 in this configuration task table.
Step 14
Router(config-if)# exit
Exits interface configuration mode.
Step 15
Router(config)# router rip
Configures the Routing Information Protocol (RIP)
for a non-ASR interface.
Step 16
Router(config)# network network-number
Specifies a list of networks for the RIP routing
process or a loopback interface in an Interior
Gateway Routing Protocol (IGRP) domain. The
network-number argument specifies the IP address of
the directly connected networks.
Step 17
Router(config)# router igrp autonomous-system
Configures Interior IGRP for an ASR interface. The
autonomous-system argument specifies the
autonomous system number that identifies the routes
to the other IGRP routers. It is also used to tag the
routing information.
Step 18
Router(config)# network network-number
Specifies a list of networks for the Routing
Information Protocol (RIP) routing process. The
network-number argument should include an ASR
interface in an IGRP domain.
Step 19
Router(config)# network loopback-addr
Includes a loopback interface in an IGRP domain.
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Step 20
Command
Purpose
Router(config)# access-list access-list-number
{permit | deny} source source-mask [destination
destination-mask] {eq | neq} [[source-object]
[destination-object] [identification] any]
Creates an access list.
The keywords and arguments are as follows:
•
access-list-number—Specifies the integer that
you choose. The number should be between 300
and 399, and it uniquely identifies the access list.
•
permit—Permits access when there is an address
match.
•
deny—Denies access when there is an address
match.
•
source—Specifies the source address. DECnet
addresses are written in the form area.node. For
example, 50.4 is node 4 in area 50. All addresses
are in decimal.
•
source-mask—Specifies the mask to be applied
to the address of the source node. All masks are
in decimal.
•
destination—(Optional) Specifies the DECnet
address of the destination node in decimal
format. DECnet addresses are written in the form
area.node. For example, 50.4 is node 4 in area 50.
All addresses are in decimal.
•
destination-mask—(Optional) Specifies the
destination mask. DECnet addresses are written
in the form area.node. For example, 50.4 is node
4 in area 50. All masks are in decimal.
•
eq—Specifies that the item matches the packet if
all the specified parts of the source object,
destination object, and identification match the
data in the packet.
•
neq—Specifies that the item matches the packet
if any of the specified parts do not match the
corresponding entry in the packet.
•
source-object—(Optional) Contains the
mandatory keyword src and one of the following
optional keywords:
– eq | neq | lt | gt—Specifies equal to, not
equal to, less than, or greater than. These
keywords must be followed by the argument
object-number, a numeric DECnet object
number.
– exp—Stands for expression; followed by a
regular-expression that matches a string. See
the “Regular Expressions” appendix in the
Cisco IOS Dial Technologies Command
Reference for a description of regular
expressions.
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Command
Purpose
•
destination-object—(Optional) Contains the
mandatory keyword dst and one of the following
optional keywords:
– eq | neq | lt | gt—Specifies equal to, not
equal to, less than, or greater than. These
keywords must be followed by the argument
object-number, a numeric DECnet object
number.
– exp—Stands for expression; followed by a
regular expression that matches a string. See
the “Regular Expressions” appendix in the
Cisco IOS Dial Technologies Command
Reference for a description of regular
expressions.
– uic—Stands for user identification code;
followed by a numeric UID expression. The
argument [group, user] is a numeric UID
expression. In this case, the bracket symbols
are literal; they must be entered. The group
and user parts can be specified either in
decimal, in octal by prefixing the number
with a 0, or in hex by prefixing the number
with 0x. The uic expression displays as an
octal number.
•
identification—(Optional) Uses any of the
following three keywords:
– id—Specifies regular expression; refers to
the user ID.
– password—Specifies regular expression;
the password to the account.
– account—Specifies regular expression; the
account string.
– any—(Optional) Specifies that the item
matches if any of the specified parts do
match the corresponding entries for
source-object, destination-object, or
identification.
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Step 21
Command
Purpose
Router(config)# interface type number [name-tag]
Enters interface configuration mode on an ASR
interface.
For an explanation of the keywords and arguments,
see Step 2 in the configuration task table in the
“Configuring a Proxy Without ASR” section on
page 333.
Step 22
Router(config-if)# ip access-group
{access-list-number | access-list-name}{in | out}
Controls access to an interface.
Use this command to set the outbound access group
and then the inbound access group.
The keywords and arguments are as follows:
Note
•
access-list-number—Specifies the number of an
access list. This is a decimal number from 1 to
199 or from 1300 to 2699.
•
access-list-name—Name of an IP access list as
specified by an IP access-list command.
•
in—Filters on inbound packets.
•
out—Filters on outbound packets.
ASR is not supported on Frame Relay or ATM interfaces for the Cisco MC3810 platform.
To start the proxy with ASR enabled on the proxy using two different autonomous systems (one that
contains the ASR network and the loopback network and another that contains the other non-ASR
networks and the loopback network), use the following commands beginning in global configuration
mode:
Command
Purpose
Step 1
Router(config)# proxy h323
Starts the proxy.
Step 2
Router(config)# interface type number [name-tag]
Enters loopback interface configuration mode.
For an explanation of the arguments, see Step 2 in the
configuration task table in the “Configuring a Proxy
Without ASR” section on page 333.
To start the proxy with ASR enabled on the proxy
using two different autonomous systems, the type
argument is loopback. The loopback type specifies
the software-only loopback interface that emulates an
interface that is always up. It is a virtual interface
supported on all platforms. The number argument is
the number of the loopback interface that you want to
create or configure. There is no limit on the number
of loopback interfaces you can create.
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Step 3
Command
Purpose
Router(config-if)# ip address ip-address mask
[secondary]
Sets a primary or secondary IP address for an
interface.
The keywords and arguments are as follows:
Step 4
Router(config-if)# h323 interface [port-number]
•
ip-address—Specifies the IP address.
•
mask—Specifies the mask for the associated IP
subnet.
•
secondary—(Optional) Specifies that the
configured address is a secondary IP address. If
this keyword is omitted, the configured address is
the primary IP address.
Signals the proxy that this interface IP address is the
one to use.
For an explanation of the arguments, see Step 3 in the
configuration task table in the “Configuring a Proxy
Without ASR” section on page 333.
Step 5
Router(config-if)# h323 h323-id h323-id
Configures the proxy name. (More than one name can
be configured if necessary.)
The h323-id argument specifies the name of the
proxy. It is recommended that this be a fully qualified
e-mail identification (ID), with the domain name
being the same as that of its gatekeeper.
Step 6
Router(config-if)# h323 gatekeeper [id
gatekeeper-id] {ipaddr ipaddr [port] | multicast}
Specifies the gatekeeper associated with a proxy and
controls how the gatekeeper is discovered.
For an explanation of the keywords and arguments,
see Step 5 in the configuration task table in the
“Configuring a Proxy Without ASR” section on
page 333.
Step 7
Router(config-if)# h323 qos {ip-precedence value |
rsvp {controlled-load | guaranteed-qos}}
Enables quality of service (QoS) on the proxy.
The keywords and arguments are as follows:
Step 8
Router(config-if)# interface type number [name-tag]
•
ip-precedence value—Specifies that Real-time
Transport Protocol (RTP) streams should set
their IP precedence bits to the specified value.
•
rsvp {controlled-load}—Specifies controlled
load class of service.
•
rsvp {guaranteed-qos}—Specifies guaranteed
QoS class of service.
If application-specific routing (ASR) is to be used,
enters the interface through which outbound H.323
traffic should be routed.
For an explanation of the keywords and arguments,
see Step 2 in the configuration task table in the
“Configuring a Proxy Without ASR” section on
page 333.
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Step 9
Command
Purpose
Router(config-if)# h323 asr [bandwidth
max-bandwidth]
Enables ASR and specifies the maximum bandwidth
for a proxy.
The optional max-bandwidth argument specifies the
maximum bandwidth on the interface. Value ranges
are from 1 to 10,000,000 kbps. If you do not specify
max-bandwidth, this value defaults to the bandwidth
on the interface. If you specify max-bandwidth as a
value greater than the interface bandwidth, the
bandwidth will default to the interface bandwidth.
Step 10
Router(config-if)# ip address ip-address mask
[secondary]
Sets up the ASR interface network number.
For an explanation of the keywords and arguments,
see Step 3 in this configuration task table.
Step 11
Router(config-if)# exit
Exits interface configuration mode and returns to
global configuration mode.
Step 12
Router(config)# interface type number [name-tag]
Enters interface configuration mode on a non-ASR
interface.
For an explanation of the keywords and arguments,
see Step 2 in the configuration task table in the
“Configuring a Proxy Without ASR” section on
page 333.
Step 13
Router(config-if)# ip address ip-address mask
[secondary]
Sets up a non-ASR interface network number.
For an explanation of the keywords and arguments,
see Step 3 in this configuration task table.
Step 14
Router(config-if)# exit
Exits interface configuration mode.
Step 15
Router(config)# router igrp autonomous-system
Configures Interior Gateway Routing Protocol
(IGRP) for a non-ASR interface. The
autonomous-system argument specifies the
autonomous system number that identifies the routes
to the other IGRP routers. It is also used to tag the
routing information.
Step 16
Router(config)# network network-number
Includes a non-ASR interface in an IGRP domain.
The network-number argument specifies the IP
address of the network of the directly connected
networks.
Step 17
Router(config)# network network-number
Includes a loopback interface in an IGRP domain.
The network-number argument specifies the IP
address of the network of the directly connected
networks.
Step 18
Router(config)# router igrp autonomous-system
Configures IGRP for an ASR interface. The
autonomous-system argument specifies the
autonomous system number that identifies the routes
to the other IGRP routers. It is also used to tag the
routing information.
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Command
Purpose
Step 19
Router(config)# network network-number
Specifies a list of networks for the Routing
Information Protocol (RIP) routing process. The
network-number argument should include an ASR
interface in an IGRP domain.
Step 20
Router(config)# network network-number
Specifies a list of networks for the RIP routing
process. The network-number argument should
include a loopback interface in an IGRP domain.
Step 21
Router(config)# access-list access-list-number
{permit | deny} source source-mask [destination
destination-mask] {eq | neq} [[source-object]
[destination-object] [identification] any]
Creates an access list.
Router(config)# interface type number [name-tag]
Enters interface configuration mode on an ASR
interface.
Step 22
For an explanation of the keywords and arguments,
see Step 20 in the configuration task table in the
“Configuring a Proxy with ASR” section on
page 337.
For an explanation of the keywords and arguments,
see Step 2 in the configuration task table in the
“Configuring a Proxy Without ASR” section on
page 333.
Step 23
Controls access to an interface.
Router(config-if)# ip access-group
{access-list-number | access-list-name} {in | out}
Use this command to set the outbound access group
and then the inbound access group.
The keywords and arguments are as follows:
•
access-list-number—Specifies the number of an
access list. This is a decimal number from 1 to
199 or from 1300 to 2699.
•
access-list-name—Name of an IP access list as
specified by an IP access-list command.
•
in—Filters on inbound packets.
•
out—Filters on outbound packets.
H.323 Gatekeeper Configuration Examples
This section includes the following configuration examples:
•
Configuring a Gatekeeper Example, page 346
•
Redundant Gatekeepers for a Zone Prefix Example, page 347
•
Redundant Gatekeepers for a Technology Prefix Example, page 347
•
E.164 Interzone Routing Example, page 347
•
Configuring HSRP on the Gatekeeper Example, page 349
•
Using ASR for a Separate Multimedia Backbone Example, page 350
– Enabling the Proxy to Forward H.323 Packets, page 351
– Isolating the Multimedia Network, page 351
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– Configuring a Co-Edge Proxy with ASR Without Subnetting Example, page 352
– Co-Edge Proxy with Subnetting Example, page 354
– Configuring an Inside-Edge Proxy with ASR Without Subnetting Example, page 356
– Configuring a QoS-Enforced Open Proxy Using RSVP Example, page 357
– Configuring a Closed Co-Edge Proxy with ASR Without Subnetting Example, page 359
•
Defining Multiple Zones Example, page 360
•
Defining One Zone for Multiple Gateways Example, page 360
•
Configuring a Proxy for Inbound Calls Example, page 361
•
Configuring a Proxy for Outbound Calls Example, page 361
•
Removing a Proxy Example, page 362
•
H.235 Security Example, page 362
•
GKTMP and RAS Messages Example, page 363
•
Prohibiting Proxy Use for Inbound Calls Example, page 363
•
Disconnecting a Single Call Associated with an H.323 Gateway Example, page 363
•
Disconnecting All Calls Associated with an H.323 Gateway Example, page 363
Configuring a Gatekeeper Example
The following is an annotated example of how to configure a gatekeeper:
hostname gk-eng.xyz.com
! This router serves as the gatekeeper for the engineering community.
! at xyz.com.
ip domain-name xyz.com
! Domain name of this company.
interface Ethernet0
ip address 172.21.127.27 255.255.255.0
! This gatekeeper can be found at address 172.21.127.27.
gatekeeper
! Enter gatekeeper config mode.
zone local gk-eng.xyz.com xyz.com
! Because a zone is, by definition, the area of control of a gatekeeper,
! we tend to use the terms “zone name” and “gatekeeper name” synonymously.
! Here we use the host name as the name of the gatekeeper and zone.
! This is not necessary, but it does simplify administration.
zone remote gk-mfg.xyz.com xyz.com 172.12.10.14 1719
zone remote gk-corp.xyz.com xyz.com 172.12.32.80 1719
! A couple of other zones within xyz.com. We make lots of calls
! between these departments, so we just configure these so we save
! a little time bypassing DNS lookup to find their gatekeepers.
use-proxy gk-eng.xyz.com remote-zone gk-mfg.xyz.com direct
use-proxy gk-eng.xyz.com remote-zone gk-corp.xyz.com direct
use-proxy gk-eng.xyz.com default proxied
! We have good QoS on our local network, so we don't need proxies when
! calling between the xyz.com zones. But for all other zones, we want
! to use proxies.
zone subnet gk-eng.xyz.com 172.21.127.0/24 enable
no zone subnet gk-eng.xyz.com default enable
! We will accept registrations from our local subnet as long as they
! do not specify some other gatekeeper name. We will not accept any
! registrations from any other subnet.
zone bw gk-eng.xyz.com 2000
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! Preserve our good QoS by not allowing excessive amounts of H.323 traffic
! on the local network. This restricts the traffic within our zone,
! for both intra-zone and interzone calls, to 2 kbps at any given time.
alias static 172.21.127.49 gkid gk-eng.xyz.com terminal h323id joeblow ras
172.21.127.49 1719
! The “user” has an H.323 terminal, which does not support RAS. So we have
! to configure his alias manually so that callers can find him.
Redundant Gatekeepers for a Zone Prefix Example
In the following example, two remote gatekeepers are configured to service the same zone prefix:
gatekeeper
zone remote
zone remote
zone prefix
zone prefix
c2600-1-gk
c2514-1-gk
c2600-1-gk
c2514-1-gk
cisco.com 172.18.194.70 1719
cisco.com 172.18.194.71 1719
919.......
919.......
Redundant Gatekeepers for a Technology Prefix Example
In the following example, two remote gatekeepers are configured to service the same technology prefix:
gatekeeper
zone remote c2600-1-gk cisco.com 172.18.194.70 1719
zone remote c2514-1-gk cisco.com 172.18.194.71 1719
gw-type-prefix 3#* hopoff c2600-1-gk hopoff c2514-1-gk
E.164 Interzone Routing Example
Interzone routing may be configured by using E.164 addresses.
In this example, there are two gatekeepers that need to be able to resolve E.164 addresses. One is in San
Jose and the other is in New York. (See Figure 61.)
E.164 Interzone Routing
Non-H.323 network
Non-H.323 network
H.323 network
H.320
terminal
(over ISDN)
sj (408)
ny (212)
gw-ny2
gw-sj2
H.324
terminal
(over POTS)
gw-sj3
gw-sj4
gk-sj
IP
gk-ny
gw-ny3
gw-ny4
H.320
terminal
(over ISDN)
H.324
terminal
(over POTS)
Speech
only
(telephone)
Speech
only
(telephone)
12885
Figure 61
In sj (San Jose in the 408 area code), the gateways are configured to register with gk-sj as follows:
•
gw-sj2 configured to register with technology prefix 2#
•
gw-sj3 configured to register with technology prefix 3#
•
gw-sj4 configured to register with technology prefix 4#
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Similarly, in ny (New York in the 212 area code), gateways are configured to register with gk-ny as
follows:
•
gw-ny2 configured to register with technology prefix 2#
•
gw-ny3 configured to register with technology prefix 3#
•
gw-ny4 configured to register with technology prefix 4#
For the gatekeeper for San Jose, the configuration commands are as follows:
gatekeeper
zone local gk-sj cisco.com
zone remote gk-ny cisco.com 172.21.127.27
use-proxy gk-sj default direct
zone prefix gk-sj 408.......
zone prefix gk-ny 212.......
gw-type-prefix 3# hopoff gk-sj
gw-type-prefix 4# default-technology
For the gatekeeper for New York, the configuration commands are as follows:
gatekeeper
zone local gk-ny cisco.com
zone remote gk-sj cisco.com 172.21.1.48
use-proxy gk-ny default direct
zone prefix gk-sj 408.......
zone prefix gk-ny 212.......
gw-type-prefix 3# hopoff gk-ny
gw-type-prefix 4# default-technology
When a call is presented to gatekeeper gk-sj with the following target address in San Jose:
2#2125551212
Gatekeeper gk-sj recognizes that 2# is a technology prefix. It was not configured as such, but because
gw-sj2 registered with it, the gatekeeper now treats 2# as a technology prefix. It strips the prefix, which
leaves the telephone number 2125551212. This is matched against the zone prefixes that have been
configured. It is a match for 212......., so gk-sj knows that gk-ny handles this call. Gatekeeper gk-sj
forwards the entire address 2#2125551212 over to Gatekeeper gk-ny, which also looks at the technology
prefix 2# and routes it to gw-ny2.
When a call is presented to gatekeeper gk-sj with the following target address in San Jose:
2125551212
Gatekeeper gk-sj checks it against known technology prefixes but finds no match. It then checks it
against zone prefixes and matches on 212....... for gk-ny, and therefore routes this call to gk-ny.
Gatekeeper gk-ny does not have any local registrations for this address, and there is no technology prefix
on the address, but the default prefix is 4#, and gw-ny4 is registered with 4#, so the call gets routed to
gw-ny4.
Another call is presented to gatekeeper gk-sj with the following target address in San Jose:
3#2125551212
The call has technology prefix 3#, which is defined as a local hopoff prefix, so gk-sj routes this call to
gw-sj3, despite the fact that it has a New York zone prefix.
In this last example, a call is presented to gatekeeper gk-sj with the following target address in San Jose:
6505551212
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Gatekeeper gk-sj checks for a technology prefix match but does not find one. It then searches for a zone
prefix match and fails again. But there is a match for default gateway prefix of 4#, and gw-sj4 is
registered with 4#, so the call is routed out on gw-sj4.
Configuring HSRP on the Gatekeeper Example
This sample configuration uses Ethernet 0 as the HSRP interface on both gatekeepers.
On the primary gatekeeper, enter these commands:
configure terminal
! Enter global configuration mode.
interface ethernet 0
! enter interface configuration mode for interface ethernet 0.
standby 1 ip 172.21.127.55
! Member of standby group 1, sharing virtual address 172.21.127.55.
standby 1 preempt
! Claim active role when it has higher priority.
standby 1 timers 5 15
! Hello timer is 5 seconds; hold timer is 15 seconds.
standby 1 priority 110
! Priority is 110.
On the backup gatekeeper, enter these commands:
configure terminal
interface ethernet 0
standby 1 ip 172.21.127.55
standby 1 preempt
standby 1 timers 5 15
The configurations are identical except that gk2 has no standby priority configuration, so it assumes the
default priority of 100—meaning that gk1 has a higher priority.
On both gk1 and gk2, set up identical gatekeeper mode configurations, as follows:
configure terminal
! Enter global configuration mode.
gatekeeper
! Enter gatekeeper configuration mode.
zone local gk-sj cisco.com 172.21.127.55
! Define local zone using HSRP virtual address as gatekeeper RAS address.
.
.
.
! Various other gk-mode configurations.
no shut
! Bring up the gatekeeper.
configure terminal
! Enter global configuration mode.
gatekeeper
! Enter gatekeeper configuration mode.
zone local gk-sj cisco.com 172.21.127.55
! Define local zone using HSRP virtual address as gatekeeper RAS address.
! Note this uses the same gkname and address as on gk1.
.
.
! Various other gk-mode configurations.
no shut
! Bring up the gatekeeper.
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Note
The no shut command is issued on both gatekeepers, primary and secondary. If the show gatekeeper
status command is issued on the two gatekeepers, gk1 will show the following:
Gatekeeper State: UP
But gk2 will show the following:
Gatekeeper State: HSRP STANDBY
Using ASR for a Separate Multimedia Backbone Example
The examples in this section illustrate a separate multimedia backbone network dedicated to transporting
only H.323 traffic. The closed functionality of the H.323 proxy is necessary for creating this type of
backbone. Place a closed H.323 proxy on each edge of the multimedia backbone to achieve the following
goals:
•
The proxy directs all inter-proxy H.323 traffic, including Q.931 signaling, H.245, and media stream,
to the multimedia backbone.
•
The proxy shields the multimedia backbone so that routers on edge networks and other backbone
networks are not aware of its existence. In this way, only H.323-compliant packets can access or
traverse the multimedia backbone.
•
The proxy drops any unintended non-H.323 packets that attempt to access the multimedia backbone.
This section contains the following subsections:
•
Enabling the Proxy to Forward H.323 Packets, page 351
•
Isolating the Multimedia Network, page 351
•
Configuring a Co-Edge Proxy with ASR Without Subnetting Example, page 352
•
Co-Edge Proxy with Subnetting Example, page 354
•
Configuring an Inside-Edge Proxy with ASR Without Subnetting Example, page 356
•
Configuring a QoS-Enforced Open Proxy Using RSVP Example, page 357
•
Configuring a Closed Co-Edge Proxy with ASR Without Subnetting Example, page 359
Figure 62 illustrates a network that has a multimedia backbone. A gatekeeper (not shown) in the edge
network (zone) directs all out-of-zone H.323 calls to the closed proxy on the edge of that network. The
closed proxy forwards this traffic to the remote zone through the multimedia backbone. A closed proxy
and the edge router may reside in the same Cisco router, or they may be in separate routers, as shown in
Figure 62.
