Cisco Unified Border Element Fundamentals and Basic Setup 12.4T

Cisco Unified Border Element Fundamentals and Basic Setup 12.4T
Cisco Unified Border
Element Fundamentals and Basic Setup
Configuration Guide, Cisco IOS Release
12.4T
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CONTENTS
Cisco Unified Border Element Fundamentals and Basic Setup 1
Finding Feature Information 1
Getting Started with Important Concepts 1
Prerequisites for Cisco Unified Border Element 2
Restrictions for Cisco Unified Border Element 2
Information About Cisco Unified Border Element 2
Gateway Functionality 3
Cisco Unified Border Element Network Topology 3
Lawful Intercept Support 5
Basic SIP-to-SIP Set-up and Functionality Features 5
Configuring Call Rate Monitoring 6
IP-to-IP Gateway SIP-to-SIP Basic Functionality 9
Finding Feature Information 9
Prerequisites for IP-to-IP Gateway SIP-to-SIP Basic Functionality 10
Restrictions for IP-to-IP Gateway SIP-to-SIP Basic Functionality 10
How to Configure SIP-to-SIP Connections in a Cisco Unified Border Element Enterprise 10
Feature Information for IP-to-IP Gateway SIP-to-SIP Basic Functionality 11
SIP-to-SIP Extended Feature Functionality for Session Border Controllers 13
Finding Feature Information 13
Prerequisites for SIP-to-SIP Extended Feature Functionality for Session Border Controllers 14
Modem Passthrough over VoIP 14
Prerequisites for the Modem Passthrough over VoIP Feature 15
Restrictions for the Modem Passthrough over VoIP Feature 16
How to Configure Modem Passthrough over VoIP 16
Configuring Modem Passthrough over VoIP Globally 17
Configuring Modem Passthrough over VoIP for a Specific Dial Peer 18
Verifying Modem Passthrough over VoIP 20
Troubleshooting Tips 21
Monitoring and Maintaining Modem Passthrough over VoIP 21
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
iii
Contents
Configuration Examples 21
Feature Information for SIP-to-SIP Extended Feature Functionality for Session Border
Controllers 23
SIP Gateway Support for the bind Command 25
Finding Feature Information 25
Prerequisites for SIP Gateway Support for the bind Command 25
Information About SIP Gateway Support for the bind Command 26
How to Configure SIP Gateway Support for the bind Command 27
Setting the Bind Address 27
Setting a Source IP Address for Signaling and Media Packets 28
Verifying and Troubleshooting Tips 31
Verifying a Bound IP Address 31
Verifying Bind Status 31
Configuration Examples for SIP Gateway Support for the bind Command 32
SIP Gateway Support for the bind Command Example 32
Feature Information for SIP Gateway Support for the bind Command 32
SIP Video Calls with Flow Around Media 35
Finding Feature Information 35
Prerequisites for SIP Video Calls with Flow Around Media 35
Restrictions for SIP Video Calls with Flow Around Media 35
How to Configure Support for SIP Video Calls with Flow Around Media 36
Feature Information for Support for SIP Video Calls with Flow Around Media 36
Additional References 39
Related Documents 39
Standards 40
MIBs 40
RFCs 41
Technical Assistance 42
Glossary 43
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
iv
Cisco Unified Border Element Fundamentals
and Basic Setup
This Cisco Unified Border Element is a special Cisco IOS software image that provides a network-tonetwork interface point for billing, security, call admission control, quality of service, and signaling
interworking. This chapter describes basic gateway functionality, software images, topology, and
summarizes supported features.
Cisco Product Authorization Key (PAK)--A Product Authorization Key (PAK) is required to configure
some of the features described in this guide. Before you start the configuration process, please register
your products and activate your PAK at the following URL http://www.cisco.com/go/license.