Figure 62
Sample Network with Multimedia Backbone
Multimedia
backbone
PX1
EP1
PX2
Edge net 2
Edge net 1
EP2
R2
R1
11391
Data backbone
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Enabling the Proxy to Forward H.323 Packets
To enable the proxy to forward H.323 packets received from the edge network to the multimedia
backbone, designate the interface that connects the proxy to the multimedia backbone to the ASR
interface by entering the h323 asr command in interface configuration mode. Enabling the proxy to
forward H.323 packets satisfies the first goal identified earlier in this section.
Because the proxy terminates two call legs of an H.323 call and bridges them, any H.323 packet that
traverses the proxy will have the proxy address either in its source field or in its destination field.
To prevent problems that can occur in proxies that have multiple IP addresses, designate only one
interface to be the proxy interface by entering the h323 interface command in interface configuration
mode. Then all H.323 packets that originate from the proxy will have the address of this interface in their
source fields, and all packets that are destined to the proxy will have the address of this interface in their
destination fields.
Figure 62 illustrates that all physical proxy interfaces belong either to the multimedia network or to the
edge network. These two networks must be isolated from each other for the proxy to be closed; however,
the proxy interface must be addressable from both the edge network and the multimedia network. For
this reason, a loopback interface must be created on the proxy and configured to the proxy interface.
It is possible to make the loopback interface addressable from both the edge network and the multimedia
network without exposing any physical subnets on one network to routers on the other network. Only
packets that originate from the proxy or packets that are destined to the proxy can pass through the proxy
interface to the multimedia backbone in either direction. All other packets are considered unintended
packets and are dropped. This can be achieved by configuring access control lists (ACLs) so that the
closed proxy acts like a firewall that only allows H.323 packets to pass through the ASR interface. This
satisfies the second goal identified earlier in this section, which is to ensure that only H.323-compliant
packets can access or traverse the multimedia backbone.
Isolating the Multimedia Network
The last step is to configure the network so that non-H.323 traffic never attempts to traverse the
multimedia backbone and so that it never risks being dropped by the proxy. This is achieved by
completely isolating the multimedia network from all edge networks and from the data backbone and by
configuring routing protocols on the various components of the networks.
The example provided in Figure 62 requires availability of six IP address classes, one for each of the
four autonomous systems and one for each of the two loopback interfaces. Any Cisco-supported routing
protocol can be used on any of the autonomous systems, with one exception: Routing Information
Protocol (RIP) cannot be configured on two adjacent autonomous systems because this protocol does not
include the concept of an autonomous system. The result would be the merging of the two autonomous
systems into one.
If the number of IP addresses are scarce, use subnetting, but the configuration can get complicated. In
this case, only the Enhanced IGRP, Open Shortest Path First (OSPF), and RIP Version 2 routing
protocols, which allow variable-length subnet masks (VLSMs), can be used.
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Assuming these requirements are met, configure the network illustrated in Figure 62 as follows:
•
Configure each of the four networks as a separate routing autonomous system and do not redistribute
routes between the multimedia backbone and any other autonomous system.
•
Create a loopback interface on the proxy and configure it to be the proxy interface. That way no
subnets of the multimedia backbone will be exposed to the edge network, or the other way around.
•
To ensure that the address of the loopback interface does not travel outside the edge network,
configure the appropriate distribution list on the edge router that connects the edge network to the
data backbone. Configuring the appropriate distribution list guarantees that any ongoing H.323 call
will be interrupted if the multimedia backbone fails. Otherwise, H.323 packets that originate from
one proxy and that are destined to another proxy might discover an alternate route using the edge
networks and the data backbone.
In some topologies, the two edge networks and the data backbone may be configured as a single
autonomous system, but it is preferable to separate them as previously described because they are
different networks with different characteristics.
The following examples illustrate the router configuration that is relevant to the closed proxy operation.
Configuring a Co-Edge Proxy with ASR Without Subnetting Example
See Figure 63 and the following configuration examples to see how to configure RIP on the two edge
networks and how to configure IGRP on the two backbone networks.
Figure 63
Sample Configuration Without Subnetting
E1: 172.22.0.1
E1: 172.22.0.2
L0: 10.0.0.1
L0: 10.0.0.0
E0: 172.20.0.1
Multimedia
backbone
PX1
E0: 172.23.0.1
PX2
EP1
Edge net 1
Edge net 2
EP2
R2
R1
Data backbone
172.23.0.2
E2: 172.21.0.2
E1: 172.21.0.1
PX1 Configuration
The following output is for the PX1 configuration:
!
proxy h323
!
interface Loopback0
ip address 10.0.0.0 255.0.0.0
!Assume PX1 is in Zone 1, and the gatekeeper resides in the same routers as PX1:
h323 interface
h323 h323-id [email protected]
h323 gatekeeper ipaddr 10.0.0.0
!
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interface Ethernet0
ip address 172.20.0.1 255.255.0.0
!
interface Ethernet1
ip address 172.22.0.1 255.255.0.0
ip access-group 101 in
ip access-group 101 out
h323 asr
!
router rip
network 172.20.0.0
network 10.0.0.0
!
router igrp 4000
network 172.22.0.0
network 10.0.0.0
!
access-list 101 permit ip any host 10.0.0.0
access-list 101 permit ip host 10.0.0.0 any
access-list 101 permit igrp any any
R1 Configuration
The following output is for the R1 configuration:
!
interface Ethernet0
ip address 172.20.0.2 255.255.0.0
!
interface Ethernet1
ip address 172.21.0.1 255.255.0.0
!
router rip
redistribute igrp 5000 metric 1
network 172.20.0.0
!
router igrp 5000
redistribute rip metric 10000 10 255 255 65535
network 172.21.0.0
distribute-list 10 out
!
access-list 10 deny ip 10.0.0.0 255.255.255
access-list 10 permit any
Note
The configuration for PX2 and R2 is the same as that for PX1 and R1.
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Co-Edge Proxy with Subnetting Example
Figure 64 and the examples that follow illustrate how to configure Enhanced IGRP on all networks.
Figure 64
Sample Configuration with Subnetting
E1: 172.21.2.2
E1: 172.21.2.1
L0: 172.21.20.1
L0: 172.21.10.1
E0: 172.21.0.1
Multimedia
backbone
PX1
E0: 172.21.3.1
PX2
EP1
Edge net 2
Edge net 1
EP2
R2
R1
172.21.3.2
Data backbone
E2: 172.21.1.2
E1: 172.21.1.1
PX1 Configuration
The following output is for the PX1 configuration:
!
proxy h323
!
interface Loopback0
ip address 172.21.10.1 255.255.255.192
h323 interface
h323 h323-id [email protected]
h323 gatekeeper ipaddr 172.21.20.1
!
interface Ethernet0
ip address 172.21.0.1 255.255.255.192
!
interface Ethernet1
ip address 172.21.2.1 255.255.255.192
ip access-group 101 in
ip access-group 101 out
h323 asr
!
router eigrp 4000
redistribute connected metric 10000 10 255 255 65535
passive-interface Ethernet1
network 172.21.0.0
distribute-list 10 out
no auto-summary
!
router eigrp 5000
redistribute connected metric 10000 10 255 255 65535
passive-interface Ethernet0
network 172.21.0.0
distribute-list 11 out
no auto-summary
!
access-list 10 deny 172.21.2.0 0.0.0.63
access-list 10 permit any
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access-list
access-list
access-list
access-list
access-list
11 deny 172.21.0.0 0.0.0.63
11 permit any
101 permit ip any host 172.21.10.1
101 permit ip host 172.21.10.1 any
101 permit eigrp any any
R1 Configuration
The following output is for the R1 configuration:
!
interface Ethernet0
ip address 172.21.0.2 255.255.255.192
!
interface Ethernet1
ip address 172.21.1.1 255.255.255.192
!
router eigrp 4000
redistribute eigrp 6000 metric 10000 10 255 255 65535
passive-interface Ethernet1
network 172.21.0.0
no auto-summary
!
router eigrp 6000
redistribute eigrp 4000 metric 10000 10 255 255 65535
passive-interface Ethernet0
network 172.21.0.0
distribute-list 10 out
no auto-summary
!
access-list 10 deny 172.21.10.0 0.0.0.63
access-list 10 permit any
Note
The configuration for PX2 and R2 is the same as that for PX1 and R1.
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Configuring an Inside-Edge Proxy with ASR Without Subnetting Example
The configuration of the co-edge proxy in Edge net 1 has already been presented above. Figure 65
illustrates the configuration of the inside-edge proxy PX2 and edge router R2 of Edge net 2. RIP is used
on the edge networks. IGRP is used on the data backbone and the multimedia backbone.
Figure 65
Edge Net 2 with Inside-Edge Proxy and No Subnetting
E1: 172.22.0.1
E1: 172.22.0.2
S0: 10.0.0.1
S0: 10.0.0.2
E0: 172.23.0.2
L0: 10.0.0.0
E0: 172.20.0.1
EP1
Multimedia
backbone
PX1
R2
Edge net 1
PX2
Edge net 2
EP2
E2: 172.21.0.2
R1
Data backbone
E1: 172.21.0.1
PX2 Configuration
The following output is for the PX2 configuration:
!
proxy h323
!
interface Ethernet0
ip address 172.23.0.2 255.255.0.0
!
interface Serial0
ip address 10.0.0.2 255.0.0.0
ip access-group 101 in
ip access-group 101 out
h323 interface
h323 asr
h323 h323-id [email protected]
h323 gatekeeper ipaddr 10.0.0.2
!
router rip
redistribute connected metric 10000 10 255 255 65535
network 172.23.0.0
!
access-list 101 permit ip any host 10.0.0.2
access-list 101 permit ip host 10.0.0.2 any
R2 Configuration
The following output is for the R2 configuration:
!
interface Ethernet0
ip address 172.23.0.1 255.255.0.0
!
interface Ethernet1
ip address 172.22.0.1 255.255.0.0
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H.323 Gatekeeper Configuration Examples
ip access-group 101 in
ip access-group 101 out
!
interface Ethernet2
ip address 172.21.0.2 255.255.0.0
!
interface Serial0
ip address 10.0.0.1 255.0.0.0
!
router rip
redistribute igrp 5000 metric 1
network 172.23.0.0
!
router igrp 4000
network 10.0.0.0
network 172.22.0.0
!
router igrp 5000
redistribute rip metric 10000 10 255 255 65535
network 172.21.0.0
distribute-list 10 out
!
ip route 10.0.0.2 255.255.255.255 Serial0
access-list 10 deny ip 10.0.0.0 255.255.255
access-list 10 permit any
access-list 101 permit ip any host 10.0.0.2
access-list 101 permit ip host 10.0.0.2 any
Note
To guarantee that all traffic between the proxy and other proxies is carried over the multimedia
backbone, run IGRP 4000 on the 10.0.0.0 network and on the 172.22.0.0 network. Make sure that the
H.323 proxy interface address (10.0.0.2) is not advertised over the data network (distribution list 10
in IGRP 5000). Doing this also eliminates the need to configure policy routes or static routes.
Configuring a QoS-Enforced Open Proxy Using RSVP Example
Figure 66 illustrates a proxy configuration that was created on a Cisco 2500 router with one Ethernet
interface and two serial interfaces. Only the Ethernet interface is in use.
Figure 66
Configuring a QoS-Enforced Open Proxy Using RSVP
PX1
172.21.127.38
R1
Edge net 1
172.21.127.39
GK1
Data backbone
11393
EP1
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PX1 Configuration
The following output is for the PX1 configuration:
!
version 11.3
no service password-encryption
service tcp-small-servers
!
hostname ExampleProxy
!
no ip domain-lookup
!
proxy h323
!
interface Ethernet0
ip address 172.21.127.38 255.255.255.192
no ip redirects
ip rsvp bandwidth 7000 7000
ip route-cache same-interface
fair-queue 64 256 1000
h323 interface
h323 qos rsvp controlled-load
h323 h323-id [email protected]
h323 gatekeeper ipaddr 172.21.127.39
!
interface Serial0
no ip address
shutdown
!
interface Serial1
no ip address
shutdown
!
router rip
network 172.21.0.0
!
ip classless
!
line con 0
exec-timeout 0 0
line aux 0
transport input all
line vty 0 4
password lab
login
!
end
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Configuring a Closed Co-Edge Proxy with ASR Without Subnetting Example
Figure 67 illustrates how to configure RIP on the edge networks and IGRP on the two backbone
networks. A Cisco 2500 router is used for the proxy.
Figure 67
Configuring a Closed Co-Edge Proxy with ASR
L0: 101.0.0.1
E0: 172.20.0.1
EP1
PX1
E0: 172.20.0.3
Edge net 1
R1
E1: 172.21.0.1
Multimedia
backbone
GK1
Data backbone
11394
E0: 172.20.0.2
S1: 172.22.0.1
PX1 Configuration
The following output is for the PX1 configuration:
!
version 11.3
no service password-encryption
service tcp-small-servers
!
hostname ExampleProxy
!
!
no ip domain-lookup
!
!
proxy h323
!
interface Loopback0
ip address 10.0.0.1 255.0.0.0
h323 interface
h323 qos ip-precedence 4
h323 h323-id [email protected]
h323 gatekeeper ipaddr 172.20.0.3
!
interface Ethernet0
ip address 172.20.0.1 255.255.255.192
no ip redirects
!
interface Serial0
no ip address
shutdown
!
interface Serial1
ip address 172.22.0.1 255.255.0.0
ip access-group 101 in
ip access-group 101 out
h323 asr
!
router rip
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network 172.20.0.0
network 10.0.0.0
!
router igrp 4000
network 172.22.0.0
network 101.0.0.0
!
ip classless
access-list 101 permit ip any host 10.0.0.1
access-list 101 permit ip host 10.0.0.1 any
access-list 101 permit igrp any any
!
!
line con 0
exec-timeout 0 0
line aux 0
transport input all
line vty 0 4
password lab
login
Defining Multiple Zones Example
The following example shows how to define multiple local zones for separating gateways:
zone
zone
zone
zone
zone
local gk408or650 xyz.com
local gk415 xyz.com
prefix gk408or650 408.......
prefix gk408or650 650.......
prefix gk415 415.......
All the gateways used for area codes 408 or 650 can be configured so that they register with gk408or650,
and all gateways used for area code 415 can be configured so that they register with gk415.
Defining One Zone for Multiple Gateways Example
The following example shows how to put all the gateways in the same zone and use the gw-priority
keyword to determine which gateways will be used for calling different area codes:
zone
zone
zone
zone
local localgk xyz.com
prefix localgk 408.......
prefix localgk 415....... gw-priority 10 gw1 gw2
prefix localgk 650....... gw-priority 0 gw1
The commands shown accomplish the following tasks:
•
Domain xyz.com is assigned to gatekeeper localgk.
•
Prefix 408....... is assigned to gatekeeper localgk, and no gateway priorities are defined for it;
therefore, all gateways that register to localgk can be used equally for calls to the 408 area code. No
special gateway lists are built for the 408....... prefix; selection is made from the master list for the
zone.
•
The prefix 415....... is added to gatekeeper localgk, and priority 10 is assigned to gateways gw1 and
gw2.
•
Prefix 650....... is added to gatekeeper localgk, and priority 0 is assigned to gateway gw1.
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A priority 0 is assigned to gateway gw1 to exclude it from the gateway pool for prefix 650........ When
gateway gw2 registers with gatekeeper localgk, it is added to the gateway pool for each prefix as follows:
•
For gateway pool for 415......., gateway gw2 is set to priority 10.
•
For gateway pool for 650......., gateway gw2 is set to priority 5.
To change gateway gw2 from priority 10 for zone 415....... to the default priority 5, enter the following
command:
no zone prefix localgk 415....... gw-pri 10 gw2
To change both gateways gw1 and gw2 from priority 10 for zone 415....... to the default priority 5, enter
the following command:
no zone prefix localgk 415....... gw-pri 10 gw1 gw2
In the preceding example, the prefix 415....... remains assigned to gatekeeper localgk. All gateways that
do not specify a priority level for this prefix are assigned a default priority of 5. To remove the prefix
and all associated gateways and priorities from this gatekeeper, enter the following command:
no zone prefix localgk 415.......
Configuring a Proxy for Inbound Calls Example
In the following example, the local zone sj.xyz.com is configured to use a proxy for inbound calls from
remote zones tokyo.xyz.com and milan.xyz.com to gateways in its local zone. The sj.xyz.com zone is
also configured to use a proxy for outbound calls from gateways in its local zone to remote zones
tokyo.xyz.com and milan.xyz.com.
gatekeeper
use-proxy sj.xyz.com
use-proxy sj.xyz.com
use-proxy sj.xyz.com
use-proxy sj.xyz.com
remote-zone
remote-zone
remote-zone
remote-zone
tokyo.xyz.com
tokyo.xyz.com
milan.xyz.com
milan.xyz.com
inbound-to gateway
outbound-from gateway
inbound-to gateway
outbound-from gateway
Because the default mode disables proxy communications for all gateway calls, only the gateway call
scenarios listed can use the proxy.
Configuring a Proxy for Outbound Calls Example
In the following example, the local zone sj.xyz.com uses a proxy for only those calls that are outbound
from H.323 terminals in its local zone to the specified remote zone germany.xyz.com:
gatekeeper
no use-proxy sj.xyz.com default outbound-from terminal
use-proxy sj.xyz.com remote-zone germany.xyz.com outbound-from
terminal
Note that any calls inbound to H.323 terminals in the local zone sj.xyz.com from the remote zone
germany.xyz.com use the proxy because the default applies.
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Removing a Proxy Example
The following example shows how to remove one or more proxy statements for the remote zone
germany.xyz.com from the proxy configuration list:
no use-proxy sj.xyz.com remote-zone germany.xyz.com
The command removes all special proxy configurations for the remote zone germany.xyz.com. After the
command is entered like this, all calls between the local zone (sj.xyz.com) and germany.xyz.com are
processed according to the defaults defined by any use-proxy commands that use the default option.
H.235 Security Example
The following example shows output from configuring secure registrations from the gatekeeper and
identifying which RAS messages the gatekeeper will check to find authentication tokens:
dial-peer voice 10 voip
destination-pattern 4088000
session target ras
dtmf-relay h245-alphanumeric
!
gateway
security password 09404F0B level endpoint
The following example shows output from configuring which RAS messages will contain gateway
generated tokens:
dialer-list 1 protocol ip permit
dialer-list 1 protocol ipx permit
radius-server host 10.0.0.1 auth-port 1645 acct-port 1646
radius-server retransmit 3
radius-server deadtime 5
radius-server key lab
radius-server vsa send accounting
!
gatekeeper
zone local GK1 test.com 10.0.0.3
zone remote GK2 test2.com 10.0.2.2 1719
accounting
security token required-for registration
no use-proxy GK1 remote-zone GK2 inbound-to terminal
no use-proxy GK1 remote-zone GK2 inbound-to gateway
no shutdown
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GKTMP and RAS Messages Example
The following is an example of a gatekeeper that has interaction with external applications. The
registration message from Server-123 establishes a connection with gatekeeper sj.xyz.com on port
20000. Server-123 sends a REGISTER RRQ message to gatekeeper sj.xyz.com to express interest in all
RRQs from voice gateways that support a technology prefix of 1# or 2#.
REGISTER RRQ
Version-id:1
From:Server-123
To:sj.xyz.com
Priority:2
Notification-Only:
Content-Length:29
t=voice-gateway
p=1#
p=2#
Prohibiting Proxy Use for Inbound Calls Example
To prohibit proxy use for inbound calls to H.323 terminals in a local zone from a specified remote zone,
enter a command similar to the following:
no use-proxy sj.xyz.com remote-zone germany.xyz.com inbound-to terminal
This command overrides the default and disables proxy use for inbound calls from remote zone
germany.xyz.com to all H.323 terminals in the local zone sj.xyz.com.
Disconnecting a Single Call Associated with an H.323 Gateway Example
The following example forces an active call on the H.323 gateway to be disconnected. The local ID
number of the active call is 12-3339.
Router> enable
Router# clear h323 gatekeeper call local-callID 12-3339
Disconnecting All Calls Associated with an H.323 Gateway Example
The following example forces all active calls on the H.323 gateway to be disconnected:
Router> enable
Router# clear h323 gatekeeper call all
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Configuring Session Initiation Protocol for Voice
over IP
This chapter introduces the Session Initiation Protocol (SIP). SIP is an alternative protocol developed by
the Internet Engineering Task Force (IETF) for multimedia conferencing over IP. SIP features are
compliant with IETF RFC 2543, SIP: Session Initiation Protocol, published in March 1999.
The Cisco SIP functionality enables Cisco access platforms to signal the setup of voice and multimedia
calls over IP networks. The SIP feature also provides nonproprietary advantages in the following areas:
•
Protocol extensibility
•
System scalability
•
Personal mobility services
•
Interoperability with different vendors
This chapter contains the following sections:
•
SIP Overview, page 366
•
How SIP Works, page 368
•
SIP Prerequisite Tasks, page 376
•
SIP Configuration Tasks List, page 376
•
SIP Configuration Examples, page 381
For a complete description of the commands used in this chapter, refer to the Cisco IOS Voice, Video,
and Fax Command Reference.
To identify the hardware platform or software image information associated with a feature in this
chapter, use the Feature Navigator on Cisco.com to search for information about the feature or refer to
the software release notes for a specific release. For more information, see the “Identifying Supported
Platforms” in the “Using Cisco IOS Software” chapter.
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SIP Overview
SIP Overview
SIP is an ASCII-based, application-layer control protocol that can be used to establish, maintain, and
terminate calls between two or more endpoints.
Like other Voice over IP protocols, SIP is designed to address the functions of signaling and session
management within a packet telephony network. Signaling allows call information to be carried across
network boundaries. Session management provides the ability to control the attributes of an end-to-end
call.
SIP provides the following capabilities:
Note
•
Determines the location of the target endpoint—SIP supports address resolution, name mapping,
and call redirection.
•
Determines the media capabilities of the target endpoint—Through Session Description Protocol
(SDP), SIP determines the lowest level of common services between the endpoints. Conferences are
established using only the media capabilities that can be supported by all endpoints.
•
Determines the availability of the target endpoint—If a call cannot be completed because the target
endpoint is unavailable, SIP determines whether the called party is connected to a call already or did
not answer in the allotted number of rings. SIP then returns a message indicating why the target
endpoint was unavailable.
•
Establishes a session between the originating and target endpoints—If the call can be completed,
SIP establishes a session between the endpoints. SIP also supports midcall changes, such as the
addition of another endpoint to the conference or the changing of a media characteristic or codec.
•
Handles the transfer and termination of calls—SIP supports the transfer of calls from one endpoint
to another. During a call transfer, SIP simply establishes a session between the transferee and a new
endpoint (specified by the transferring party) and terminates the session between the transferee and
the transferring party. At the end of a call, SIP terminates the sessions among all parties.