•
•
•
•
Finding Feature Information, page 1
Getting Started with Important Concepts, page 1
Basic SIP-to-SIP Set-up and Functionality Features, page 5
Configuring Call Rate Monitoring, page 6
Finding Feature Information
Your software release may not support all the features documented in this module. For the latest feature
information and caveats, see the release notes for your platform and software release. To find information
about the features documented in this module, and to see a list of the releases in which each feature is
supported, see the Feature Information Table at the end of this document.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.
Getting Started with Important Concepts
•
•
•
•
Prerequisites for Cisco Unified Border Element, page 2
Restrictions for Cisco Unified Border Element, page 2
Information About Cisco Unified Border Element, page 2
Lawful Intercept Support, page 5
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
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Prerequisites for Cisco Unified Border Element
Getting Started with Important Concepts
Prerequisites for Cisco Unified Border Element
Cisco Unified Border Element Hardware
•
Install the routers that will serve as session border controllers in your VoIP network.
Cisco Unified Border Element Software
•
•
•
Obtain the appropriate feature license for each router on which you will install an image that supports
the Unified Border Element feature. Additional information on obtaining a feature license can be
found at: http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_data_
sheet09186a00801da698.html
Cisco Product Authorization Key (PAK)--A Product Authorization Key (PAK) is required to configure
some of the features described in this guide. Before you start the configuration process, please register
your products and activate your PAK at the following URL http://www.cisco.com/go/license .
Install the appropriate Cisco IOS image on each router and configure a working VoIP network.
Use Cisco Feature Navigator to find information about platform support and software image support. Cisco
Feature Navigator enables you to determine which Cisco IOS and Catalyst OS software images support a
specific software release, feature set, or platform. To access Cisco Feature Navigator, go to http://
www.cisco.com/go/cfn . An account on Cisco.com is not required.
Restrictions for Cisco Unified Border Element
•
•
•
•
•
•
Cisco Unified Border Elements that require the Registration, Admission, and Status (RAS) protocol
must have a via-zone-enabled gatekeeper or equivalent.
Cisco fax relay is reported as a voice call on an Cisco Unified Border Element. Fax relay is enabled by
default for all systems. No further configuration is needed.
Cisco Unified Border Element supports T.38 fax relay (H.323 Annex D). However, endpoints
configured with Named Signaling Events (NSE) may result in reduced fax transmission quality and are
not supported.
Codec filtering must be based on codec types; filtering based on byte size is not supported.
When a Tcl script is running on an Cisco Unified Border Element, the Cisco Unified Border Element
does not support ringback tone generation.
Transcoding is not supported.
Information About Cisco Unified Border Element
When you configure SIP on a router, the ports on all its interfaces are open by default. This makes the
router vulnerable to malicious attackers who can execute toll fraud across the gateway if the router has a
public IP address and a public switched telephone network (PSTN) connection. To eliminate the threat, you
should bind an interface to private IP address that is not accessible by untrusted hosts. In addition, you
should protect any public or untrusted interface by configuring a firewall or an access control list (ACL) to
prevent unwanted traffic from traversing the router.A Cisco Unified Border Element facilitates connectivity
between independent VoIP networks by enabling SIP and H.323 VoIP and videoconferencing calls from
one IP network to another. This gateway performs most of the same functions of a PSTN-to-IP gateway,
but typically joins two IP call legs, rather than a PSTN and an IP call leg. Media packets can flow either
through the gateway (thus hiding the networks from each other) or around the border element, if so
configured.
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
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Cisco Unified Border Element Fundamentals and Basic Setup
Gateway Functionality
Cisco Unified Border Element is a special Cisco IOS software image that runs on the Cisco AS1000
platform. It provides a network-to-network interface point for billing, security, call admission control,
quality of service, and signaling interworking.
Cisco UBE is designed to meet the interconnection needs of Internet telephony service providers (ITSPs)
and of enterprises. One set of images provides basic interconnection and a second set provides
interconnection through an Open Settlement Protocol (OSP) provider, enabling ITSPs to gain the benefits
of the Cisco Unified Border Element while making use of the routing, billing, and settlement capabilities
offered by OSP-based clearinghouses.