The term conference means an established session (or call) between two or more endpoints.
Conferences consist of two or more users and can be established using multicast or multiple unicast
sessions.
Components of SIP
SIP is a peer-to-peer protocol. The peers in a session are called user agents (UAs). A user agent can
function in one of the following roles:
•
User agent client (UAC)—A client application that initiates the SIP request.
•
User agent server (UAS)—A server application that contacts the user when a SIP request is received
and that returns a response on behalf of the user.
Typically, a SIP endpoint is capable of functioning as both a UAC and a UAS, but functions only as one
or the other per transaction. Whether the endpoint functions as a UAC or a UAS depends on the UA that
initiated the request.
From an architectural standpoint, the physical components of a SIP network can be grouped into two
categories: clients and servers. Figure 68 illustrates the architecture of a SIP network.
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SIP Overview
Figure 68
SIP Architecture
SIP proxy and
redirect servers
SIP
SIP
SIP
SIP user
agents (UAs)
SIP gateway
RTP
PSTN
Legacy PBX
Note
42870
IP
The SIP servers can interact with other application services, such as Lightweight Directory Access
Protocol (LDAP) servers, location servers, a database application, or an extensible markup language
(XML) application. These application services provide back-end services such as directory,
authentication, and billing services.
SIP Clients
SIP clients include the following:
•
Phones—Can act as either a UAS or UAC. SoftPhones (PCs that have phone capabilities installed)
and Cisco SIP IP phones can initiate SIP requests and respond to requests.
•
Gateways—Provide call control. Gateways provide many services, the most common being a
translation function between SIP conferencing endpoints and other terminal types. This function
includes translation between transmission formats and between communications procedures. In
addition, the gateway translates between audio and video codecs and performs call setup and
clearing on both the LAN side and the switched-circuit network side.
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How SIP Works
SIP Servers
SIP servers include the following:
•
Proxy server—Receives SIP messages and forwards them to the next SIP server in the network. The
proxy server is an intermediate device that receives SIP requests from a client and then forwards the
requests on behalf of the client. Proxy servers can provide functions such as authentication,
authorization, network access control, routing, reliable request retransmission, and security.
•
Redirect server—Provides the client with information about the next hop or hops that a message
should take. The client then contacts the next hop server or UAS directly.
•
Registrar server—Processes requests from UACs for registration of their current location. Registrar
servers are often located near a redirect or proxy server.
How SIP Works
SIP is a simple, ASCII-based protocol that uses requests and responses to establish communication
among the various components in a network and ultimately to establish a conference between two or
more endpoints.
Users in a SIP network are identified by unique SIP addresses. A SIP address is similar to an e-mail
address and is in the format of sip:[email protected] The user ID can be either a username or an
E.164 address.
Users register with a registrar server using their assigned SIP addresses. The registrar server then
provides the registration information to the location server upon request.
When a user initiates a call, a SIP request is sent to a SIP server (either a proxy or a redirect server). The
request includes the address of the caller (in the “from” header field) and the address of the intended
callee (in the “to” header field). The following sections provide simple examples of successful
point-to-point calls established using a proxy and a redirect server.
Over time, a SIP end user might move between end systems. The location of the end user can be
dynamically registered with the SIP server. The location server can use one or more protocols (including
finger, rwhois, and LDAP) to locate the end user. Because the end user can be logged in at more than one
station and because the location server can sometimes have inaccurate information, the SIP server might
return more than one address for the end user. If the request is coming through a SIP proxy server, the
proxy server will try each of the returned addresses until it locates the end user. If the request is coming
through a SIP redirect server, the redirect server forwards all the addresses to the caller in the “contact”
header field of the invitation response.
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How SIP Works
Using a Proxy Server
If a proxy server is used, the caller UA sends an INVITE request to the proxy server. The proxy server
determines the path and then forwards the request to the callee, as shown in Figure 69.
Figure 69
SIP Request Through a Proxy Server
Invite
IP-based
network
Client
Client
Invite
Server
Server
User agents
User agents
server
Proxy
Redirect
42871
Client
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The callee responds to the proxy server, which in turn forwards the response to the caller, as shown in
Figure 70.
Figure 70
SIP Response Through a Proxy Server
Response 200 OK
IP-based
network
Client
Client
Response 200 OK
Server
Server
User agents
User agents
Server
Proxy
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How SIP Works
The proxy server forwards the acknowledgments of both parties. A session is then established between
the caller and callee. Real-Time Transfer Protocol (RTP) is used for the communication between the
caller and the callee, as shown in Figure 71.
Figure 71
SIP Session Through a Proxy Server
IP-based
network
Ack
RTP
Client
Client
Ack
Server
Server
User agents
User agents
server
Proxy
Redirect
42873
Client
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Using a Redirect Server
If a redirect server is used, the caller UA sends an INVITE request to the redirect server. The redirect
server contacts the location server to determine the path to the callee, and the redirect server sends that
information back to the caller. The caller then acknowledges receipt of the information, as shown in
Figure 72.
Figure 72
SIP Request Through a Redirect Server
Invite
302 Moved temporarily
Ack
Server
Server
User agents
User agents
Proxy
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How SIP Works
The caller then sends a request to the device indicated in the redirection information (which could be the
callee or another server that will forward the request). Once the request reaches the callee, it sends back
a response, and the caller acknowledges the response. RTP is used for the communication between the
caller and the callee, as shown in Figure 73.
Figure 73
SIP Session Through a Redirect Server
Invite
200 OK
Ack
Client
Client
RTP
Server
Server
User agents
User agents
Proxy
Redirect
42875
IP-based
network
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How SIP Works
SIP Enhancements
SIP provides the following feature enhancements:
•
Ability to specify the maximum number of SIP redirects.
•
Ability to specify SIP or H.323 on a dial-peer basis.
•
Configurable SIP message timers and retries.
•
Interoperability with unified call services (UCS).
•
Support for a variety of signaling protocols, including ISDN, PRI, and channel associated signaling
(CAS).
•
Support for a variety of interfaces, including
– Analog interfaces: Foreign Exchange Station (FXS)/Foreign Exchange Office (FXO)/recEive
and transMit (E&M) analog interfaces.
– Digital interfaces: T1 CAS, T1 PRI, E1 CAS, E1 PRI, and E1 R2
•
Support for SIP redirection messages and interaction with SIP proxies. The gateway can redirect an
unanswered call to another SIP gateway or SIP-enabled IP phone. In addition, the gateway supports
proxy-routed calls.
•
Interoperability with DNS servers, including support for DNS SRV and “A” records to look up SIP
URLs according to RFC2052 formatting.
•
Support for SIP over TCP and User Datagram Protocol (UDP).
•
Support RTP/RTCP for media transport in VoIP networks.
•
Support for the following codecs:
– G711ulaw—0
– G711alaw—8
– G723r63—4
– G726r32—2
– G728—15
– G729r8—18
•
Support for record-route headers.
•
Support for IP quality of service (QoS) and IP precedence.
•
Support for IP Security (IPSec) for SIP signaling messages.
•
Authentication, authorization, and accounting (AAA) support. For accounting, the gateway device
generates call data record (CDR) accounting records for export. For authentication, the SIP gateway
sends validation requests to the AAA server. For authorization, the existing access lists are used.
•
Support for call hold and call transfer features. The call hold sends a midcall INVITE message,
which requests that the remote endpoint stop sending media streams. The call transfer is done
without consultation (blind transfer). The transfer can be initiated by a remote SIP endpoint.
•
Support for configurable expiration time for SIP INVITEs and maximum number of proxies or
redirect servers that can forward a SIP request.
•
Ability to hide the identity of the calling party by setting the ISDN presentation indicator.
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SIP Restrictions and Considerations
Before configuring your router (Cisco 2600, Cisco 3600, or Cisco AS5300) with the SIP feature, you
should note the following restrictions and considerations:
•
The SIP gateway does not support codecs other than those listed in the section, “SIP Enhancements.”
•
SIP requires that all times be sent in Greenwich Mean Time (GMT). The INVITE is sent in GMT.
However, the default for routers is to use Coordinated Universal Time (UTC). To configure the
router to use GMT, issue the clock timezone command in global configuration mode and specify
GMT.
•
With call transfer, the Requested-By header identifies the party initiating the transfer. The
Requested-By header is included in the INVITE request that is sent to the transferred-to party only
if a Requested-By header was also included in the Bye request.
•
With call transfer, the Also header identifies the transferred-to party. To invoke a transfer, the user
portion of the Also header must be defined explicitly or with wildcards as a destination pattern on a
VoIP dial peer. The transferred call is routed using the session target parameter on the dial peer
instead of the host portion of the Also header. Therefore, the Also header can contain [email protected],
but the host portion is ignored for call routing purposes.
•
The grammar for the Also and Requested-By headers is not fully supported. Only the name-addr is
supported. This implies that the crypto-param, which might be present in the Bye request, will not
be populated in the ensuing Invite to the transferred-to party.
•
Cisco SIP gateways do not support the “user=np-queried” parameter in a Request URI.
•
If a Cisco SIP gateway receives an ISDN Progress message, it generates a 183 Session progress
message. If the gateway receives an ISDN ALERT, it generates a 180 Ringing message.
•
SIP supports plain old telephone service (POTS)-to-POTS hairpinning (which means that the call
comes in one voice port and is routed out another voice port). It also supports POTS-to-IP call legs
and IP-to-POTS call legs. However, it does not support IP-to-IP hairpinning. This means that the SIP
gateway cannot take an inbound SIP call and reroute it back to another SIP device using the VoIP
dial peers.
•
The SIP gateway requires each INVITE to include a Session Description Protocol (SDP) header.
•
The contents of the SDP header cannot change between the 180 Ringing message and the 200 OK
message.
•
VoIP dial peers allow a user to configure the bytes parameter associated with a codec. Cisco SIP
gateways present or respond to the a=ptime parameter in the SDP body of a SIP message. However,
only one a=ptime attribute is allowed per m-line block.
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SIP Prerequisite Tasks
SIP Prerequisite Tasks
Before you configure your router with the SIP feature, you must perform the following tasks:
•
Configure your gateway to support voice functionality for SIP or H.323.
•
Establish a working IP network.
For more information about configuring IP, refer to the Cisco IOS IP Configuration Guide.
•
Configure VoIP.
•
Ensure that your Cisco 2600 or Cisco 3600 series router has16 MB Flash and 64 MB DRAM
memory, minimum. A Cisco AS5300 must have 16 MB Flash and 64 MB DRAM memory,
minimum.
SIP Configuration Tasks List
To configure SIP functions on the Cisco AS5300, Cisco 2600, or the Cisco 3600 series router, perform
the following tasks:
•
Configuring SIP Support for VoIP Dial Peers, page 376
•
Changing the Configuration of the SIP User Agent, page 377 (Optional)
•
Configuring SIP Call Transfer, page 378 (Optional)
•
Configuring Gateway Accounting, page 379 (Optional)
For more information on SIP configuration, including call flows, refer to the document Session
Initiation Protocol Gateway Call Flows, Version 2 in Cisco IOS Release 12.1(3)T found on Cisco.com.
Configuring SIP Support for VoIP Dial Peers
To configure SIP support for a VoIP dial peer, use the following commands beginning in global
configuration mode:
Command
Purpose
Step 1
Router(config)# dial-peer voice number
voip
Enters dial-peer configuration mode to configure a VoIP dial peer.
Step 2
Router(config-dial-peer)# session
transport {udp | tcp}
Enters the session transport type for the SIP user agent. The default
is udp.
The transport protocol (udp or tcp) specified with the session
transport command must be identical to the protocol specified with
the transport command.
Step 3
Router(config-dial-peer)# session
protocol {cisco | sipv2}
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Enters the session protocol type. The keywords are as follows:
•
cisco—Configures the dial peer to use proprietary CiscoVoIP
session protocol.
•
sipv2—Configures the dial peer to use IETF SIP. SIP users
should use this option.
Configuring Session Initiation Protocol for Voice over IP
SIP Configuration Tasks List
Step 4
Step 5
Command
Purpose
Router(config-sip-ua)# sip-server
{dns:[hostname] |
ipv4:ip_addr:[port-num]}
Enters the host name or IP address of the SIP server interface. If you
use this command, you can then specify session target sip-server
for each dial peer instead of repeatedly entering the SIP server
interface address for each dial peer. The keywords and arguments
are as follows:
Router(config-dial-peer)# session target
{sip-server | dns:[$s$. | $d$. | $e$. |
$u$. [hostname]| ipv4:ip_addr:[port-num]}
•
dns:hostname—Sets the global SIP server interface to a domain
name server (DNS) host name. A valid DNS host name takes
the following format: name.gateway.xyz.
•
ipv4:ip_addr:—Sets the IP address.
•
portnum (Optional)—Sets the UDP port number for the SIP
server.
Specifies a network-specific address for a dial peer. The keywords
and arguments are as follows:
•
sip-server— Sets the session target to the global SIP server.
Used when the sip-server command has already specified the
host name or IP address of the SIP server interface.
•
dns:hostname—Sets the global SIP server interface to a domain
name server (DNS) host name. A valid DNS host name takes
the following format: name.gateway.xyz.
•
ipv4:ip_addr:—Sets the IP address.
•
portnum—(Optional) Sets the UDP port number for the SIP
server.
Note
Wildcards can be used when defining the session target for
VoIP peers.
Changing the Configuration of the SIP User Agent
It is not necessary to configure a SIP user agent (UA) in order to place a call. A SIP UA is configured to
listen by default. However, if you want to adjust any of the settings, use the following commands
beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# sip-ua
Enters the SIP user agent (sip-ua) configuration mode to configure
SIP-UA related commands.
Step 2
Router(config-sip-ua)# transport
{udp | tcp}
Configures the SIP user agent (sip-ua) for SIP signaling messages.
The default is udp.
The transport protocol (udp or tcp) specified with the session
transport command must be identical to the protocol specified with
the transport command.
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SIP Configuration Tasks List
Step 3
Step 4
Command
Purpose
Router(config-sip-ua)# timers {trying
number | connect number | disconnect
number | expires number}
(Optional) Configures the SIP signaling timers. The keywords are as
follows:
•
trying—Sets the time to wait for a 100 response to an INVITE
request. The default is 500.
•
connect—Sets the time to wait for a 200 response to an ACK
request. The default is 500.
•
disconnect—Sets the time to wait for a 200 response to a BYE
request. The default is 500.
•
expires—Limits the time duration (in milliseconds) for which
an INVITE is valid. The default is 180000.
(Optional) Configures the SIP signaling timers for retry attempts.
The keywords are as follows:
Router(config-sip-ua)# retry {invite
number | response number | bye number |
cancel number}
•
invite—Number of INVITE retries. The default is 6.
•
response—Number of RESPONSE retries. The default is 6.
•
bye—Number of BYE retries. The default is 10.
•
cancel—Number of Cancel retries. The default is 10.
Step 5
Router(config-sip-ua)# max-forwards
number
(Optional) Limits the number of proxy or redirect servers that can
forward a request. The default is 6.
Step 6
Router(config-sip-ua)# max-redirects
number
(Optional) Sets the maximum number of redirect servers. The default
is 1.
Step 7
Router(config-sip-ua)# default
{max-forwards | retry {invite | response
| bye | cancel} | sip-server | timers
{trying | connect | disconnect |
expires} | transport}
(Optional) Resets the value of a SIP user agent command to its
default.
Configuring SIP Call Transfer
To configure SIP call transfer for a POTS dial peer, use the following commands beginning in global
configuration mode:
Command
Purpose
Step 1
Router(config)# dial-peer voice number
pots
Enters dial-peer configuration mode to configure a POTS dial peer.
Step 2
Router(config-dial-peer)# application
session
Specifies that the standard session application will be invoked for this
dial peer.
Step 3
Router(config-dial-peer)#
destination-pattern pattern
Specifies the telephone number associated with the dial peer.
Step 4
Router(config-dial-peer)# port
{slot-number/subunit-number/port} |
{slot/port:ds0-group-no}
(Cisco 2600 and Cisco 3600 series routers) Specifies the local voice
port through which incoming VoIP calls will be received.
Step 5
Router(config-dial-peer)# port
{controller number:D}
(Cisco AS5300 universal access server) Specifies the local voice port
through which incoming VoIP calls will be received.
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SIP Configuration Tasks List
To configure SIP call transfer for a VoIP dial peer, use the following commands beginning in global
configuration mode:
Command
Purpose
Step 1
Router(config)# dial-peer voice number
voip
Enters the dial-peer mode to configure a VoIP dial peer.
Step 2
Router(config-dial-peer)# application
session
Specifies that the standard session application will be invoked for this
dial peer.
Step 3
Router(config-dial-peer)#
destination-pattern pattern
Specifies the telephone number associated with the dial peer.
Step 4
Router(config-dial-peer)# session
target ipv4:x.x.x.x
Specifies the IP address of the destination gateway for outbound dial
peers.
Note
For information about the commands used to configure translation rules, see the “Configuring Dial
Plans, Dial Peers, and Digit Manipulation” chapter.
Configuring Gateway Accounting
There are three keywords that configure gateway accounting for SIP:
•
The voip keyword sends the call data record (CDR) to the RADIUS server. Use this keyword with
the SIP feature.
•
The H323 keyword sends the call data record (CDR) to the RADIUS server.
•
The syslog keyword uses the system logging facility to record the CDRs.
To enable gateway-specific accounting for SIP, use the following command in global configuration
mode:
Command
Purpose
Router(config)# gw-accounting {voip |
syslog | h323 [syslog]}
(Optional) Enables gateway-specific accounting in global configuration
mode.
For general accounting information, refer to the Cisco IOS Security Configuration Guide.
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SIP Configuration Tasks List
Verifying SIP Configuration
Enter the show running-config command to verify your configuration, or use the show sip-ua command
to verify the SIP configurations.
The following example shows sample output for the show sip-ua statistics command:
Router# show sip-ua statistics
SIP Response Statistics (Inbound/Outbound)
Informational:
Trying 0/0, Ringing 0/0,
Forwarded 0/0, Queued 0/0,
SessionProgress 0/0
Success:
OkInvite 0/0, OkBye 0/0,
OkCancel 0/0, OkOptions 0/0
Redirection (Inbound only):
MultipleChoice 0, MovedPermanently 0,
MovedTemporarily 0, SeeOther 0,
UseProxy 0, AlternateService 0
Client Error:
BadRequest 0/0, Unauthorized 0/0,
PaymentRequired 0/0, Forbidden 0/0,
NotFound 0/0, MethodNotAllowed 0/0,
NotAcceptable 0/0, ProxyAuthReqd 0/0,
ReqTimeout 0/0, Conflict 0/0, Gone 0/0,
LengthRequired 0/0, ReqEntityTooLarge 0/0,
ReqURITooLarge 0/0, UnsupportedMediaType 0/0,
BadExtension 0/0, TempNotAvailable 0/0,
CallLegNonExistent 0/0, LoopDetected 0/0,
TooManyHops 0/0, AddrIncomplete 0/0,
Ambiguous 0/0, BusyHere 0/0
Server Error:
InternalError 0/0, NotImplemented 0/0,
BadGateway 0/0, ServiceUnavail 0/0,
GatewayTimeout 0/0, BadSipVer 0/0
Global Failure:
BusyEverywhere 0/0, Decline 0/0,
NoExistAnywhere 0/0, NotAcceptable 0/0
SIP Total Traffic Statistics (Inbound/Outbound)
Invite 0/0, Ack 0/0, Bye 0/0,
Cancel 0/0, Options 0/0
Retry Statistics
Invite 0, Bye 0, Cancel 0, Response 0
The following example shows sample output for the show sip-ua status command:
Router# show sip-ua status
SIP
SIP
SIP
SIP
User Agent Status
User Agent for UDP : ENABLED
User Agent for TCP : ENABLED
max-forwards :6
The following example shows sample output for the show sip-ua timers command:
Router# show sip-ua timers
SIP UA Timer Values (millisecs)
trying 500, expires 180000, connect 500, disconnect 500
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SIP Configuration Examples
SIP Configuration Examples
The following shows a basic SIP configuration. This output was created by using the
show running-config command.
Router1# show running-config
Building configuration...
Current configuration:
!
version 12.2
service timestamps debug datetime
service timestamps log uptime
no service password-encryption
!
hostname router1
!
!
!
clock timezone GMT 5
voice-card 1
!
ip subnet-zero
ip tcp path-mtu-discovery
ip name-server 172.18.192.48
!
isdn voice-call-failure 0
!
!
controller T1 1/0
framing esf
clock source line primary
linecode b8zs
!
controller T1 1/1
!
!
voice-port 2/0/0
!
voice-port 2/0/1
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g723r63
codec preference 3 g723r53
!
!
dial-peer voice 100 pots
destination-pattern 3660110
port 2/0/0
!
dial-peer voice 200 pots
application session
destination-pattern 3660120
port 2/0/1
!
dial-peer voice 101 voip
destination-pattern 3660210
session protocol sipv2
session target ipv4:172.16.244.73
codec g711ulaw
!
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SIP Configuration Examples
dial-peer voice 201 voip
application session
destination-pattern 3660220
session protocol sipv2
session target dns:3660-2.sip.com
codec g711ulaw
!
dial-peer voice 999 voip
destination-pattern 5551111
session protocol sipv2
session target ipv4:172.20.53.89
session transport tcp
!
dial-peer voice 300 pots
destination-pattern 2101100
!
dial-peer voice 350 voip
destination-pattern 3100607
session protocol sipv2
session target ipv4:172.18.192.197
codec g711ulaw
!
dial-peer voice 301 voip
application session
destination-pattern 1234
session protocol sipv2
session target ipv4:172.18.192.193
codec g711ulaw
!
dial-peer voice 333 voip
application session
destination-pattern 1235
session protocol sipv2
session target ipv4:172.18.192.199
codec g711ulaw
!
dial-peer voice 888 voip
destination-pattern 888
session protocol sipv2
session target ipv4:172.20.53.89
session transport tcp
codec g711ulaw
!
dial-peer voice 260011 voip
destination-pattern 260011
session protocol sipv2
session target ipv4:172.18.192.164
codec g711ulaw
!
dial-peer voice 444 voip
destination-pattern 2339000
session protocol sipv2
session target ipv4:172.18.192.205
codec g711ulaw
!
dial-peer voice 111 voip
destination-pattern 111
session protocol sipv2
session target sip-server
codec g711ulaw
!
dial-peer voice 7777777 voip
destination-pattern 19197777777
session protocol sipv2
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SIP Configuration Examples
session target ipv4:172.18.192.38
codec g711ulaw
!
!
sip-ua
retry invite 2
retry response 2
retry bye 2
retry cancel 2
no inband-alerting
sip-server dns:server
!