Feature benefits include the following:
•
•
•
Capacity control and improved call routing control using carrier-based routing with the Cisco Unified
Border Element feature and routing traffic through the gateways.
Improved billing and settlement capabilities.
Provides key services at the edge of the network for scalability.
•
•
Gateway Functionality, page 3
Cisco Unified Border Element Network Topology, page 3
Gateway Functionality
Gateways are responsible for the following tasks.
•
•
•
•
•
•
•
•
Media stream handling and speech path integrity
DTMF relay
Fax relay and passthrough
Digit translation and call processing
Dial peers and codec filtering
Carrier ID handling
Gateway-based billing
Termination and re-origination of signaling and media
Cisco Unified Border Element Network Topology
In the current VoIP market, ITSPs who provide wholesale VoIP services use their own IP-to-TDM
gateways to exchange calls with the PSTN. Problems occur when a wholesaler receives a call from an
originating ITSP and decides to terminate the call to another ITSP. Because it does not own the PSTN
gateways, the wholesaler does not receive call setup or release information and therefore cannot bill for the
call. Wholesalers are forced either to forbid these connections, thereby foregoing a potential revenue
source, or to set up the call through a combination of back-to-back IP-to-TDM gateways. This solution
results in reduced quality due to double media coding and decoding, and it wastes TDM port resources.
Cisco Unified Border Element allows the wholesaler to terminate the call from the originating ITSP and
then reoriginate it, thereby providing a point at which accurate call detail records (CDRs) can be collected
for billing.
The superior interconnect capability provided by the Cisco Unified Border Element enables service
providers to conceal their internal network and business relationships while improving call admission
control, flexible routing, and protocol interworking capabilities.
The Cisco Unified Border Element includes the following changes to gateways and gatekeepers to allow
Cisco UBE call legs:
•
Support for H.323-to-H.323, H.323-to-SIP, and SIP-to-SIP connection types
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
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Cisco Unified Border Element Fundamentals and Basic Setup
Cisco Unified Border Element Network Topology
•
•
•
Support for transparent codec on H.323-to-H.323 connection types
Support for H.323 call capacities
Introduction of gatekeeper via-zones. Via-zone is a Cisco term for a zone that contains Cisco Unified
Border Elements and via-zone-enabled gatekeepers. A via-zone-enabled gatekeeper is capable of
recognizing via-zones and sending traffic to via-zone gateways. Cisco via-zone-enabled gatekeepers
include a via-zone command-line interface (CLI) command.
Via-zones are usually located on the edge of an ITSP network and are like a VoIP transfer point, or tandem
zone, where traffic passes through on the way to the remote zone destination. Gateways in this zone
terminate requested calls and reoriginate traffic to its final destination. Via-zone gatekeepers operate as
usual for applications that are not Cisco UBE gatekeepers in via-zones support resource management (for
example, gateway selection and load balancing) using the Capacities field in the H.323 Version 4 RAS
messages.
The figure below shows a simple topology example of the Cisco Unified Border Element using via-zone
gatekeepers.
Figure 1
Cisco Unified Border Element Feature Sample Topology
The gatekeeper in Domain A and the gatekeeper in Domain B are connected to the via-zone gatekeeper.
GK408 and the via-zone gatekeeper exchange Registration, Admission, and Status (RAS) messages for the
originating side. Then the connection is made between the originating gateway and the Cisco Unified
Border Element. The via-zone gatekeeper exchanges RAS messages with GK919 for the terminating side.
If the call is accepted, the Cisco Unified Border Element completes the connection from GW408 to
GW919, and the media flows through the Cisco Unified Border Element.