!
interface FastEthernet0/0
ip address 172.18.192.194 255.255.255.0
load-interval 30
speed auto
half-duplex
!
interface FastEthernet0/1
ip address 172.16.245.230 255.255.255.224
load-interval 30
speed auto
half-duplex
!
ip classless
ip route 0.0.0.0 0.0.0.0 172.18.192.1
ip route 172.16.0.0 255.255.0.0 172.16.245.225
no ip http server
!
access-list 101 permit ip host 10.0.2.30 host 10.0.2.31
access-list 101 deny
udp any eq rip any
access-list 101 deny
udp any any eq rip
access-list 101 deny
udp any eq isakmp any
access-list 101 deny
udp any any eq isakmp
access-list 101 permit ip any any
snmp-server engineID local 000000090200003094202740
snmp-server community public RW
!
line con 0
exec-timeout 0 0
transport input none
line aux 0
line vty 0 4
password xxx
login
!
end
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Voice over Layer 2 Protocols
Configuring Voice over Frame Relay
This chapter describes the configuration of Voice over Frame Relay (VoFR) and contains the following
sections:
•
VoFR Overview, page 387
•
VoFR Prerequisite Tasks, page 393
•
VoFR Configuration Task List, page 393
•
VoFR Configuration Examples, page 409
For a description of the VoFR configuration commands using the FRF.11 implementation agreement,
refer to the Cisco IOS Voice, Video, and Fax Command Reference. For additional information about the
FRF.12 implementation agreement and wide-area networks (WANs), refer to the Cisco IOS Wide-Area
Networking Configuration Guide and Cisco IOS Wide-Area Networking Command Reference. For
information about voice port configurations, refer to the “Configuring Voice Ports” chapter.
To identify the hardware platform or software image information associated with a feature in this
chapter, use the Feature Navigator on Cisco.com to search for information about the feature or refer to
the software release notes for a specific release. For more information, see the “Identifying Supported
Platforms” in the “Using Cisco IOS Software” chapter.
VoFR Overview
VoFR enables a router to carry voice traffic (for example, telephone calls and faxes) over a Frame Relay
network, using the FRF.11 protocol. This specification defines multiplexed data, voice, fax, dual tone
multi frequency (DTMF) digit-relay, and channel-associated signaling (CAS)/robbed-bit signaling
frame formats. The Frame Relay backbone must be configured to include the map class and Local
Management Interface (LMI).
The Cisco VoFR implementation enables dynamic- and tandem-switched calls and Cisco trunk calls.
Dynamic-switched calls have dial-plan information included that processes and routes calls based on the
telephone numbers. The dial-plan information is contained within dial-peer entries. For more
information, see “Switched Calls” section on page 389.
Tandem-switched calls are switched from incoming VoFR to an outgoing VoFR enabled data-link
connection identifier (DLCI) and tandem nodes enable the process. The nodes also switch Cisco trunk
calls.
Permanent calls are processed over Cisco private-line trunks and static FRF.11 trunks that specify the
frame format and coder types for voice traffic over a Frame Relay network. For more information, see
“Permanent Calls” section on page 390.
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VoFR Overview
VoFR connections depend on the hardware platform and type of call. The types of calls are:
Note
•
Switched (user dialed or auto-ringdown and tandem)
•
Permanent (Cisco trunk or static FRF.11 trunk)
Calls to Cisco MC3810 multiservice concentrators running Cisco IOS releases before 12.0(7)XK and
12.1(2)T require specific procedures for VoFR configuration and are described in separate sections.
VoFR Dial Peers
Dial peers are addressable call endpoints that identify the origin and destination of a call. Dial peers
define the characteristics applied to each call leg in the call connection. A call leg is a logical connection
between two routers or between a router and a telephony device.
A traditional voice call over the Public Switched Telephone Network (PSTN) uses a dedicated 64K
circuit end-to-end. In contrast, a voice call over the packet network is made up of call legs. A voice call
has four call legs, two from the perspective of the originating router and two from the perspective of the
destination router, as shown in Figure 74.
Dial Peer Call Legs
Source
Destination
IP network
V
Call leg 1
(POTS dial peer)
Call leg 2
(VoIP dial peer)
V
Call leg 3
(VoIP dial peer)
35950
Figure 74
Call leg 4
(POTS dial peer)
A dial peer is associated with each call leg. Attributes that are defined in a dial peer and applied to the
call leg include codec, Quality of Service (QoS), voice activity detection (VAD), and fax rate. To
complete a voice call, you must configure a dial peer for each of the four call legs in the call connection.
Two kinds of dial peers are possible in VoFR configurations:
•
POTS—Dial peer describing the characteristics of a traditional telephony network connection.
POTS dial peers map a dialed string to a specific voice port on the local router, normally the voice
port connecting the router to the local PSTN, PBX, or telephone.
•
VoFR—Dial peer that is connected between a Frame Relay WAN backbone and a specific
voice-network device. VoFR dial peers map a dialed string to the destination router.
VoFR peers point to specific voice-network devices by associating destination telephone numbers with
a specific Frame Relay DLCI so that outgoing calls can be placed. Both POTS and VoFR dial peers are
needed to establish VoFR connections if the sending and receiving of calls are required.
Understanding the the relationship between the destination pattern and the session target is critical to
understanding VoFR dial peers. The destination pattern is the telephone number of the voice device
attached to the voice port. The session target defines the route to a serial port on the peer router at the
other end of the Frame Relay connection.
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Note
For tandem voice nodes, POTS dial peers are not configured.
For additional information on POTS dial peers, see the “Configuring Dial Plans, Dial Peers, and Digit
Manipulation” chapter.
Switched Calls
The Cisco-switched VoFR protocol handles call setup and parameter negotiation for both endpoints and
intermediate nodes within the multihop call path. The call setup mechanism originally implemented in
the Cisco MC3810 multiservice concentrator can be used for permanent-switched (Cisco trunk) or
dynamic-switched calls. The Cisco VoFR protocol includes forwarding of the called telephone number
and supports tandem switching of the call over multiple Frame Relay permanent virtual connection
(PVC) hops.
Cisco addresses the lack of end-to-end call parameter negotiation and call setup syntax in FRF.11 by
implementing a proprietary Q.931-like session protocol running on a user-configurable channel ID
(CID) of an FRF.11-format multiplexed DLCI.
Tandem Switching
Dynamic switching of voice calls between VoFR or VoATM PVCs and subchannels is also called tandem
switching (often encountered in multihop VoFR call connection paths). Tandem switching uses nodes
that are intermediate router nodes within the Frame Relay call path.
Each node switches the frames from one PVC subchannel to another (from one VoFR dial peer to another
VoFR dial peer) as the frames traverse the network. Use of tandem router nodes avoids the need to have
complete dial-plan information present on every router.
Dynamic-Switched Calls
Dynamic-switched calls are regular telephone calls in which the switching is performed by the Cisco
router. The destination endpoint of the call is selected by the router based on the dialed telephone number
and the dial peer configuration entries. This implementation is different from permanent calls (Cisco
trunk calls) in which the call endpoints are permanently fixed at configuration time. The dial peer uses
the Cisco proprietary session protocol.
Cisco Trunk Calls
A Cisco trunk call is a dynamic-switched call of indefinite duration that uses a fixed-destination
telephone number and includes optional transparent end-to-end signaling. The telephone number of the
destination endpoint is permanently configured into the router so that it always selects a fixed
destination. Once established, at boot-up or when configured, the call stays up until one of the voice
ports or network ports is shut down or until a network disruption occurs. The dial peer is configured to
invoke the Cisco proprietary session protocol.
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Permanent Calls
Permanent calls are transmitted and received on FRF.11 and Cisco trunks. FRF.11 trunk interoperability
for standards-based vendors enables specification of the frame format and coder types to be used when
sending voice traffic through a Frame Relay network. However, FRF.11 does not have specifications for
end-to-end negotiation, call setup process, or any other form of communication between the Frame
Relay nodes.
As a result, static FRF.11 trunks are set up by manually configuring each router within the voice trunk
path with compatible parameters: a voice port and a specific subchannel on a DLCI are explicitly bound
on each end router. Signaling information is packed and sent transparently end-to-end.
The two ends of an FRF.11 call must use the same compatible speech compression codecs. If not, the
call exists and voice packets are sent and received, but no usable voice path is created.
When configured, a static FRF.11 trunk remains up until the voice or serial port is shut down or until a
network disruption occurs. The FRF.11 specification does not include any standardized methods for
performing Operation, Administration, and Maintenance (OAM) functions. There is no standard
protocol for detecting faults and providing rerouting of connection paths.
FRF.11 enables up to 255 subchannels to be multiplexed onto a single Frame Relay DLCI. The current
implementation supports the multiplexing of a single data channel with many voice channels. However,
subchannels from zero to three are reserved and cannot be configured for voice or data.
Frame Relay Fragmentation
Cisco has developed three methods of performing Frame Relay fragmentation that are described in the
following sections:
•
End-to-End FRF.12 Fragmentation, page 391
•
Frame Relay Fragmentation Using FRF.11 Annex C, page 392
•
Cisco Proprietary Voice Encapsulation, page 392
FRF.11 can only be used when an end-to-end PVC is available between the voice ports at each end of
the connection. At intermediate Frame Relay nodes, the entire PVC must be routed. Because the entire
PVC is routed, no prioritization of voice packets is possible at the intermediate Frame Relay. Connection
ID-based routing (individual channel-ID switching) is not supported.
FRF.11 specifies that a device can pack multiple FRF.11 subframes within a single Frame Relay frame;
however, the Cisco implementation of VoFR currently does not support multiple subframes within a
frame. VoFR frames are never fragmented, regardless of size. If fragments arrive out of sequence,
packets are dropped. Fragmentation is performed after frames are removed from the weighted fair
queuing (WFQ). WFQ at the PVC level is the only queueing strategy that can be used.
Frame Relay Traffic Shaping (FRTS) must be configured to enable Frame Relay fragmentation.
Frame Relay fragmentation can be configured in conjunction with VoFR or independently of it. For
additional information regarding FRF.12 fragmentation and the implementation commands, refer to the
Cisco IOS Wide-Area Networking Configuration Guide and Cisco IOS Wide-Area Networking Command
Reference.
VoFR provides support for various FRF.11 features depending on the hardware platform used (see
Table 27).
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Table 27
FRF.11 Forum Features Supported by Hardware Platform
Cisco 2600/3600
Series Routers
Cisco 7500 Series
Routers with VIP
Support
Class 1–Compliance Requirements (sec. 4.1) Not supported
Not supported
Not supported
Class 2–Compliance Requirements (sec. 4.2) Supported
Supported
Supported
Annex A–Dialed Digits Transfer Syntax
Supported
Supported
Supported
Annex B–Signaling Bit Transfer Syntax
Supported
Supported
Supported
Annex C–Data Transfer Syntax
Supported
Supported
Supported
Annex D–Fax Relay Transfer Syntax
Supported
Supported
Supported
FRF.11 Forum Features
Cisco MC3810
Multiservice
Concentrator
Annex E–CS-ACELP Transfer Syntax
(G.729/G.729A)
•
Sequence Number
Supported
Supported
Supported
•
Packing Factor
Supported
Supported
Supported
Annex F–Generic PCM/ADPCM Voice
Transfer Syntax
Supported
Supported
Supported
Annex G –G.727 Discard-Eligible
E-ADPCM Voice Transfer Syntax
Not supported
Not supported
Not supported
Annex H–G.728 LD-CELP Transfer Syntax Not supported
Supported
Supported
Annex I–G.723.1 Dual Rate Speech Coder
Not supported
Supported
Supported
Transmission and reception of multiple
subframes within a single Frame Relay
frame
Not supported
Not supported
Not supported
End-to-End FRF.12 Fragmentation
FRF.12 fragmentation is defined by the FRF.12 standard. The FRF.12 implementation agreement enables
long data frames to be fragmented into smaller pieces and interleaved with real-time frames. In this way,
real-time voice and nonreal-time data frames can be carried together on lower-speed links without
causing excessive delay to the real-time traffic.
Use this fragmentation type when the PVC is not carrying voice, but is sharing the link with other PVCs
that are carrying voice. The fragmentation header is included only for frames that are greater than the
fragment size configured. FRF.12 is the recommended fragmentation for VoIP packets.
Note
VoIP packets should not be fragmented. However, VoIP packets can be interleaved with fragmented
packets.
The Cisco 2600 series, 3600 series, and 7200 series routers and the Cisco MC3810 multiservice
concentrator support end-to-end fragmentation on a per-PVC basis. Fragmentation is configured through
a map class that applies to one or many PVCs, depending on how the class is applied.
When end-to-end FRF.12 fragmentation is used, the VoIP packets do not include the FRF.12 header,
provided the size of the VoIP packet is smaller than the fragment size configured. However, when FRF.11
Annex C or Cisco proprietary fragmentations are used, VoIP packets do include the fragmentation
header.
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Configuring Voice over Frame Relay
VoFR Overview
Frame Relay Fragmentation Using FRF.11 Annex C
When VoFR and fragmentation are configured on a PVC, the Frame Relay fragments are sent in the
FRF.11 Annex C format. FRF.11 fragmentation is used when voice traffic is sent on the PVC, and Annex
C format is used for data. With FRF.11, all data packets contain fragmentation headers, regardless of
size. This form of fragmentation is not recommended for use with VoIP.
Cisco Proprietary Voice Encapsulation
Cisco proprietary voice encapsulation was implemented for the Cisco MC3810 multiservice
concentrator and was used for data packets on a PVC and voice traffic. This fragmentation type is used
on data packets on PVCs that carry voice traffic.
When VoFR is configured on a DLCI and fragmentation is enabled on a map class, the Cisco 7500 series
router with Versatile Interface Processor (VIP) can interoperate with Cisco 2600 series, 3600 series,
7200 series, and other 7500 series routers as tandem nodes, but it cannot perform call termination with
Cisco MC3810 multiservice concentrators running Cisco IOS releases before 12.0(3)XG or 12.0(4)T.
Map Classes and Voice Packet Queues
You must create and configure a Frame Relay map class before configuring a Frame Relay DLCI for
voice traffic. The map class has configuration information about voice bandwidth, fragmentation size,
and traffic shaping attributes. These attributes are required for sending voice traffic on the PVC.
Traffic Shaping
When a Frame Relay PVC is configured to support voice traffic, the carrier must be able to accommodate
the traffic rate or profile sent on the PVC. If too much traffic is sent at once, the carrier might discard
frames causing disruptions to real-time voice traffic. The carrier might also deal with traffic bursts by
queueing up the bursts and delivering them at a metered rate. Excessive queueing also causes disruption
to real-time voice traffic. Traffic shaping compensates for this condition and is necessary to prevent the
carrier from discarding eligible discard bits on ingress and to prevent excessive burst data from affecting
voice quality.
When the outgoing Excess Burst (Be) size is configured, the Committed Burst (Bc) size and the
committed information rate (CIR) values must be obtained from the carrier. The configured values on
the router must match those of the carrier.
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Configuring Voice over Frame Relay
VoFR Prerequisite Tasks
VoFR Prerequisite Tasks
Before configuring the router for VoFR, perform the following tasks:
•
Complete the company dial plan and establish a working telephony network based on the dial plan:
– Integrate the dial plan and telephony network into the existing Frame Relay network topology.
Make routing or dialing transparent to the user; for example, avoid secondary dial tones from
secondary switches, where possible.
– Contact the PBX vendor for instructions on how to reconfigure the appropriate PBX interfaces.
•
Establish a working IP and Frame Relay network. For more information about configuring IP, see
the “IP Overview,” “Configuring IP Addressing,” and “Configuring IP Services” chapters in the
Cisco IOS IP Configuration Guide. For more information about configuring Frame Relay, see the
Cisco IOS Wide-Area Networking Configuration Guide.
•
Configure the required codecs and POTs dial peer configurations in “Configuring Dial Peers, Dial
Plans, and Digit Manipulation” chapter.
•
Configure voice ports. For more information, see the “Configuring Voice Ports” chapter.
•
Configure the clock source interfaces. For more information, refer to the “Configuring Synchronous
Clocking” appendix.
VoFR Configuration Task List
This section describes the following tasks:
•
Configuring Frame Relay to Support Voice, page 393
•
Configuring VoFR Dial Peers, page 395
•
Configuring Switched Calls, page 400
•
Configuring Cisco Trunk Calls, page 404
For information regarding the configuring of voice ports and dial peers, refer to the “Configuring Voice
Ports” and “Configuring Voice Dial Peers, Dial Plans, and Digit Manipulation” chapters.
Configuring Frame Relay to Support Voice
To configure Frame Relay to support voice, a map class must be applied to a single DLCI or to a group
of DLCIs, depending on how the class has been applied to the virtual circuit. If there is a large number
of PVCs to configure, assign the same traffic-shaping properties to the PVCs. The values for each PVC
are not statically defined. Multiple map classes with different variables for each map class can also be
created.
When the frame-relay voice bandwidth command is entered, a special queue is created for voice
packets only so that time-sensitive voice packets have preference over data packets.
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Configuring Voice over Frame Relay
VoFR Configuration Task List
This section describes the configuration of map classes as follows:
•
Configuring a Map Class to Support Voice Traffic, page 394
•
Configuring a Map Class for Traffic-Shaping Parameters, page 395
To configure the map class to support FRF.12 fragmentation, refer to the Cisco IOS Wide-Area
Networking Configuration Guide and Command Reference for more information.
Configuring a Map Class to Support Voice Traffic
When you are configuring a Frame Relay map class to support voice traffic, you must reserve the
appropriate amount of voice bandwidth. If there is not enough bandwidth reserved, new calls are
rejected. When calculating the amount of required voice bandwidth, include the voice packetization
overhead and not just the raw compressed speech codec bandwidth.
Remember that there are a six or seven bytes of total overhead per voice packet, including standard
Frame Relay headers and flags. For subchannels (CIDs) numbered less than 64, the overhead is 6 bytes.
For subchannels numbered greater than or equal to 64, the overhead is 7 bytes. Add one byte if voice
sequence numbers are enabled in the voice packets.
To determine the required voice bandwidth, use the following calculation:
required_bandwidth = codec_bandwidth * (1 + overhead/payload_size)
This calculation addresses the amount of bandwidth consumed on the physical network interface. The
figure does not necessarily represent the amount of connection bandwidth used within the Frame Relay
network itself, which may be higher because the overhead of switching small packets.
When 30-ms duration voice packets are used, an approximate general rule is to add 2000 bps overhead
to the raw voice compressed speech codec rate. With the 32 kbps G.726 adaptive differential pulse code
modulation (ADPCM) speech coder, a 30-ms speech frame uses 120 bytes voice payload plus 6 to 7 bytes
overhead, and the overall bandwidth requirement is about 34 kbps for each call.
To configure a Frame Relay map class to support voice traffic on DLCIs, use the following commands
beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# map-class frame-relay map-class-name
Creates a map class name to assign to a group of
PVCs and enters map-class configuration mode. A
map class name must be unique.
Step 2
Router(config-map-class)# frame-relay voice
bandwidth bps_reserved
Enters the bandwidth in bits per second (bps) and
determines the number of voice calls enabled on the
DLCIs where the map class is associated. The
keywords and arguments are as follows:
•
Note
It is recommended that the bps be no higher than the minimum CIR if the voice quality is impacted
when burst is being sent.
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bps_reserved—Reserved bandwidth. Valid range
is from 8,000 to 45,000,000 bps. The default is 0
(disables all voice calls).
Configuring Voice over Frame Relay
VoFR Configuration Task List
Configuring a Map Class for Traffic-Shaping Parameters
To configure a Frame Relay map class for the traffic shaping parameters for one or more DLCIs, use the
following commands in map-class configuration mode:
Command
Purpose
Step 1
Router(config-map-class)# frame-relay bc out bits
Configures the outgoing bc size for this group of
PVCs. Configure the bits value to a minimum of 1000
for voice traffic. Ensure that the bc size matches the
carrier to prevent the carrier from discarding DE bits
on ingress.
Step 2
Router(config-map-class)# frame-relay be out bits
Configures the outgoing be size for this group of
PVCs. Ensure that the Excess Burst size matches the
carrier to prevent the carrier from discarding DE bits
on ingress.
Step 3
Router(config-map-class)# frame-relay min-cir {in |
out} bps
Configures the minimum acceptable incoming or
outgoing CIR for this group of PVCs.
Step 4
Router(config-map-class)# frame-relay cir out bits
Configures the outgoing excess CIR for this group of
PVCs. Configured the CIR size to match your carrier
to prevent the carrier from discarding DE bits on
ingress.
Step 5
Router(config-map-class)# frame-relay cir in bits
(Optional) Configures the incoming CIR size for this
group of PVCs.
Step 6
Router(config-map-class)# frame-relay adaptive
shaping becn
(Optional) Configures the adaptive traffic rate
adjustment to support backward explicit congestion
notification (BECN) on this group of PVCs.
Configuring VoFR Dial Peers
To configure a VoFR dial peer, you must uniquely identify the peer (by assigning it a unique tag number)
and define the outgoing serial port number and the virtual circuit number.
Depending on your dial plan configuration, you might need to consider how to configure voice networks
with variable-length dial plans, number expansion, excess digit playout, forward digits, and default voice
routes, or use hunt groups with dial peer preferences.
Note
On the Cisco MC3810 multiservice concentrator, a voice class can be configured to assign idle state
and out-of-service (OOS) signaling attributes to a VoFR dial peer. For more information, see the
“Configuring Trunk Connections and Conditioning Features” chapter.
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VoFR Configuration Task List
To configure a VoFR dial peer, use the following commands beginning in global configuration mode:
Step 1
Command
Purpose
Router(config)# dial-peer voice number vofr
Defines a VoFR dial peer and enters dial peer
configuration mode. All subsequent commands that
are entered in dial peer voice configuration mode
before exiting apply to this dial peer.
The number argument identifies the dial peer and
must be unique on the router. Do not duplicate a
specific tag number.
Step 2
Router(config-dial-peer)#
destination-pattern[+]string[T]
Configures the dial peer destination pattern. The
same restrictions for the string listed in the POTS dial
peer configuration also apply to the VoFR destination
pattern. Also configures standard VoFR dial peers for
switched calls on the tandem routers.
•
Plus sign (+)—(Optional) Indicates an E.164
standard number. The plus sign (+) is not
supported on the Cisco MC3810 multiservice
concentrator.
•
string—Specifies the E.164 or private dialing
plan telephone number. Valid entries are the
digits 0 through 9, the letters A through D, and
the following special characters:
– Asterisk (*) and pound sign (#) that appear
on standard touch-tone dial pads.
– Comma (,) inserts a pause between digits.
– Period (.) matches any entered digit (this
character is used as a wildcard).
•
Note
Step 3
Router(config-dial-peer)# session target interface
dlci [cid]
Cisco IOS Voice, Video, and Fax Configuration Guide
Tandem-switched calls are not allowed when
the call type is an FRF.11 trunk call. The
Cisco 7200 series routers can serve only as
tandem nodes in the VoFR network using
Cisco IOS Release 12.1. This is the only dial
peer procedure supported on the Cisco 7200
series.
Configures the Frame Relay session target for the dial
peer.
Note
VC-396
T—(Optional) Indicates that the
destination-pattern value is a variable length
dial-string.
The cid argument is required for FRF.11 trunk
calls.
Configuring Voice over Frame Relay
VoFR Configuration Task List
Step 4
Command
Purpose
Router(config-dial-peer)# session protocol
{cisco-switched | frf11-trunk}
(Optional) Configures the session protocol to support
switched calls or FRF.11 trunk calls. If FRF.11 trunk
calls are sent over the Frame Relay network, the
VoFR dial peers must be statically configured on both
sides of the trunk specifically to support FRF.11 trunk
calls.
FRF.11 trunk calls cannot be used in conjunction with
dial plans or be sent through tandem nodes.
Note
Step 5
Step 6
Router(config-dial-peer)# codec {type} [bytes
payload_size]
Router(config-dial-peer)# dtmf-relay
The cisco-switched keyword is the default.
Specifies the voice coder rate of speech and payload
size for the dial peer. The default dial peer codec is
g729r8. The keywords and arguments are as follows:
•
type—Specifies the coder rate of speech. The
rates are hardware-specific. Refer to the
Cisco IOS Voice, Video, and Fax Command
Reference.
•
bytes—(Optional) Specifies the payload size.
Each codec type defaults to a different payload
size if a value is not specified.
•
payload_size—(Optional) Specifies the payload
size by entering the bytes value. Each codec type
defaults to a different payload size if a value is
not specified. To obtain a list of the default
payload sizes, enter the codec command and the
bytes option followed by a question mark (?).
Note
The Cisco MC3810 multiservice
concentrator is limited to a maximum of 12
calls when using g729r8. Use g729ar8 to
support up to 24 calls on the Cisco MC3810
multiservice concentrator.
Note
If configuring switched voice calls on the
Cisco MC3810 multiservice concentrator,
configure the codec type on the voice port.
Note
For FRF.11 trunk calls, the codec values must
be set the same on both sides of the
connection.
(Optional) Specifies support for the DTMF relay to
improve end-to-end transport of the DTMF tones, if
the codec type configured is a low bit-rate codec such
as g729 or g723. DTMF tones do not always
propagate reliably with low bit-rate codecs.
DTMF relay is disabled by default.
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VoFR Configuration Task List
Step 7
Command
Purpose
Router(config-dial-peer)# signal-type {cas | cept |
ext-signal | transparent}
If Cisco trunk permanent calls are being configured,
the signal type is required. The signal type defines the
ABCD signaling packets that are generated by the
voice port and sent to the data network. Use the cas,
cept, ext-signal, and transparent keywords.
To configure FRF.11 calls, use only the cas and
ext-signal keywords. These keywords are optional on
Cisco 2600/3600 series routers and configure the
signal type on these routers for FXS-FXS trunks. The
keywords are as follows:
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•
cas—Default signaling type that is North
American CAS/robbed-bit signaling.
•
cept—Provides basic E1 ABCD protocol,
primarily Conférence Européenne des Postes et
des Télécommunications (CEPT) E&M
signaling, on the Cisco MC3810 multiservice
concentrator. This keyword is used for European
voice networks. If the keyword is used with FXS
or FXO voice ports, the signaling is equivalent to
Mercury Exchange Limited (MEL) CAS. The
keyword is not supported on the Cisco 2600/3600
series.
•
ext-signal—Used for required external signaling
channels (for example, common channeling
signaling), or when no signaling information is
sent over a permanent “dumb” voice pipe (for
example, carrying audio for a public address
system).
Configuring Voice over Frame Relay
VoFR Configuration Task List
Command
Purpose
•
transparent—Used on the Cisco MC3810
multiservice concentrator with digital voice ports
when the ABCD signaling bits are copied and
passed transparently from the T1/E1 interface
without interpretation (also known as transparent
FRF.11 signaling). The keyword enables the
Cisco MC3810 multiservice concentrator to
handle or transport unknown signaling protocols.
On the Cisco MC3810 multiservice concentrator
with analog voice ports, the transparent
keyword does not apply and is equivalent to the
cept keyword. This keyword is not supported on
the Cisco 2600 series and 3600 series in Cisco
IOS Release 12.2.
Note
By default, the Cisco MC3810 multiservice
concentrator, when configured using
transparent, operates the voice path in a
permanently open state so that voice packets
are sent (and network bandwidth consumed)
regardless of the state of the call.
The signal type must be configured in such a way that
the signal type is the same at both ends of the
permanent voice call. When a permanent connection
is configured between a T1/E1 Cisco MC3810
multiservice concentrator and an analog voice port on
a Cisco 2600 or Cisco 3600 series routers, the signal
type should be set to cas, which is the default.
Step 8
Router(config-dial-peer)# called-number
termination-string
Required for the Cisco 2600/3600 series routers only.
Configures the termination string for FRF.11 trunk
calls. This command is required to enable the router
to establish an incoming trunk connection.
This command applies only when the session
protocol command is set to frf11-trunk.
Note
Although this command is visible on the
Cisco MC3810 multiservice concentrator, the
command is disabled.
Step 9
Router(config-dial-peer)# no vad
(Optional) Disables VAD on the dial peer. This
command is enabled by default.
Step 10
Router(config-dial-peer)# sequence-numbers
(Optional) Enables the voice sequence number if
required for your configuration. This command is
disabled by default.
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Configuring Voice over Frame Relay
VoFR Configuration Task List
Command
Purpose
Step 11
Router(config-dial-peer)# preference value
(Optional) Configures a preference for the VoFR dial
peer. The value argument is a number from 0 to 10
where the lower the number, the higher the preference
in hunt groups.
Step 12
Router(config-dial-peer)# fax rate {2400 | 4800 |
7200 | 9600 | 14400 | disable | voice}
(Optional) Configures the transmission speed (in bps)
at which a fax will be sent to the dial peer.
The default is voice, which specifies the highest
possible transmission speed allowed by the voice
rate.
To configure another VoFR dial peer, exit dial peer configuration mode and repeat Steps 1 through 10.
Note
Repeat this procedure on the destination router on the other side of the FRF.11 trunk.
Configuring Switched Calls
To configure switched calls on Cisco 2600, 3600, and 7200 series routers and Cisco MC3810
multiservice concentrators, use the following commands beginning in interface configuration mode:
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Configuring Voice over Frame Relay
VoFR Configuration Task List
Command
Purpose
Step 1
Router(config-if)# frame-relay interface-dlci dlci
Enters the DLCI configuration mode.
Step 2
Router(config-fr-dlci)# vofr [data cid]
[call-control][cid]
Configures the Frame Relay DLCI to support VoFR.
When the vofr command is used, all subchannels on
the DLCI are configured for FRF.11 encapsulation
The keywords and arguments are as follows:
•
data—Selects a subchannel (CID) for data other
than the default subchannel (CID 4). The
recommended setting is vofr data 4
call-control 5.
•
cid—Specifies the subchannel to use for data.
Valid values are from 4 to 255. The default is 4.
If data is specified, a valid CID must be entered.
•
call-control—(Optional) Specifies that a
subchannel is reserved for call-control signaling.
Call-control is not supported on Cisco MC3810
multiservice concentrators.
•
cid—(Optional) Specifies the subchannel to use
for call-control signaling. Valid values are from
4 to 255. The default is 5. If call-control is
specified and a CID is not entered, the default
CID is used. If the vofr command is entered
without any keywords or arguments, the data
subchannel (cid) is 4 and there is no call-control
subchannel.
Note
The vofr command uses WFQ at the PVC
level. If the vofr cisco command is used,
WFQ cannot be disabled.
or
Router(config-fr-dlci)# vofr cisco
Configures the DLCI and the Cisco proprietary voice
encapsulation for switched calls to Cisco MC3810
multiservice concentrators. When this command is
entered, data CID 4 and call-control CID 5 are
automatically assigned.
If user-dialed calls are being configured, stop here. If
auto-ringdown calls are being configured, continue
to the next step.
Step 3
Router(config)# voice-port
Identifies the voice port to configure and enters the
voice-port configuration mode.
Note
Step 4
Router(config-voice-port)# connection [plar |
tie-line] destination-string
The voice-port command is hardware
specific. For more information, refer to the
Cisco IOS Voice, Video, and Fax Command
Reference.
Configures the private-line, auto-ringdown (PLAR)
or tie-line connection, specifying the telephone
number in the destination-string.
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VoFR Configuration Task List
Table 28 lists the supported VoFR connections and the appropriate commands to configure switched
calls.
Table 28
Supported VoFR Connections for Switched Calls
Switched Calls
(User-Dialed or Auto-Dialed)
Data
Fragmentation
Supported
To routers supporting VoFR
FRF.11 Annex C
To a Cisco MC3810 multiservice
concentrator running Cisco IOS
Releases before 12.1(2)T
Cisco
proprietary5
Frame Relay DLCI
Command1
Session Protocol
Command2
Voice Port
Command
vofr [data cid]
[call-control
[cid]]3
session protocol
cisco-switched4
For user-dialed calls: none
vofr cisco6
session protocol
cisco-switched
For user-dialed calls: none
For auto-ringdown calls:
connection plar
destination-string
For auto-ringdown calls:
connection plar
destination-string
1. The voice-encap option of the frame-relay interface-dlci command on the Cisco MC3810 multiservice concentrator is no longer supported.
2. Dial peer configuration mode.
3. The recommended use of this command is vofr data 4 call-control 5.
4. The session protocol cisco-switched command is the default setting. If the command is not entered, the setting still applies.
5. Cisco proprietary fragmentation is based on an early draft of FRF.12 and is compatible with Cisco MC3810 multiservice concentrators.
6. This command uses data CID 4 and call-control CID 5.
Tandem Switching of Switched Calls
Depending on which router is the end node and which is the tandem node, the correct Frame Relay PVC
type must be configured. Table 29 shows the router combinations that can serve as end and tandem nodes
and the command that is required to enable VoFR.
Table 29
VoFR End and Tandem Node Combinations
End Node
Tandem Node
Required VoFR Command
Cisco 2600, Cisco 3600, or Cisco 7200 and
Cisco MC3810 multiservice concentrator
Cisco 2600, Cisco 3600, or Cisco 7200 and
Cisco MC3810 multiservice concentrator
vofr call-control
Cisco 2600 or Cisco 3600 and Cisco MC3810 Cisco MC3810 multiservice concentrator
multiservice concentrator
running Cisco IOS releases before 12.1(2)T
vofr cisco
Cisco MC3810 multiservice concentrator
running Cisco IOS releases before 12.1(2)T
vofr cisco
Note
Cisco 2600, Cisco 3600, or Cisco 7200
When you are creating voice networks with a mixture of router types, the Cisco MC3810 multiservice
concentrator must be running Cisco IOS Release 12.0(3)XG, 12.0(4)T, or later releases, to act as a
tandem node. For each configured tandem node, two VoFR dial peers must be configured, one for
each tandem connection.
To configure VoFR dial peers on tandem routers, use the following commands beginning in global
configuration mode:
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VoFR Configuration Task List
Command
Purpose
Step 1
Router(config)# dial-peer voice number vofr
Defines a VoFR dial peer and enters dial peer
configuration mode. All subsequent commands that
are entered in dial peer voice configuration mode
before exiting apply to this dial peer.
Step 2
Router(config-dial-peer)# destination-pattern
[+]string[T]
Configures the dial peer destination pattern. The
same restrictions for the string listed in the POTS dial
peer configuration also apply to the VoFR destination
pattern.
Step 3
Router(config-dial-peer)# session target interface
dlci
Configures the Frame Relay session target for the dial
peer.
Step 4
Router(config-dial-peer)# preference value
(Optional) Configures a preference for the VoFR dial
peer. The value argument is a number from 0 to 10
where the lower the number, the higher the preference
in hunt groups.
To configure the next VoFR dial peer, exit dial peer configuration mode by entering exit, and repeat Steps
1 through 4. On tandem nodes, at least two VoFR dial peers are required, one for each call leg.
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VoFR Configuration Task List
Configuring Cisco Trunk Calls
Before configuring the Cisco trunk calls, consider the following restrictions and recommendations:
•
VoFR dial peers must be configured to send Cisco trunk calls over the Frame Relay network. Cisco
trunk calls are permanent calls. One critical task is configuring the signal type for the dial peer. It
must be the same at both ends of the permanent voice call. See the “Configuring Dial Peers, Dial
Plans, and Digit Manipulation” chapter for more information.
•
When a permanent connection between a T1/E1 Cisco MC3810 multiservice concentrator and an
analog voice port on a Cisco 2600 or Cisco 3600 series routers is configured, the default signal type
is cas.
•
Use of Cisco trunks for permanent calls is recommended over FRF.11 trunk calls unless FRF.11
compliant standards-based interworking is required with non-Cisco devices. The Cisco trunk
protocol is a superset of the FRF.11 protocol and contains Cisco proprietary extensions designed to
support switched call routing and other advanced features.
Table 30 lists the supported VoFR connections and the commands to enter.
Table 30
VoFR Connections for Cisco Trunk Calls
Cisco Trunk Calls
Data Fragmentation
Supported
VoFR
Command
Session Protocol
Command1
Voice Port
Command
To routers supporting VoFR
FRF.11 Annex C
vofr data cid
call-control cid
session protocol
cisco-switched
connection trunk
destination-string
[answer mode]
To a Cisco MC3810 multiservice
concentrator running Cisco IOS
Releases before 12.0(7) XK and
12.1(2)T
Cisco proprietary
vofr cisco2
session protocol
cisco-switched
connection trunk
destination-string
[answer mode]
1. The session protocol cisco-switched command, whether entered or not, is the default setting.
2. When the cisco keyword is entered, Cisco proprietary data implementation is enabled. This implementation is used only for backward compatibility to
earlier releases.
To configure Cisco trunk permanent calls, use the following commands beginning in interface
configuration mode:
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Configuring Voice over Frame Relay
VoFR Configuration Task List
Step 1
Command
Purpose
Router(config-if)# frame-relay interface-dlci dlci
Configures the DLCI to support VoFR.
Note
Step 2
Router(config-if)# vofr [[cisco] | [[data cid]
[call-control][cid]]]]
The voice-encap option of the frame-relay
interface-dlci command on the Cisco
MC3810 multiservice concentrator is no
longer supported beginning in Cisco IOS
12.2.
Enables VoFR on the DLCI. If the vofr command is
entered without any keywords or arguments, the data
subchannel is CID 4, and there is no call-control
subchannel.
Note
When the vofr command is used, all
subchannels on the DLCI are configured for
FRF.11 encapsulation. This configuration
uses the standard FRF.11 Annex C
fragmentation.
The vofr command uses WFQ at the PVC level. If the
vofr cisco command is used, WFQ cannot be
disabled.
If only tandem calls are being configured, stop here,
otherwise proceed to Step 3.
Step 3
Router(config]# voice-port
Identifies the voice port to configure and enters
voice-port configuration mode.
Note
The voice-port command is hardware
specific. See the Cisco IOS Voice, Video, and
Fax Command Reference Guide for more
information.
Step 4
Router(config-voice-port)# connection trunk
destination-string [answer-mode]
Configures the trunk connection by specifying the
telephone number in destination-string. One side
must be the call initiator (master) and the other side
is the call answerer (slave). By default, the voice port
is the master. The answer-mode keyword specifies
the voice port that operates in slave mode.
Step 5
Router(config-voice-port)# shutdown
Shuts down the voice port.
Step 6
Router(config-voice-port)# no shutdown
Reactivates the voice port to enable the trunk
connection.
Note
When the connection trunk or no connection trunk command is entered, the voice port must be
toggled by entering shutdown, and then no shutdown before the changes take effect.
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VoFR Configuration Task List
Configuring FRF.11 Trunk Calls
On the Cisco MC3810 multiservice concentrators and Cisco 2600 and 3600 series routers, FRF.11 trunk
calls to a second router can be configured, except tandem FRF.11 trunk calls. Configuring FRF.11 trunk
calls to a second router requires that the session protocol dial peer configuration command be set to
frf11-trunk.
Table 31 lists the supported VoFR connections and the required commands to configure FRF.11 trunk
calls.
Table 31
VoFR Connections for FRF.11 Trunk (Private-Line) Calls
FRF.11 Trunk Calls
Data Fragmentation
Supported
VoFR
DLCI Command1
Session Protocol
Command
Voice Port
Command
To routers supporting VoFR
FRF.11 Annex C
vofr [data cid]
[call-control cid]2
session protocol
frf11-trunk
connection trunk
destination-string
[answer mode]
1. Dial peer configuration mode.
2. For FRF.11 trunk calls, the call-control option is not required. It is required only if you mix FRF.11 trunk calls with other types of voice calls on the same
PVC.
To configure FRF.11 trunk calls, use the following commands beginning in interface configuration
mode:
Command
Purpose
Step 1
Router(config-if)# frame-relay interface-dlci dlci
Configures the DLCI and enters DLCI configuration
mode.
Step 2
Router(config-fr-dlci)# vofr [data cid]
[call-control cid]
Configures the DLCI and optionally enters the data
and call-control CIDs. When the keywords and
arguments are configured, all subchannels on the
DLCI are configured for FRF.11 encapsulation
except the data subchannel. If no keywords or
arguments are entered, the data subchannel is CID 4,
and there is no call-control subchannel.
Step 3
Router(config)# voice-port
Identifies the voice port to configure and enters
voice-port configuration mode.
Note
The voice-port command is hardware
specific. Refer to the Cisco IOS Voice, Video,
and Fax Command Reference publication for
more information.
Step 4
Router(config-voice-port)# connection trunk
destination-string [answer-mode]
Configures the trunk connection by specifying the
telephone number in destination-string. One side of a
call must act as the call initiator (master) and the
other side as the call answerer (slave). By default, the
voice port is the master.
Step 5
Router(config-voice-port)# shutdown
Shuts down the voice port.
Step 6
Router(config-voice-port)# no shutdown
Reactivates the voice port to enable the trunk
connection.
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VoFR Configuration Task List
Note
When the connection trunk or no connection trunk command is entered, the voice port must be
toggled by entering shutdown, and then no shutdown before the changes take effect.
Verifying the Voice Connections
To verify switched calls voice connections, perform the following tasks:
•
Pick up the telephone handset and verify that there is a dial tone.
•
Call from a local telephone to the configured dial peer and verify that the call completes.
To verify the FXO-FXS trunk calls to a remote PBX, perform the following tasks:
•
Pick up the telephone and listen for a dial tone from the remote PBX.
•
Dial a telephone number, so that the remote PBX routes the call.
To verify the voice connections, perform the following tasks:
•
Check the validity of the dial peer and voice port configuration by performing the following tasks:
– Enter the show dial-peer voice command to verify that the data configured is correct.
– Enter the show dial-peer voice summary command to check the validity of the dial peer
configurations.
– Enter the show voice port command to show the status of the voice ports.
– Enter the show call active voice with the keyword brief to show the call status for all voice
ports.
– Enter the show voice call command to check the validity of the voice port configuration.
– Enter the show voice dsp command to show the current status of all DSP voice channels.
– Enter the show voice permanent command to show the status of Cisco trunk permanent calls.
– Enter the show call history command to show the active call table.
•
Check the validity of the VoFR configuration on the DLCI by performing the following task:
– Enter the show frame-relay vofr [interface [dlci [cid]]] command to show the VoFR
configuration. This command is not supported on the Cisco MC3810 multiservice concentrator
when the vofr cisco command is configured.
Verifying the Frame Relay Configuration
Check the validity of the configuration by performing the following tasks:
•
Enter the show frame-relay pvc command to show the status of the PVCs.
•
Enter the show frame-relay vofr command with the arguments interface, dlci, and cid to show
statistics and information on the open subchannels. This command does not display if the vofr cisco
command is entered on the Cisco MC3810 multiservice concentrator.
•
Enter the show frame-relay fragment command with the arguments interface number and dlci to
show the Frame Relay fragmentation configuration.
•
Enter the show traffic-shape queue command to display the traffic-shaping information if Frame
Relay traffic shaping is configured. The queue option displays the queueing statistics.
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VoFR Configuration Task List
Troubleshooting Tips
To troubleshoot and resolve configuration issues, perform the following tasks:
•
If no calls are going through, ensure that the frame-relay voice bandwidth command is configured.
•
If VoFR is configured on a PVC and there are problems with data connectivity on that PVC, ensure
that the frame-relay fragment command has been configured.
•
If data is not being transmitted but fragmentation is configured, ensure that Frame Relay traffic
shaping is turned on.
•
If the problem is with the dial plan or the dial peers, use the show dial-plan number command with
the argument dial string to display which dial peers are being used when a specific number is called.
•
If there are problems connecting an FRF.11 trunk call, ensure that the session protocol dial peer
command is set to frf11-trunk.
•
If FRF.11 trunk calls on the Cisco 2600 or Cisco 3600 series routers are being configured, verify
that the called-number vofr dial peer command is configured and that its number matches the
destination pattern of the corresponding POTS dial peer.
•
Ensure that the voice port is set to no shutdown.
•
Ensure that the serial port or the T1/E1 controller is set to no shutdown.
•
Toggle the voice port by first entering shutdown, and then no shutdown every time the connection
trunk or no connection trunk command is entered.
Monitoring and Maintaining the VoFR Configuration
To monitor and maintain the VoFR configuration, use the following commands in EXEC mode as
needed:
Command
Purpose
Router# show call active voice [brief]
Displays the active call table.
Router# show call history voice [last number] | [brief]
Displays the call history table.
or
Router# show call history voice record
Router# show dial-peer voice
Displays configuration information and call statistics for
dial peers.
Router# show frame-relay fragment
Displays information about the Frame Relay fragmentation
taking place in the Cisco router.
Router# show frame-relay pvc
Displays statistics about PVCs for Frame Relay interfaces.
Router# show frame-relay vofr
Displays the FRF.11 subchannels information on VoFR
DLCIs.
Router# show interfaces serial
Displays information about a serial interface.
Router# show traffic-shape queue
Displays information about the elements queued at a
particular time at the VC (DLCI) level.
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VoFR Configuration Examples
Command
Purpose
Router# show voice call
Displays the call status for all voice ports on the Cisco
MC3810 multiservice concentrators.
Router# show voice permanent-call
Displays information about the permanent calls on a voice
interface.
VoFR Configuration Examples
This section provides specific configuration examples for VoFR connections and includes:
•
Two Routers Using Frame Relay Fragmentation Example, page 409
•
Two Routers Using a VoFR PVC Example, page 410
•
Router Using VoFR PVCs Connected to Cisco MC3810s Before 12.1(2)T Example, page 410
•
Cisco Trunk Calls Between Two Routers Example, page 411
•
FRF.11 Trunk Calls Between Two Routers Example, page 412
•
Tandem Configuration Examples, page 413
•
Cisco Trunk Call with Hunt Groups Example, page 418
Two Routers Using Frame Relay Fragmentation Example
Figure 75 shows an example of Frame Relay fragmentation between two routers. This configuration uses
FRF.12 fragmentation.
Figure 75
Two Routers Using Frame Relay Fragmentation
Cisco 3600
Cisco 3600
Router A
DLCI
100
DLCI
100
Router B
18293
Frame Relay
Router A
Router B
hostname 3600A
!
interface serial 0/0
ip address xxx.xxx.xxx 255.255.255.0
frame-relay traffic shaping
!
frame-relay interface-dlci 100
class toto
!
map-class frame-relay toto
encapsulation frame-relay
frame-relay cir s
frame-relay bc u
frame-relay fragment y
hostname 3600B
!
interface serial 0/0
ip address xxx.xxx.xxx 255.255.255.0
frame-relay traffic shaping
frame-relay class toto
frame-relay interface-dlci 100
!
map-class frame-relay toto
encapsulation frame-relay
frame-relay cir s
frame-relay bc u
frame-relay fragment y
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VoFR Configuration Examples
Two Routers Using a VoFR PVC Example
Figure 76 shows an example of two routers that use FRF.11 Annex C fragmentation with connections
using a VoFR PVC.
Figure 76
Two Cisco 3600 Series Routers Using a VoFR PVC
Cisco 3600
Cisco 3600
Router A
DLCI
100
DLCI
100
Router B
18292
Frame Relay
Router A
Router B
hostname 3600A
!
interface serial 0/0
frame-relay traffic shaping
!
frame-relay interface-dlci 100
vofr data z
class toto
!
map-class frame-relay toto
frame-relay voice-bandwidth t
frame-relay min-cir x
frame-relay cir s
frame-relay bc u
frame-relay fragment y
hostname 3600B
!
interface serial 0/0
frame-relay traffic shaping
frame-relay class toto
!
frame-relay interface-dlci 100
vofr data z
!
map-class frame-relay toto
frame-relay voice-bandwidth t
frame-relay min-cir x
frame-relay cir s
frame-relay bc u
frame-relay fragment y
Router Using VoFR PVCs Connected to Cisco MC3810s Before 12.1(2)T Example
Figure 77 shows an example of a Cisco 3600 series router with connections to a Cisco MC3810
multiservice concentrator running a Cisco IOS release before12.1(2)T. In this example, the VoFR
interface on both the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator is
configured by using the vofr cisco command. This configuration uses FRF.11 Annex C fragmentation.
Figure 77
Router Using VoFR PVCs Connected to a Cisco MC3810 Multiservice Concentrator
Cisco 3600
Cisco MC3810
Router A
DLCI
100
Cisco IOS Voice, Video, and Fax Configuration Guide
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DLCI
100
Router B
18294
Frame Relay
Configuring Voice over Frame Relay
VoFR Configuration Examples
Router A
Router B
interface serial 0/0
ip address xxx.xxx.xxx
255.255.255.0
frame-relay traffic shaping
!
frame-relay interface-dlci 100
vofr cisco
class toto
!
map-class frame-relay toto
frame-relay voice-bandwidth t
frame-relay min-cir x
frame-relay cir s
frame-relay bc u
frame-relay fragment y
interface serial 0
ip address xxx.xxx.xxx
255.255.255.0
frame-relay traffic shaping
frame-relay class toto
!
frame-relay interface-dlci 100
vofr cisco
!
map-class frame-relay toto
frame-relay voice-bandwidth t
frame-relay min-cir x
frame-relay cir s
frame-relay bc u
frame-relay fragment y
Cisco Trunk Calls Between Two Routers Example
Figure 78 shows an example of VoFR Cisco trunk calls between two routers.
7001
PBX
Cisco Trunk (Private-Line) Calls Between Two Routers
Cisco
MC3810
Cisco 3600
2/0/0
Router A
6001
Frame Relay
DLCI
100
DLCI
100
1/5
Router B
Router A
Router B
interface serial 0/0
ip address xxx.xxx.xxx
255.255.255.0
encapsulation frame-relay
frame-relay traffic shaping
frame-relay interface-dlci 100
class voice
vofr data 4 call-control 5
!
map-class frame-relay voice
frame relay cir s
frame relay bc u
frame-relay voice bandwidth v
frame-relay min-cir x
frame-relay fragment y
!
voice-port 2/0/0
connection trunk 6001 answer-mode
!
dial-peer voice 1 pots
destination-pattern 7001
interface serial 0
ip address xxx.xxx.xxx
255.255.255.0
encapsulation frame-relay
frame-relay traffic shaping
frame-relay interface-dlci 100
class voice
vofr data 4 call-control 5
!
map-class frame-relay voice
frame relay cir s
frame relay bc u
frame-relay voice bandwidth v
frame-relay min-cir x
frame-relay fragment y
!
voice-port 1/5
connection trunk 7001
!
dial-peer voice 2 pots
destination-pattern 6001
18295
Figure 78
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Configuring Voice over Frame Relay
VoFR Configuration Examples
Router A
Router B
port 2/0/0
!
dial-peer voice 2 vofr
codec x bytes y
destination-pattern 6001
session protocol cisco-switched
session target Sn 100
port 1/5
!
dial-peer voice 4 vofr
codec x bytes y
destination-pattern 7001
session protocol cisco-switched
session target Sn 100
FRF.11 Trunk Calls Between Two Routers Example
Figure 79 shows an example of FRF.11 trunk calls configured between two routers.
Figure 79
7001
Cisco 3600
2/0/0
Router A
Cisco
MC3810
Frame Relay
FRF.11 trunk
DLCI
100
DLCI
100
6001
1/5
Router B
18296
PBX
FRF.11 Trunk Calls Between Two Routers
Cisco IOS Voice, Video, and Fax Configuration Guide
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Configuring Voice over Frame Relay
VoFR Configuration Examples
Router A
Router B
hostname 3600A
!
interface serial 0/0
ip address xxx.xxx.xxx 255.255.255.0
encapsulation frame-relay
frame-relay traffic shaping
frame-relay interface-dlci 100
class voice
vofr data 4
!
map-class frame-relay voice
frame-relay cir s
frame-relay min-cir in x
frame-relay bc u
frame-relay voice bandwidth v
frame-relay fragment y
!
voice-port 2/0/0
connection trunk 6001
!
dial-peer voice 1 pots
destination-pattern 7001
port 2/0/0
!
dial-peer voice 2 vofr
codec x bytes y
destination-pattern 6001
session protocol frf11-trunk
session target Sn 100 d
called-number 7001
dtmf-relay
vad
hostname mc3810B
!
interface serial 0
ip address xxx.xxx.xxx 255.255.255.0
encapsulation frame-relay
frame-relay traffic shaping
frame-relay interface-dlci 100
class voice
vofr data 4
!
map-class frame-relay voice
frame-relay cir s
frame-relay min-cir in x
frame-relay bc u
frame-relay voice bandwidth v
frame-relay fragment y
!
voice-port 1/5
connection trunk 7001
!
dial-peer voice 2 pots
destination-pattern 6001
port 1/5
!
dial-peer voice 4 vofr
codec x bytes y
destination-pattern 7001
session protocol frf11-trunk
session target Sn 100 d
dtmf-relay
vad
Tandem Configuration Examples
Figure 80 shows an example of a tandem configuration with two Cisco 3600 series routers as endpoints
and a third Cisco 3600 series router as a tandem node.
Figure 80
Tandem Configuration with Three Routers for Switched Calls
1001
2001
Serial
0/0
1/0/0
DLCI
100
Router A
PBX
Frame Relay
Cisco 3600
vofr
vofr
Router C
Cisco
3600
Serial
0/1
1/0/0
DLCI
200
Router B
PBX
18297
Cisco
3600
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Configuring Voice over Frame Relay
VoFR Configuration Examples
Router A Endpoint
Router C Tandem Node
Router B Endpoint
hostname 3600A
!
interface serial 0/0
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 100
class voice
vofr data 4 call-control 5
!
map-class frame-relay voice
frame-relay cir a
frame-relay min-cir t
frame-relay bc b
frame-relay voice bandwidth c
frame-relay fragment d
!
dial-peer voice 1 pots
destination-pattern 1001
port 1/0/0
!
dial-peer voice 2 vofr
destination-pattern 2...
session target serial 0/0 100
!
voice-port 1/0/0
hostname3600C
!
interface serial 0/0
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 100
class voice
vofr data 4 call-control 5
!
interface serial 0/1
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 200
class voice
vofr
!
map-class frame-relay voice
frame-relay cir a
frame-relay min-cir t
frame-relay bc b
frame-relay voice bandwidth c
frame-relay fragment d
!
dial-peer voice 1 vofr
destination-pattern 1...
session target serial 0/0 100
!
dial-peer voice 2 vofr
destination-pattern 2...
session target serial 0/1 200
hostname3600B
!
interface serial 0/0
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 100
class voice
vofr data 4 call-control 5
!
map-class frame-relay voice
frame-relay cir a
frame-relay min-cir t
frame-relay bc b
frame-relay voice bandwidth c
frame-relay fragment d
!
dial-peer voice 1 pots
destination-pattern 2001
port 1/0/0
!
dial-peer voice 2 vofr
destination-pattern 1...
session target serial 0/0 200
!
voice-port 1/0/0
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VoFR Configuration Examples
Figure 81 shows an example of a tandem configuration with a Cisco MC3810 multiservice concentrator
acting as a tandem node.
Figure 81
Tandem Configuration with a Cisco MC3810 Multiservice Concentrator Tandem Node for
Switched Calls
1001
Cisco
3600
2001
Cisco
Serial 3600
0/1
1/0/0
DLCI
200
Router B
PBX
18298
Serial
0/0
1/0/0
DLCI
100
Router A
PBX
Frame Relay
Cisco MC3810
vofr
vofr
Cisco
Cisco
Router C
Router A Endpoint
Router C Tandem Node
Router B Endpoint
interface serial 0/0
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 100
class voice
vofr data 4 call-control 5
!
map-class frame-relay voice
frame-relay cir a
frame-relay min-cir t
frame-relay bc b
frame-relay voice bandwidth c
frame-relay fragment d
!
dial-peer voice 1 pots
destination-pattern 1001
port 1/0/0
!
!
dial-peer voice 2 vofr
destination-pattern 2...
session target serial 0/0 100
!
voice-port 1/0/0
!
!
!
!
interface serial 0
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 100
class voice
vofr data 4 call-control 5
!
interface serial 1
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 200
class voice
vofr data 4 call-control 5
!
map-class frame-relay voice
frame-relay cir a
frame-relay min-cir t
frame-relay bc b
frame-relay voice bandwidth c
frame-relay fragment d
!
dial-peer voice 1 vofr
destination-pattern 1...
session target serial 0/0 100
!
dial-peer voice 2 vofr
destination-pattern 2...
session target serial 0/1 200
interface serial 0/0
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 100
class voice
vofr data 4 call-control 5
!
map-class frame-relay voice
frame-relay cir a
frame-relay min-cir t
frame-relay bc b
frame-relay voice bandwidth c
frame-relay fragment d
!
dial-peer voice 1 pots
destination-pattern 2001
port 1/0/0
!
!
dial-peer voice 2 vofr
destination-pattern 1...
session target serial 0/0 200
!
voice-port 1/0/0
!
!
!!
!
Figure 82 shows an example of a tandem configuration with a Cisco MC3810 multiservice concentrator
acting as an endpoint node for Cisco trunk calls. When a Cisco MC3810 multiservice concentrator is on
a VoFR network, the configuration for connections to and from the Cisco MC3810 multiservice
concentrator is slightly different than for other routers that support VoFR. The vofr cisco command is
required for those connections.
Cisco IOS Voice, Video, and Fax Configuration Guide
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Configuring Voice over Frame Relay
VoFR Configuration Examples
Figure 82
Tandem Configuration with a Cisco MC3810 Multiservice Concentrator Endpoint Node
1001
2001
Cisco
2600
Cisco
MC3810
Serial
0
1/1
DLCI
200
Router B
PBX
18300
Serial
0/0
1/0/0
DLCI
100
Router A
PBX
Frame Relay
Cisco 3600
vofr
vofr
Cisco
Router C
Trunk connection
Router A Endpoint
Router C Tandem Node
Router B Endpoint
interface serial 0/0
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 100
class voice
vofr data 4 call-control 5
!
map-class frame-relay voice
frame-relay cir a
frame-relay min-cir t
frame-relay bc b
frame-relay voice bandwidth c
frame-relay fragment d
!
dial-peer voice 1 pots
destination-pattern 1001A
port 1/0/0
!
dial-peer voice 2 vofr
destination-pattern 2...
session target serial 0/0 100
!
voice-port 1/0/0
connection trunk 2001A answer-mode
!
!
!
!
interface serial 0/0
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 100
class voice
vofr data 4 call-control 5
!
interface serial 0/1
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 200
class voice
vofr data 4 call-control 5
!
map-class frame-relay voice
frame-relay cir a
frame-relay min-cir t
frame-relay bc b
frame-relay voice bandwidth c
frame-relay fragment d
!
dial-peer voice 1 vofr
destination-pattern 1...
session target serial 0/0 100
!
dial-peer voice 2 vofr
destination-pattern 2...
session target serial 0/1 200
interface serial 0
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 200
class voice
vofr data 4 call-control 5
!
map-class frame-relay voice
frame-relay cir a
frame-relay min-cir t
frame-relay bc b
frame-relay voice bandwidth c
frame-relay fragment d
!
dial-peer voice 1 pots
destination-pattern 2001A
port 1/1
!
dial-peer voice 2 vofr
destination-pattern 1...
session target serial 0 200
!
voice-port 1/1
connection trunk 1001A
!
!
!
!
Figure 83 shows an example of a tandem configuration with Cisco MC3810 multiservice concentrators
as both endpoint and tandem nodes.
Note
When a Cisco MC3810 multiservice concentrator running Cisco IOS software releases earlier than
12.1(2)T are used on a VoFR network, the configuration for connections to and from that Cisco
MC3810 multiservice concentrator is slightly different from what is used for other routers that
support VoFR. The vofr cisco command is required for these connections on the Cisco MC3810
multiservice concentrator.
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VoFR Configuration Examples
Configuration with All Cisco MC3810 Multiservice Concentrators as Endpoint and Tandem
Nodes
1001
2001
Frame Relay
Cisco MC3810
Cisco
MC3810
1/1
PBX
Router A
DLCI
100
vofr
Cisco
Cisco
MC3810
vofr
Cisco
Router C
Serial 0
DLCI
200
1/1
Router B
PBX
18301
Figure 83
Serial 1
Router A Endpoint
Router C Tandem Node
(Cisco IOS Releases Before 12.1(2)T)
Router B Endpoint
interface serial 0
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 100
class voice
vofr cisco
!
map-class frame-relay voice
frame-relay cir a
frame-relay bc b
frame-relay voice bandwidth c
frame-relay min-cir t
!
!
dial-peer voice 1 pots
destination-pattern 1001
port 1/1
!
dial-peer voice 2 vofr
destination-pattern 2...
session target serial 0 100
!
voice-port 1/1
!
!
!
!
!
interface serial 0
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 100
class voice
vofr cisco
!
interface serial 1
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 200
class voice
vofr cisco
!
map-class frame-relay voice
frame-relay cir a
frame-relay min-cir t
frame-relay bc b
frame-relay voice bandwidth c
frame-relay fragment d
!
dial-peer voice 1 vofr
destination-pattern 1...
session target serial 0 100
!
dial-peer voice 2 vofr
destination-pattern 2...
session target serial 1 200
interface serial 0
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 200
class voice
vofr cisco
!
map-class frame-relay voice
frame-relay cir a
frame-relay bc b
frame-relay voice bandwidth c
frame-relay fragment d
frame-relay min-cir t
!
dial-peer voice 1 pots
destination-pattern 2001
port 1/1
!
dial-peer voice 2 vofr
destination-pattern 1...
session target serial 0 200
!
voice-port 1/1
!
!
!
!
!
Cisco IOS Voice, Video, and Fax Configuration Guide
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Configuring Voice over Frame Relay
VoFR Configuration Examples
Cisco Trunk Call with Hunt Groups Example
Figure 84 shows an example of a Cisco trunk call with hunt groups configured. In this example, the two
routers are in master-slave mode with a backup path. Router B is configured as a slave and Router A is
configured as the master. The master makes periodic attempts to establish the trunk until the trunk is
established.
Two dial peers match the destination string configured in the voice port, but one dial peer has a higher
preference, so the call setup is attempted through that dial peer. If the call setup fails, the master can
continue attempting call setups using the next available dial peer. After all dial peers are exhausted, the
master can continue following the list cyclically by starting again from the dial peer with the highest
preference.
Cisco Trunk Call with Hunt Groups
Serial 0
Cisco
MC3810
1001
1
Serial 0
Frame Relay
Cisco
MC3810
2001
DLCI 100
1/1
2
PBX
Router A
X = Preference
1/1
DLCI 200
Serial 1
Router B
Serial 1
number for
hunt groups
Router A
Router B
interface serial 0
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 100
class voice
vofr data 4 call-control 5
!
interface serial 1
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 200
class voice
vofr data 4 call-control 5
!
map-class frame-relay voice
frame-relay cir a
frame-relay bc b
frame-relay voice bandwidth c
frame-relay min-cir t
!
dial-peer voice 1 pots
destination-pattern 1001A
port 1/1
!
dial-peer voice 100 vofr
destination-pattern 2...
interface serial 0
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 100
class voice
vofr data 4 call-control 5
!
interface serial 1
encapsulation frame-relay
frame-relay traffic-shaping
frame-relay interface-dlci 200
class voice
vofr data 4 call-control 5
!
map-class frame-relay voice
frame-relay cir a
frame-relay bc b
frame-relay voice bandwidth c
frame-relay min-cir t
!
dial-peer voice 1 pots
destination-pattern 2001A
port 1/1
!
dial-peer voice 100 vofr
destination-pattern 1...
Cisco IOS Voice, Video, and Fax Configuration Guide
VC-418
18302
Figure 84
Configuring Voice over Frame Relay
VoFR Configuration Examples
Router A
Router B
session target serial0 100
preference 1
!
dial-peer voice 200 vofr
destination-pattern 2...
session target serial1 200
preference 2
!
voice-port 1/1
connection trunk 2005A
description FXO port
!
!
session target serial0 100
preference 1
!
dial-peer voice 200 vofr
destination-pattern 1...
session target serial1 200
preference 2
!
voice-port 1/1
description FXS port
connection trunk 1001A answer-mode
!
!
Cisco IOS Voice, Video, and Fax Configuration Guide
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Configuring Voice over Frame Relay
VoFR Configuration Examples
Cisco IOS Voice, Video, and Fax Configuration Guide
VC-420
Configuring Voice over ATM
This chapter describes Voice over ATM (VoATM) and contains the following sections:
•
VoATM Overview, page 421
•
VoATM Prerequisite Tasks, page 425
•
VoATM Configuration Task List, page 426
•
VoATM Configuration Examples, page 443
For a description of the VoATM commands, see the Cisco IOS Voice, Video, and Fax Applications
Command Reference. For information about software configuration requirements for the Digital T1
Packet Voice trunk network modules on the Cisco 2600 and Cisco 3600, see the “Configuring Voice
Ports” chapter. For more information about configuring ATM for data transmission, see the Cisco IOS
Wide-Area Networking Configuration Guide and Command Reference.
To identify the hardware platform or software image information associated with a feature in this
chapter, use the Feature Navigator on Cisco.com to search for information about the feature or refer to
the software release notes for a specific release. For more information, see the “Identifying Supported
Platforms” section in the “Using Cisco IOS Software” chapter.
VoATM Overview
VoATM enables a router to carry voice traffic (for example, telephone calls and faxes) over an ATM
network. An ATM network is a cell-switching and multiplexing technology designed to combine the
benefits of circuit switching (constant transmission delay and guaranteed capacity) and packet switching
(flexibility and efficiency for intermittent traffic).
All traffic to or from an ATM network is prefaced with a virtual path identifier (VPI) and virtual channel
identifier (VCI). A VPI-VCI pair is considered a single virtual circuit. Each virtual circuit is a private
connection to another node on the ATM network. Each virtual circuit is treated as a point-to-point
mechanism to another router or host and is capable of supporting bidirectional traffic.
Each ATM node establishes a separate connection to every other node in the ATM network with which
it must communicate. All such connections are established by means of a permanent virtual circuit
(PVC) or a switched virtual circuit (SVC) with an ATM signaling mechanism. This signaling is based
on the ATM Forum User-Network Interface (UNI) Specification V3.0.
Each virtual circuit is considered a complete and separate link to a destination node. Data can be
encapsulated as needed across the connection, and the ATM network disregards the contents of the data.
The only requirement is that data be sent to the ATM processor card of the router in a manner that follows
the specific ATM adaptation layer (AAL) format.
Cisco IOS Voice, Video, and Fax Configuration Guide
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Configuring Voice over ATM
VoATM Overview
An ATM connection transfers raw bits of information to a destination router or host. The ATM router
takes the common part convergence sublayer (CPCS) frame, carves it up into 53-byte cells, and sends
the cells to the destination router or host for reassembly. In AAL5 format, 48 bytes of each cell are used
for the CPCS data and the remaining 5 bytes are used for cell routing. The 5-byte cell header contains
the destination VPI-VCI pair, payload type, cell loss priority (CLP), and header error control (HEC)
information.
AAL Technology
AAL defines the conversion of user information into the ATM cells. AAL protocols perform a
convergence function; that is, they take whatever traffic is to be sent across the ATM network, establish
the appropriate connections, and then package the traffic received from the higher layers into the 48-byte
information payload that is passed down to the ATM layer for transmission. At the receiving level, the
AAL layer must receive the information payloads passed up from the ATM layer and put the payloads
into the form expected by the higher layer.
The AAL layers provide a service to the higher layers that corresponds to the four classes of traffic.
AAL1 and AAL2 handle isochronous traffic, such as voice and video, but are not relevant to the router.
AAL3/4 and AAL5 support data communications by segmenting and reassembling packets.
AAL2 is a bandwidth-efficient, standards-based trunking method for transporting compressed voice,
voice-band data, circuit-mode data, and frame-mode data. VoATM with AAL2 trunking provides the
following functionality:
•
Increased quality of service (QoS) capabilities
•
Robust architecture
•
Signalling transparency
•
CAS and CCS support
AAL5 is designed to support only message-mode, nonassured operation. AAL5 packets contain 48 bytes
of data and a 5-byte header.
Variable Bit Rate Real-Time Options for Traffic Shaping
Variable bit rate (VBR) is a QoS class defined by the ATM Forum for ATM networks. VBR is subdivided
into a real-time (RT) class and nonreal time (NRT) class. RT VBR is used for connections in which there
is a fixed timing relationship between samples, as in the case of traffic shaping. NRT VBR is used for
connections in which there is no fixed timing relationship between samples, but which still need a
guaranteed QoS.
Traffic shaping prevents a carrier from discarding incoming calls from a Cisco router. Traffic shaping is
performed by configuring the peak, average, and burst options for voice traffic. Burst is required if the
PVC is carrying bursty traffic. Peak, average, and burst are required so the PVC can effectively handle
the bandwidth for the number of voice calls.
Cisco IOS Voice, Video, and Fax Configuration Guide
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Configuring Voice over ATM
VoATM Overview
Cisco Trunk Calls on Cisco MC3810 Multiservice Concentrators
Cisco trunk (private-line) calls are basically dynamic switched calls of indefinite duration that use a
fixed destination telephone number and include optional transparent end-to-end signaling. The telephone
number of the destination endpoint is permanently configured into the router so that it always selects a
fixed destination. After the call is established, either at boot-up or when configured, the call stays up
until one of the voice ports or network ports is shut down or a network disruption occurs.
The Cisco trunk call functionality provides the following benefits:
•
True permanent, private-line connections.
•
Comprehensive busyout support for trunk connections. For more information, see to the
“Configuring Trunk Connections and Conditioning Features” chapter.
•
Transparent CAS protocol transport to enable the trunk to carry arbitrary ABCD signaling protocols.
•
Conversion from North American signaling protocols to CEPT (Conférence Européenne des Postes
et des Télécommunications) signaling protocols used for European voice networks.
•
Remote analog-to-digital channel-bank operation for converting from digital voice multiplexer
(DVM) to ATM voice multiplexer (AVM) configurations on the Cisco MC3810 multiservice
concentrator.
VoATM Dial Peers
Establishing two-way communications using VoATM requires a specific voice connection between two
defined endpoints. As shown in Figure 85, the plain old telephone service (POTS) dial peer establishes
the source (the originating telephone number and voice port) of the call, and the VoATM dial peer
establishes the destination by associating the destination phone number with a specific ATM virtual
circuit.
Calls from the Perspective of Router 1
Source
Destination
Voice port
2/0/0
4085554000
Router
No. 1
Router
No. 2
PVC 150
ATM
PVC 150
Voice port
2/0/0
27855
Figure 85
3105551000
In Figure 85 the destination string, 14085554000, coming from the source, maps to U.S. phone number
555-4000, with the digit “1” plus the area code “408” preceding the number. When configuring the
destination pattern, set the dial string to match the local dial conventions.
Cisco IOS Voice, Video, and Fax Configuration Guide
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Configuring Voice over ATM
VoATM Overview
When both POTS dial peers are connected to the same router and share the same destination IP address,
the VoATM dial peer does not need to be configured (see Figure 86).
Figure 86
Communication Between Dial Peers Sharing the Same Router
Dial peer 3
Source
4085554001
Voice port
Cisco 3600
2/0/0
ATM
27857
2/0/1
Dial peer 1
Destination
4085554000
When configuring VoATM dial peers, an understanding of the relationship between the destination
pattern and the session target is critical. The destination pattern represents the pattern for the device at
the voice connection endpoint, such as a telephone or a PBX. The session target represents the serial port
on the peer router at the other end of the ATM connection. Figure 87 and Figure 88 show the relationship
between the destination pattern and the session target, as seen from the perspective of both routers in a
VoATM configuration.
Figure 87
Relationship Between the Destination Pattern and Session Target from the Perspective of
Router 1
VoATM dial-peer
session target
POTS dial-peer
destination pattern
388…
VoATM dial-peer
destination pattern
Router
No. 2
Serial
port
Voice
port
389…
ATM
Router
No. 1
Figure 88
388…
Serial port
Voice port
27858
PBX
Relationship Between the Destination Pattern and Session Target from the Perspective of
Router 2
Voice
port
Router
No. 1
Router
No. 2
Serial
port
389…
ATM
PBX
VoATM dial-peer
destination pattern
VoATM dial-peer
session target
Cisco IOS Voice, Video, and Fax Configuration Guide
VC-424
Voice port
POTS dial-peer
destination pattern
27859
Serial port
Configuring Voice over ATM
VoATM Restrictions
For more information regarding dial peers, see the “Configuring Dial Plans, Dial Peers, and Digit
Manipulation” chapter.
VoATM Restrictions
The following are restrictions regarding V0ATM:
•
VoATM is supported only on the Cisco MC3810 multiservice concentrators ATM port 0
(compressed VoATM). When VoATM is enabled, the channel group, time-division multiplexing
(TDM) group, and channel associated signaling (CAS) functionality are not available on the
multiflex trunk (MFT) because ATM uses all T1/E1 time slots.
•
VoATM on the Cisco 3600 series router requires the installation of one of the following modules:
– Multiport T1/E1 ATM network module with inverse multiplexing over ATM (IMA). The
multiport T1/E1 ATM network module with IMA supports up to eight T1/E1 lines. For more
information, see the Cisco IOS Release 12.0(5)T online document Configuring Multiport T1/E1
ATM Network Modules with Inverse Multiplexing over ATM on Cisco 2600 and 3600 Series
Routers.
– OC3 ATM network module. The OC3 ATM network module supports one OC3 line. For more
information about the Digital T1 packet voice trunk network modules, see the Cisco IOS
Release 12.0(3)T online document ATM OC-3 Network Module for the Cisco 3600 Series
Routers.
•
The following AAL2 capabilities are not supported:
– Data services over AAL2 (Nx64K circuit mode and N>=1)
– Fax/modem relay (fax demodulation and remodulation)
– Idle code detection or idle channel suppression
– Cisco-switched AAL2 trunking
•
Only AAL5 is supported on the Cisco 3600 series routers. AAL2 is not supported.
•
VoATM SVCs are not supported since Cisco IOS Release12.0(7)XK. The ATM SVCs for data are
still supported.
VoATM Prerequisite Tasks
Before configuring VoATM, perform the following tasks:
•
Install the required network modules into the Cisco 3600 series router. For more information, see
the “VoATM Restrictions” section on page 425.
•
Establish a working ATM network. For more information, refer to the Cisco IOS Wide-Area
Networking Configuration Guide.
•
Configure Local Management Interface (LMI) support if the carrier is using LMI because ATM
defaults to Integrated Local Management Interface (ILMI).
•
Configure the clock source for the Cisco MC3810 multiservice concentrator interfaces. For more
information, see the “Configuring Synchronous Clocking on the Cisco MC3810 Multiservice
Concentrators” appendix.
Cisco IOS Voice, Video, and Fax Configuration Guide
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Configuring Voice over ATM
VoATM Configuration Task List
•
Complete your company dial plan and establish a working telephony network based on the plan and:
– Integrate the dial plan and telephony network into the existing ATM network topology. Make
routing and dialing transparent to the user; for example, avoid secondary dial tones from
secondary switches where possible.
– Contact the PBX vendor for instructions about how to reconfigure the appropriate PBX
interfaces.
•
Ensure that the voice ports and dial peers are configured. For more information, see the
“Configuring Voice Ports” and “Configuring Voice Dial Peers, Dial Plans, and Digit Manipulation”
chapters.
VoATM Configuration Task List
To configure VoATM, perform the following tasks:
•
Configuring ATM Interfaces for Voice Traffic Using AAL5, page 426
•
Configuring AAL2 Encapsulation for VoATM, page 429
•
Configuring VoATM Dial Peers, page 433
•
Configuring Dial-Peer Hunting, page 438
•
Configuring Cisco Trunk Permanent Calls, page 439
•
Configuring Cisco Trunk Permanent Calls, page 439
Configuring ATM Interfaces for Voice Traffic Using AAL5
ATM interfaces must be configured for voice traffic using AAL5 and the VoATM configuration must be
performed on both sides of the voice connection. The only commands in ATM virtual circuit
configuration mode that are used for ATM voice PVCs are encapsulation aal5mux voice, vbr-rt, and
ilmi. For more information on the encapsulation command, see the Cisco IOS Wide-Area Networking
Command Reference.
To calculate the minimum peak, average, and burst values for the number of voice calls, perform the
following calculations:
•
Peak value: (2 x the maximum number of calls) x 16 Kb
•
Average value: (1 x the maximum number of calls) x 16 Kb
– The average value correlates to the carrier sustainable cell rate (SCR).
•
Burst value: 4 x the maximum number of calls
– The burst value is the burst size in cells.
Cisco IOS Voice, Video, and Fax Configuration Guide
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Configuring Voice over ATM
VoATM Configuration Task List
To configure ATM interfaces to support voice traffic, use the following commands beginning in global
configuration mode:
Command
Purpose
Step 1
Router(config)# interface ATM slot/number
Enters ATM interface configuration mode.
Step 2
Router(config-if)# pvc [name] vpi/vci [ilmi | qsaal |
smds]
Creates an ATM PVC for voice traffic and enters
virtual circuit configuration mode. The keywords and
arguments are as follows:
•
name—(Optional) Supports up to 16 characters.
•
vpi/—Valid range is from 0 to 255.
•
vci—Valid range is from 0 to 1 less than the
maximum value set for the interface by the atm
vc-per-vp command. Lower values 0 to 31 are
reserved for specific traffic (for example, F4
OAM, ILMI, etc.) and should not be used.
VCI is a 16-bit field in the header of the ATM cell.
The value is unique only on a single link. not
throughout the ATM network, because it has local
significance only.
Note
The vpi and vci arguments cannot both be set
to 0.
•
ilmi—(Optional) Sets up communication with
the ILMI. The vpi and vci values are 0 and 16,
respectively.
•
qsaal—(Optional) Signaling-type PVC used for
setting up or tearing down data SVCs. The
associated vpi and vci values are 0 and 5,
respectively.
The default is that the PVC is not defined. When the
PVC is defined, the global default of the
encapsulation command applies (aal-encap =
aal5snap).
Step 3
Router(config-if-atm-pvc)# encapsulation aal5mux
voice
Sets the encapsulation of the PVC to support AAL5
voice.
Cisco IOS Voice, Video, and Fax Configuration Guide
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Configuring Voice over ATM
VoATM Configuration Task List
Step 4
Command
Purpose
Router(config-if-atm-pvc)# vbr-rt peak-rate
average-rate [burst]
Configures the peak and average rates and burst cell
size to perform traffic shaping between voice and
data PVCs for real-time voice networks. The
arguments are as follows:
•
peak rate—Sets to the line rate if it does not
exceed the carrier allowable rate (for example,
1536 kbps for T1-ATM).
•
average rate—Calculates according to the
maximum number of PVC calls times the
bandwidth per call. The following formulas
calculate the average rate in kbps:
– G.711 with 40- or 80-byte sample size:
maximum calls x 85
– G.726 with 40- or 80-byte sample size:
maximum calls x 43
– G.729 with 30-byte sample size: maximum
calls x 15
– G.729 with 20-byte sample size: maximum
calls x 22
– G.729 with 10-byte sample size: maximum
calls x 43
If VAD is enabled, the bandwidth usage is
reduced by as much as 12 percent with the
maximum number of calls in progress. With
fewer calls in progress, bandwidth is less.
•
burst (Optional)—Sets the burst size as large as
possible and never less than the minimum burst
size. Guidelines are as follows:
– Minimum: number of voice calls x 4.
– Maximum: maximum allowed by the carrier.
Step 5
Router(config-if-atm-pvc)# exit
Exits ATM virtual circuit configuration mode.
Step 6
Router(config-if)# pvc [name] vpi/vci
Creates an ATM PVC for data traffic and enters
virtual circuit configuration mode.
Step 7
Router(config-if-atm-pvc)# encapsulation aal5snap
Sets the encapsulation of the PVC to support ATM
data traffic. In ATM PVC configuration mode,
configure the ubr, ubr+ or the vbr-nrt traffic
shaping commands for the data PVC as appropriate.
Note
Step 8
Router(config-if-atm-pvc)# exit
Cisco IOS Voice, Video, and Fax Configuration Guide
VC-428
Calculate the overhead as voice rate x 1.13.
See the Cisco IOS Wide-Area Network
Configuration Guide for more information.
Exits ATM virtual circuit configuration mode. Repeat
Steps 6 and 7 for each data PVC configured.
Configuring Voice over ATM
VoATM Configuration Task List
Verifying the ATM PVC Configuration
Verify the ATM PVC configuration by using the show atm vc command. To verify connectivity, do not
use the ping command over a voice PVC because the command applies to data only. Use the ping
command over the data PVC to verify that the data and voice PVCs are set to the same destination.
Configuring AAL2 Encapsulation for VoATM
AAL2 encapsulation for VoATM must be configured and the VoATM configuration must be performed
on the Cisco MC3810 multiservice concentrators at both ends of the ATM link. AAL2 is not supported
on the Cisco 3600 series routers.
Note
If any DS0 groups (CAS groups), channel groups, or clear channels are configured on T1/E1
controller 0, remove them before configuring VoATM. Because ATM uses all the DS0 timeslots on
the controller, the ATM configuration cannot take place if any DS0s on controller 0 are used by other
applications.
To configure AAL2 encapsulation for VoATM, perform the following tasks:
•
Configuring T1/E1 Trunks, page 429
•
Configuring Call Admission Control, page 431
•
Configuring Subcell Multiplexing, page 432
Configuring T1/E1 Trunks
To configure the T1/E1 trunk, use the following commands beginning in global configuration mode:
Step 1
Command
Purpose
Router(config)# controller {t1 | e1} 0
Selects the T1 or E1 controller 0.
Note
Step 2
Step 3
Router(config-controller)# mode atm
Router(config-controller)# no shutdown
On the Cisco MC3810 multiservice concentrator,
ATM is supported only on controller 0.
Specifies controller support for ATM encapsulation and
creates ATM interface 0. When the controller is set to ATM
mode, the following takes place:
•
Controller framing is automatically set to Extended
SuperFrame (ESF) on T1 and to CRC4 on E1.
•
The linecode is automatically set to B8ZS on T1 and to
HDB3 on E1.
Ensures that the controller is activated.
Cisco IOS Voice, Video, and Fax Configuration Guide
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Configuring Voice over ATM
VoATM Configuration Task List
Step 4
Step 5
Command
Purpose
Router(config)# interface atm0
[subinterface-number [multipoint |
point-to-point]]
Enters interface configuration mode to configure ATM
interface 0 or an ATM subinterface. The keywords and
arguments are as follows:
Router(config-if)# pvc [name] {vpi/vci | vci}
•
subinterface-number—Configures the subinterface.
Valid range is from 1 to 4294967293.
•
multipoint (Optional)—Assumes that is a fully
meshed network. This is the default setting.
•
point-to-point (Optional)—Specifies the VoATM
connection over point-to-point network.
Creates an ATM PVC for voice traffic and enters ATM
virtual circuit configuration mode.
Note
Step 6
Router(config-if-atm-pvc)# encapsulation aal2
The ilmi and qsaal options are not supported for
AAL2.
Sets the PVC encapsulation to support AAL2 voice traffic.
This automatically creates channel identifiers (CIDs) 1
through 255.
or
Router(config-if-atm-pvc)# ip address ip-address
mask
Assigns the IP address and subnet mask to the interface on
the Cisco MC3810 multiservice concentrator.
and
Step 7
Router(config-if-atm-pvc)# encapsulation
aal5mux
Sets the encapsulation of the PVC to support voice traffic
on the Cisco MC3810 multiservice concentrator.
Router(config-if-atm-pvc)# vbr-rt peak-rate
average-rate [burst]
Configures the VBR for real-time voice traffic.
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Configuring Voice over ATM
VoATM Configuration Task List
Step 8
Command
Purpose
Router(config-if-atm-pvc)# oam-pvc [manage]
[frequency]
(Optional) Configures transmission of end-to-end F5
operation, administration, and maintenance (OAM)
loopback cells on a PVC; specifies the number of seconds
between loopback cells; and enables OAM management of
the connection. The keyword and argument are as follows:
•
manage—(Optional) Enables OAM management.
•
frequency (Optional)—Valid range is 0 to 600. The
default is 10.
Note
Step 9
Router(config-if-atm-pvc)# oam retry up-count
down-count retry-frequency
The oam-pvc command does not apply to AAL2.
(Optional) Specifies OAM management parameters for
verifying connectivity of a PVC connection. This command
is supported only if OAM management is enabled. The
arguments are as follows:
•
up-count—Number of OAM loopback cell responses
received to change the PVC connection to up. The
range is from 1 to 600; the default is 3.
•
down-count—Number of OAM loopback cell
responses not received to change the PVC connection
to down. The range is from 1 to 600; the default is 5.
•
retry-frequency—Number of seconds between
loopback cells sent to verify the down state of a PVC.
The range is from 1 to 1000; the default is 1.
Note
Enter the oam retry command only once with all of
the arguments in the order shown. The first number
always specifies up-count; the second, down-count;
and the third, retry-frequency.
Note
The oam retry command does not apply to AAL2.
Configuring Call Admission Control
Configuring the call admission control (CAC) is optional for the Cisco MC3810 multiservice
concentrator because the MC3810 multiservice concentrator can be configured as master or slave. By
default, a Cisco MC3810 multiservice concentrator is a CAC slave.
Typically the ATM trunk is configured with the CAC master at one end (performing CAC during
fax/modem up speed) and slave at the opposite end. When the Cisco MC3810 multiservice concentrator
is configured as a slave, it sends a request for CAC to the CAC master.
To configure a Cisco MC3810 multiservice concentrator as a CAC master, usethe following commands
beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# voice service voatm
Enters voice-service configuration mode.
Step 2
Router(config-voice-service)# session protocol aal2
Enters voice-service-session configuration mode and
specifies AAL2 trunking.
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Configuring Voice over ATM
VoATM Configuration Task List
Command
Purpose
Step 3
Router(config-voice-service-session)# cac master
Configures the Cisco MC3810 multiservice
concentrator as a CAC master. Default is that the
concentrator acts as a CAC slave.
Step 4
Router(config-voice-service-session)# exit
Exits voice-service session configuration mode. To
return to global configuration mode, enter the exit
command again.
To return a Cisco MC3810 multiservice concentrator to its default operation as a CAC slave, use the
following commands beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# voice service voatm
Enters voice-service configuration mode.
Step 2
Router(config-voice-service)# session protocol aal2
Enters voice-service-session configuration mode and
specifies AAL2 trunking.
Step 3
Router(config-voice-service-session)# no cac master
Configures this Cisco MC3810 multiservice
concentrator as a CAC slave.
Step 4
Router(config-voice-service-session)# exit
Exits voice-service session configuration mode. To
return to global configuration mode, enter the exit
command again.
Configuring Subcell Multiplexing
This section describes the configuration tasks necessary to enable AAL2 common part sublayer (CPS)
subcell multiplexing when the Cisco MC3810 multiservice concentrator interoperates with a voice
interface service module (VISM) in an MGX switch. The commands and procedures in this section are
specific to the Cisco MC3810 multiservice concentrator.
To configure the Cisco MC3810 multiservice concentrator to perform subcell multiplexing, use the
following commands beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# voice service voatm
Enters voice-service configuration mode.
Step 2
Router(config-voice-service)# session protocol aal2
Enters voice-service-session configuration mode and
specifies AAL2 trunking.
Step 3
Router(config-voice-service-session)# subcell-mux
Enables subcell multiplexing. By default, subcell
multiplexing is not enabled.
Step 4
Router(config-voice-service-session)# exit
Exits voice-service session configuration mode. To
return to global configuration mode, enter the exit
command again.
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Configuring VoATM Dial Peers
Configuring dial peers to support VoATM should be performed in a back-to-back configuration before
separating them across the ATM network. The back-to-back configuration enables the testing of a voice
connection. If a voice connection cannot be made after both peers are placed in the network, then you
have a network problem. For information about configuring POTS dial peers, see the “Configuring Dial
Plans, Dial Peers, and Digit Manipulation” chapter.
To configure VoATM dial peers, use the following commands beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# dial-peer voice number voatm
Defines a VoATM dial peer and enters dial-peer
configuration mode. The number argument identifies
the dial peer. Do not duplicate a specific number.
Step 2
Router(config-dial-peer)# destination-pattern string
Configures the destination pattern. The special
characters are as follows: The string values are as
follows:
Step 3
Router(config-dial-peer)# session target ATM x/y pvc
{name | vpi/vci | vci}
•
Asterisk (*) and pound sign (#) that appear on
standard touch-tone dial pads.
•
Comma (,) can be used only in prefixes and
inserts a one-second pause.
•
Period (.) can be entered as a wildcard digit.
Network dial peers typically use wildcards to
represent a range of destination telephone
numbers (for example, 1408555... for all
numbers in area code 408 with a 555 prefix).
•
Timer (T) can be used to configure
variable-length dial plans.
Configures the ATM session target. On the
Cisco 3600, if a vpi/vci combination is specified, the
valid values depend on the network module installed,
as follows:
•
For multiport T1/E1 ATM with IMA, the valid
ranges are:
– vpi is from 0 to 15
– vci is from 1 to 255
•
For OC3 ATM, the valid ranges are:
– vpi is from 0 to 15
– vci is from 1 to 1023
Step 4
Router(config-dial-peer)# preference value
(Optional) Configures a preference. The value
argument has a valid range is from 0 to 10 (the lower
the number, the higher the preference).
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Step 5
Command
Purpose
Router(config-dial-peer)# codec type [bytes
payload_size]
Specifies the rate of speech and payload size. The
default codec is g729r8. The keyword and arguments
are as follows:
•
type—Assigns codec values to the voice port for
regular switched voice calls.
•
bytes—(Optional) Specifies the payload size.
Each codec type defaults to a different payload
size if one is not specified.
•
payload_size—(Optional) Specifies the payload
size by entering the bytes. Each codec type
defaults to a different payload size if a value is
not specified.
Note
To obtain a list of the default payload sizes,
enter the codec command and the bytes
option followed by a question mark (?).
Step 6
Router(config-dial-peer)# dtmf-relay
(Optional) Specifies support for dual tone
multifrequency (DTMF) relay. If the codec type is a
low bit-rate codec such as g729 or g723, the
end-to-end transport of DTMF tones is improved.
DTMF tones do not always propagate reliably with
low bit-rate codecs. DTMF relay is disabled by
default.
Step 7
Router(config-dial-peer)# signal-type {cas | cept |
ext-signal | transparent}
(Optional) Defines the ABCD signaling packets that
are generated by the voice port and sent to the data
network. The signal type must be configured to the
same setting at both ends of the permanent voice call.
The keywords are as follows:
•
cas—Support for CAS.
•
cept—Support for the European CEPT standard
(related to Mercury Exchange Limited (MEL)
CAS).
•
ext-signal—Indicates that ABCD signaling
packets should not be sent for configurations in
which the line signaling information is carried
externally to the voice port.
•
transparent—(for digital T1/E1 interfaces)
Reads the ABCD signaling bits directly from the
T1/E1 interface without interpretation and
transparently passes them to the data network.
Also known as transparent FRF.11 signaling.
Step 8
Router(config-dial-peer)# no vad
(Optional) Disables voice activity detection (VAD).
This command is enabled by default.
Step 9
Router(config-dial-peer)# sequence-numbers
(Optional) Enables the voice sequence number if
required. This command is disabled by default.
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Command
Purpose
Step 10
Router(config-dial-peer)# preference value
(Optional) Configures a preference for the VoATM
dial peer. The value argument has valid ranges from 0
to 10 (the lower the number, the higher the preference
in hunt groups).
Step 11
Router(config-dial-peer)# session protocol
cisco-switched
(Optional) Configures the session protocol to support
Cisco-trunk permanent trunk calls. The
cisco-switched keyword is the default setting and is
not required.
Note
Step 12
Router(config-dial-peer)# exit
Use the no session protocol cisco-switched
command if the dial peer does not support
Cisco trunk calls.
Exits dial-peer configuration mode. Repeat the steps
to configure each dial peer.
Configuring VoATM Dial Peers to Support AAL2
To configure the voice network dial peers to support AAL2 on a Cisco MC3810 multiservice
concentrator, use the following commands beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# dial-peer voice number voatm
Defines the dial peer and enters dial-peer
configuration mode.
Step 2
Router(config-dial-peer)# destination-pattern string
Configures the destination pattern.
Step 3
Router(config-dial-peer)# session protocol aal2-trunk
Configures the session protocol to support
AAL2-trunk permanent trunk calls.
Step 4
Router(config-dial-peer)# session target atm 0 pvc
{name | vpi/vci | vci}
Configures the ATM session target for the dial peer.
Be sure to specify atm 0 as the interface for the PVC.
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Step 5
Command
Purpose
Router(config-dial-peer)# codec aal2 profile {itut |
custom} profile-number codec
Specifies a codec profile for the DSP. Use this
command instead of the codec (dial-peer) command
for AAL2 trunk applications. The keywords and
arguments are as follows:
•
itut—Specifies the profile-number as an ITU-T
type:
– 1: G.711ulaw
– 2: G.711ulaw with silence insertion
descriptor (SID)
– 7: G.711ulaw and G.729ar8
•
custom—Specifies the profile-number as a
custom type:
– 100: G.711ulaw and G.726r32
– 110: G.711ulaw, G.726r32, and G.729ar8
•
profile-number—The available profile-number
selections depend on the profile type.
•
codec—Enter one codec for the domain specific
part (DSP). The possible codec entries depend on
the profile-number. The valid entries are as
follows:
– For ITU 1: g711ulaw
– For ITU 2: g711ulaw
– For ITU 7: g711ulaw or g729ar8
– For custom 100: g711ulaw or g726r32
– For custom 110: g711ulaw or g726r32 or
g729ar8
See the Cisco IOS Voice, Video, and Fax Command
Reference for the codec options available for each
AAL2 profile.
Step 6
Router(config-dial-peer)# dtmf-relay
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(Optional) Specifies support for DTMF relay to
improve end-to-end transport of DTMF tones if the
codec type is a low bit-rate codec such as g729 or
g723. DTMF tones do not always propagate reliably
with low bit-rate codecs. DTMF relay is disabled by
default.
Configuring Voice over ATM
VoATM Configuration Task List
Step 7
Command
Purpose
Router(config-dial-peer)# signal-type {ext-signal |
transparent}
(Optional) Defines the type of ABCD signaling
packets that are generated by the voice port and sent
over the ATM network. The signal type must be
configured to the same setting at both ends of the
PVC. The keywords are as follows:
•
ext-signal—Identifies common-channel
signaling (CCS). ABCD signaling packets are
not sent.
•
transparent—Identifies nonswitched trunks
using channel associated signaling (CAS).
ABCD signaling bits are passed transparently to
the ATM network.
Step 8
Router(config-dial-peer)# no vad
(Optional) Disables VAD on the dial peer. VAD is
enabled by default.
Step 9
Router(config-dial-peer)# exit
Exits dial-peer configuration mode. Repeat the steps
to configure each dial peer.
Configuring VoATM Dial Peers for Cisco Trunk Calls
If Cisco trunk calls are transmitted over ATM, the dial peers must be configured to specifically support
the calls. Cisco trunk calls are permanent calls.
Note
A voice class to configure trunk conditioning values for the idle and out-of-service (OOS) states can
be configured with the voice class assigned to the VoATM dial peer. For more information, see the
“Configuring Trunk Management Features” chapter.
To configure a VoATM dial peer to support Cisco trunk calls, use the following commands beginning in
global configuration mode:
Step 1
Command
Purpose
Router(config)# dial-peer voice number voatm
Defines a VoATM dial peer and enters dial-peer
configuration mode.
Note
Step 2
Router(config-dial-peer)# session protocol
cisco-switched
The VoATM dial peers must already be
configured.
Configures the session protocol to support Cisco
trunk calls.
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Configuring Dial-Peer Hunting
To configure dial-peer hunting, use the following commands in global configuration mode:
Step 1
Command
Purpose
Router(config)# dial-peer hunt hunt-order-number
Specifies the hunt selection order for dial peers. The
hunt-order-number has valid ranges from 0 to 7 as
follows:
•
0: Longest match in phone number, explicit
preference, random selection. This is the default
hunt order number.
•
1: Longest match in phone number, explicit
preference, least recent use.
•
2: Explicit preference, longest match in phone
number, random selection.
•
3: Explicit preference, longest match in phone
number, least recent use.
•
4: Least recent use, longest match in phone
number, explicit preference.
•
5: Least recent use, explicit preference, longest
match in phone number.
•
6: Random selection.
•
7: Least recent use.
The default is the longest match in a phone number,
explicit preference, and random selection (hunt order
number 0).
Step 2
Router(config)# dial-peer terminator character
(Optional) Designates a special character for variable
length dialed numbers. The character argument has
valid numbers and characters that are as follows:
•
Pound sign (#)
•
Asterisk (*)
•
Numbers from zero to nine
•
Letters from a to d
The default is #.
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To disable dial-peer hunting, use the following commands beginning in global configuration mode:
Command
Purpose
Step 1
Router(config)# dial-peer voice number {pots |
voatm}
Enters dial-peer configuration mode for the specified
dial peer.
Step 2
Router(config-dial-peer)# huntstop
Disables dial-peer hunting on the dial peer. No further
hunting is enabled if a call fails on the specified dial
peer.
Note
To reenable dial-peer hunting on a dial peer,
enter the no huntstop command.
Configuring Cisco Trunk Permanent Calls
The Cisco trunk call functionality provides true permanent, private-line connections; comprehensive
busyout support for trunk connection; and transparent CAS protocol transport to allow the trunk to carry
arbitrary ABCD signaling protocols. Conversion from North American signaling protocols to CEPT
(Conférence Européenne des Postes et des Télécommunications) signaling protocols used for European
voice networks and remote analog to digital channel-bank operation for converting from DVM to AVM
configurations is also provided.
To configure Cisco-trunk permanent calls, use the following commands beginning in global
configuration mode:
Step 1
Command
Purpose
Router(config)# voice-port
Enters voice-port configuration mode.
Note
The voice-port command is hardware
specific. Refer to the Cisco IOS Voice, Video,
and Fax Command Reference for more
information.
Step 2
Router(config-voiceport)# connection trunk
destination-string [answer-mode]
Configures the trunk connection, specifying the
telephone number in the destination-string argument.
The answer-mode keyword specifies that the voice
port should operate in slave mode. The default is
master mode.
Step 3
Router(config-voiceport)# shutdown
Shuts down the voice port.
Step 4
Router(config-voiceport)# no shutdown
Reactivates the voice port to enable the trunk
connection to take effect.
Note
When the connection trunk or no connection trunk command is entered, the voice port must be
toggled by entering shutdown, and then no shutdown before the changes take effect.
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Verifying the Voice Connection
To verify that the voice connection is working, perform the following steps:
Step 1
Pick up the telephone handset and verify that a dial tone is present.
Step 2
Make a call from the local telephone to a configured dial peer and verify that the call attempt is
successful.
Step 3
Use the show dial-peer voice command to verify that the configured data is correct.
Step 4
Use the show voice port command to show the status of the voice ports.
Step 5
Use the show voice call command to show the call status for all voice ports.
Step 6
Use the show voice dsp command to show the current status of all DSP voice channels.
Troubleshooting Tips
To resolve suspected problems, perform the following tasks:
Step 1
Use the show dial-peer voice command on the local and remote concentrators to verify that the data is
configured correctly on both.
Step 2
Use the show interface command to verify that the ATM interface is up.
Step 3
Ensure that the voice port, serial port, and controller T1 0 is set to no shutdown.
Note
ATM defaults to Interim Local Management Interface (ILMI). If the carrier is using LMI, be sure to
configure LMI support on the router.
Verifying the ATM Interface Configuration
To verify the ATM interface configuration, perform the following tasks:
•
Enter the privileged EXEC show atm vc command to view the SVCs (data only) and PVCs set. The
following is a sample output:
Router# show atm vc
VCD /
Interface
0
0
0
0
0
0
0
0
0
0
Name
1
2
379
986
14
15
16
17
18
19
VPI
0
0
0
0
0
0
0
0
0
0
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VCI
5
16
60
84
133
134
135
136
137
138
Type
PVC
PVC
SVC
SVC
SVC
SVC
SVC
SVC
SVC
SVC
Peak
Encaps
SAAL
ILMI
SNAP
SNAP
VOICE
VOICE
VOICE
VOICE
VOICE
VOICE
Avg/Min Burst
SC
Kbps
Kbps
Cells
UBR
0
UBR
0
UBR
0
UBR
0
VBR
64
16
10
VBR
64
16
10
VBR
64
16
10
VBR
64
16
10
VBR
64
16
10
VBR
64
16
10
Sts
UP
UP
UP
UP
UP
UP
UP
UP
UP
UP
Configuring Voice over ATM
VoATM Configuration Task List
0
0
0
0
0
0
0
0
0
Note
•
20
21
22
23
24
25
26
27
28
0
0
0
0
0
0
0
0
0
139
140
141
142
143
144
145
146
147
SVC
SVC
SVC
SVC
SVC
SVC
SVC
SVC
SVC
VOICE
VOICE
VOICE
VOICE
VOICE
VOICE
VOICE
VOICE
VOICE
VBR
VBR
VBR
VBR
VBR
VBR
VBR
VBR
VBR
64
64
64
64
64
64
64
64
64
16
16
16
16
16
16
16
16
16
10
10
10
10
10
10
10
10
10
UP
UP
UP
UP
UP
UP
UP
UP
UP
VoATM SVCs are not supported since Cisco IOS Release12.0(7)XK. The ATM SVCs for
data are still supported.
Enter the show atm pvc command with the VPI/VCI specified to view the PVCs that are set up for
ILMI management and Q.SAAL signaling. The following is a sample output:
Router# show atm pvc 0/5
ATM0: VCD: 2, VPI: 0, VCI: 5, Connection Name: SAAL
UBR, PeakRate: 56
AAL5-SAAL, etype:0x4, Flags: 0x26, VCmode: 0x0
OAM frequency: 0 second(s), OAM retry frequency: 1 second(s), OAM retry frequenc
y: 1 second(s)
OAM up retry count: 3, OAM down retry count: 5
OAM Loopback status: OAM Disabled
OAM VC state: Not Managed
ILMI VC state: Not Managed
InARP DISABLED
InPkts: 2044, OutPkts: 2064, InBytes: 20412, OutBytes: 20580
InPRoc: 2044, OutPRoc: 2064, Broadcasts: 0
InFast: 0, OutFast: 0, InAS: 0, OutAS: 0
OAM cells received: 0
F5 InEndloop: 0, F5 InSegloop: 0, F5 InAIS: 0, F5 InRDI: 0
F4 InEndloop: 0, F4 InSegloop: 0, F4 InAIS: 0, F4 InRDI: 0
OAM cells sent: 0
F5 OutEndloop: 0, F5 OutSegloop: 0, F5 OutRDI: 0
F4 OutEndloop: 0, F4 OutSegloop: 0, F4 OutRDI: 0
OAM cell drops: 0
Compress: Disabled
Status: INACTIVE, State: NOT_IN_SERVICE
!
Router# show atm pvc 0/16
ATM0: VCD: 1, VPI: 0, VCI: 16, Connection Name: ILMI
UBR, PeakRate: 56
AAL5-ILMI, etype:0x0, Flags: 0x27, VCmode: 0x0
OAM frequency: 0 second(s), OAM retry frequency: 1 second(s), OAM retry frequenc
y: 1 second(s)
OAM up retry count: 3, OAM down retry count: 5
OAM Loopback status: OAM Disabled
OAM VC state: Not Managed
ILMI VC state: Not Managed
InARP DISABLED
InPkts: 398, OutPkts: 421, InBytes: 30493, OutBytes: 27227
InPRoc: 398, OutPRoc: 421, Broadcasts: 0
InFast: 0, OutFast: 0, InAS: 0, OutAS: 0
OAM cells received: 0
F5 InEndloop: 0, F5 InSegloop: 0, F5 InAIS: 0, F5 InRDI: 0
F4 InEndloop: 0, F4 InSegloop: 0, F4 InAIS: 0, F4 InRDI: 0
OAM cells sent: 0
F5 OutEndloop: 0, F5 OutSegloop: 0, F5 OutRDI: 0
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F4 OutEndloop: 0, F4 OutSegloop: 0, F4 OutRDI: 0
OAM cell drops: 0
Compress: Disabled
Status: INACTIVE, State: NOT_IN_SERVICE
•
Enter the show atm interface command in privileged EXEC mode and specify ATM 0 to display
the ATM interface. The following is a sample output:
Router# show interface atm 0
ATM0 is up, line protocol is up
Hardware is PQUICC Atom1
Internet address is 9.1.1.6/8
MTU 1500 bytes, sub MTU 1500, BW 1536 Kbit, DLY 20000 usec,
reliability 255/255, txload 22/255, rxload 11/255
NSAP address: 47.0091810000000002F26D4901.000011116666.06
Encapsulation ATM
292553397 packets input, -386762809 bytes
164906758 packets output, 1937663833 bytes
0 OAM cells input, 0 OAM cells output, loopback not set
Keepalive not supported
Encapsulation(s):, PVC mode
1024 maximum active VCs, 28 current VCCs
VC idle disconnect time: 300 seconds
Signalling vc = 1, vpi = 0, vci = 5
UNI Version = 4.0, Link Side = user
Last input 00:00:00, output 2d05h, output hang never
Last clearing of "show interface" counters never
Input queue: -1902/75/0 (size/max/drops); Total output drops: 205
Queueing strategy: weighted fair
Output queue: 0/1000/64/0 (size/max total/threshold/drops)
Conversations 0/0/256 (active/max active/max total)
Reserved Conversations 0/0 (allocated/max allocated)
5 minute input rate 67000 bits/sec, 273 packets/sec
5 minute output rate 136000 bits/sec, 548 packets/sec
76766014 packets input, 936995443 bytes, 0 no buffer
Received 0 broadcasts, 0 runts, 0 giants, 0 throttles
0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort
367264676 packets output, 3261882795 bytes, 0 underruns
0 output errors, 0 collisions, 2 interface resets
0 output buffer failures, 0 output buffers swapped out
•
Enter the show atm video-voice address privileged EXEC command to display the ATM interface
address and confirm the ILMI status (ILMI PVC is set up to enable SVC management). The ATM
interface is assigned automatically with the atm voice aesa command. The following is a sample
output:
Router# show atm video-voice address
nsap address
47.0091810000000002F26D4901.00107B4832E1.FE
47.0091810000000002F26D4901.00107B4832E1.C8
type
VOICE_AAL5
VIDEO_AAL1
ilmi status
Confirmed
Confirmed
Verifying the VoATM Connection
Verify that the voice connection is working by performing the following steps:
Step 1
Pick up the handset on a telephone connected to the configuration and verify that there is dial tone.
Step 2
Make a call from the local telephone to a configured dial peer to verify the connection.
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Step 3
Check the validity of the dial-peer and voice-port configuration by performing the following tasks:
•
If there are relatively few dial peers configured, use the show dial-peer voice command to verify
that the data configured is correct.
•
To show the status of the voice ports, use the show voice port command.
•
To show the call status for all voice ports, use the show voice call command.
•
To show the current status of all DSP voice channels, use the show voice dsp command.
Troubleshooting Tips
If a call does not connect, resolve the problem by performing the following tasks:
•
Verify dial peer configuration by using the show dial-peer voice command on the local and remote
concentrators.
•
Verify that ATM interface 0 is up by using the show interface command.
•
Ensure that the voice port, serial port, and controller T1 0 are set to no shutdown.
VoATM Configuration Examples
Configuration examples for VoATM are shown in the following sections:
•
Back-to-Back VoATM PVCs Example, page 443
•
Voice and Data Traffic over ATM PVCs Example, page 444
•
VoATM for Cisco 3600 Series Routers Configuration Example, page 447
•
VoATM for the Cisco MC3810 Multiservice Concentrator Configuration Example, page 451
Back-to-Back VoATM PVCs Example
Figure 89 shows a configuration example for two Cisco MC3810 multiservice concentrators configured
back-to-back, with VoATM configured for both concentrators. This setup is a useful for testing the
VoATM configuration locally to ensure that voice connections can be made before configuring VoATM
across a larger network. Following the figure are the commands required for configuring the Cisco
MC3810 multiservice concentrators in this example.
Back-to-Back VoATM PVCs Configuration
Cisco
MC3810 No. 1
Cisco
MC3810 No. 2
Back-to-back
ATM T1
ATM0
ATM0
10BaseT
PC1
10BaseT
PC2
S6923
Figure 89
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Cisco MC3810 Multiservice Concentrator 1
Cisco MC3810 Multiservice Concentrator 2
hostname location1
no ip domain-lookup
!
interface Ethernet0
ip address 10.1.10.1 255.255.255.0
no ip mroute-cache
no ip route-cache
!
hostname location2
no ip domain-lookup
!
interface Ethernet0
ip address 10.1.20.1 255.255.255.0
no ip mroute-cache
no ip route-cache
!
controller T1 0
clock source internal
mode atm
!
interface atm0 point-to-point
ip address 10.1.1.1 255.255.255.0
no ip mroute-cache
!
controller T1 0
clock source line
mode atm
!
interface atm0 point-to-point
ip address 10.1.1.2 255.255.255.0
no ip mroute-cache
!
pvc 1 1 100
encapsulation aal5mux voice
vbr-rt 384 192 48
!
pvc 1 1 100
encapsulation aal5mux voice
vbr-rt 384 192 48
!
pvc 2 1 200
encapsulation aal5snap
map-group atm1
!
router rip
redistribute connected
network 10.0.0.0
!
no ip classless
!
pvc 2 1 200
encapsulation aal5snap
map-group atm1
!
router rip
redistribute connected
network 10.0.0.0
!
no ip classless
!
map-list atm1
ip 10.1.1.2 atm pvc 2 broadcast
!
dial-peer voice 1 pots
destination-pattern 10
port 1/1
!
dial-peer voice 202 voatm
destination-pattern 2.
session target ATM0 1
!
end
map-list atm1
ip 10.1.1.1 atm pvc 2 broadcast
!
dial-peer voice 1 pots
destination-pattern 20
port 1/1
!
dial-peer voice 202 voatm
destination-pattern 1.
session target ATM0 1
!
end
Voice and Data Traffic over ATM PVCs Example
Figure 90 shows an example for both voice and data traffic over ATM between two Cisco MC3810
multiservice concentrators, including configuration for voice ports and dial peers. Following the figure
are the commands required for configuring the Cisco MC3810 multiservice concentrators in this
example.
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3488801
3388801
Voice port
1/1
Voice port
1/1
ATM0
3488802
ATM
ATM PVC 27
1/2 Cisco
MC3810 No. 1
ATM0
Cisco 1/2
MC3810 No. 2
3388802
S6924
Figure 90
Cisco MC3810 Multiservice Concentrator 1
Cisco MC3810 Multiservice Concentrator 2
interface Ethernet0
ip address 172.22.124.239 255.255.0.0
!
controller T1 0
mode ATM
!
interface atm0 point-to-point
ip address 223.223.224.229 255.255.255.0
no ip mroute-cache
no ip route-cache
map-group atm1
!
interface Ethernet0
ip address 172.22.124.247 255.255.0.0
!
controller T1 0
mode ATM
!
interface atm0 point-to-point
ip address 223.223.224.228 255.255.255.0
no ip mroute-cache
no ip route-cache
map-group atm1
!
pvc 26 26 200
encapsulation aal5snap
!
pvc 27 27 270
encapsulation aal5mux voice
vbr-rt 384 192 48
!
no ip classless
!
pvc 26 26 200
encapsulation aal5snap
!
pvc 27 27 270
encapsulation aal5mux voice
vbr-rt 384 192 48
!
no ip classless
!
Cisco IOS Voice, Video, and Fax Configuration Guide
VC-445
Configuring Voice over ATM
VoATM Configuration Examples
map-list atm1
ip 223.223.224.228 atm pvc 26 broadcast
!
voice-port 1/1
!
voice-port 1/2
!
voice-port 1/3
!
voice-port 1/4
!
map-list atm1
ip 223.223.224.229 atm pvc 26 broadcast
!
login
line vty 1 4
login
!
voice-port 1/1
!
voice-port 1/2
!
voice-port 1/3
!
voice-port 1/4
!
dial-peer voice 1 pots
destination-pattern 3488801
port 1/1
!
dial-peer voice 2 pots
destination-pattern 3488802
port 1/2
!
end
dial-peer voice 1 pots
destination-pattern 3388801
port 1/1
!
dial-peer voice 2 pots
destination-pattern 3388802
port 1/2
!
dial-peer voice 1001 voatm
destina