In a basic call scenario, on receiving a location request (LRQ) message from the originating gatekeeper
(GK408), the via-zone-enabled gatekeeper (GKVIA) processes the message and determines that the call
should be set up using the Cisco Unified Border Element. After the originating gateway receives its
admission confirmation (ACF) message, it sets up the call.
With the Cisco Unified Border Element feature, instead of the originating gateway signaling the
terminating gateway directly, the Cisco Unified Border Element controls the call set-up both the signaling
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
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Lawful Intercept Support
Basic SIP-to-SIP Set-up and Functionality Features
and media channel. The Cisco Unified Border Element is terminating the signaling and media channels, but
the information associated with the media is propagated through to the opposite call leg. This process
allows the endpoints to determine what media channel capabilities to use for the call. When the call is
established, the audio stream flows through the Cisco Unified Border Element, meaning that the gateway
terminates the audio channel on one call leg and then reorginates it to the other leg.
The following scenario illustrates a basic call from the originating gateway to the terminating gateway,
using the Cisco Unified Border Element and gatekeepers.
1 GW408 (the originating gateway) calls someone in the 919 area code, which is serviced by GW919 (the
terminating gateway).
2 GW408 sends an ARQ with the called number having the 919 area code to a gatekeeper in its zone
(GK408).
3 GK408 resolves 919 to belong to a via-zone gatekeeper (GKVIA). GK408 then sends an LRQ to
GKVIA.
4 GKVIA receives the LRQ for the 919 number. GKVIA resolves the 919 prefix to belong to the Cisco
Unified Border Element. GKVIA is configured to route requests for 919 prefix calls through its Cisco
Unified Border Element. GKVIA sends an LCF to GK408.
5 GK408 returns an ACF specifying Cisco Unified Border Element to GW408.
6 GW408 sends a SETUP message to Cisco Unified Border Element for the 919 number.
7 Cisco Unified Border Element consults GKVIA with an ARQ message with the answerCall=true
parameter to admit the incoming call.
8 GKVIA responds with an ACF to admit the call. From the perspective of the gatekeeper, the first call
leg has been established.
9 Cisco Unified Border Element has a dial peer specifying that RAS messages should be sent to GKVIA
for all prefixes. Cisco Unified Border Element initiates the resending of the call by sending the ARQ
message with the answerCall parameter set to, false to GKVIA for 919.
10 GKVIA knows that prefix 919 belongs to GK919, and since the source zone is the via-zone, the
GKVIA sends an LRQ to GK919.
11 GK919 sees prefix 919 as a local zone and sends an LCF pointing to GW919.
12 GKVIA returns an ACF specifying GW919.
13 Cisco Unified Border Element sends a SETUP message to GW919 for the 919 call.
14 GW919 sends an ARQ to GK919 to request admission for the call.
15 GK919 sends an ACF with the answerCall=true parameter.
All other messages (for example, Proceeding, Alerting, and Connect) are created as two legs between
GW408, and GW919, with the Cisco Unified Border Element acting as an intermediate gateway.
Lawful Intercept Support
Lawful Intercept (LI) is the term used to describe the process by which law enforcement agencies conduct
electronic surveillance of circuit communications as authorized by judicial or administrative order. Cisco
Service Independent Intercept (SII) supports voice and data intercept and intercept requests are initiated by
MD using SNMPv3.
Basic SIP-to-SIP Set-up and Functionality Features
SIP-to-SIP Set-up
•
SIP-to-SIP Basic Functionality
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
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Cisco Unified Border Element Fundamentals and Basic Setup
Configuring Call Rate Monitoring
•
•
Transport Control Protocol (TCP) and User Datagram Protocol (UDP) interworking
Transport Control Protocol (TCP) and User Datagram Protocol (UDP)
http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_t3.html#wp1625679
•
Cisco Unified Border Element and Cisco Unified Communications Manager Express Support for
Universal Packaging
http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_m3.html#wp1396382
IP Addressing
•
•
•
SIP--Gateway Support for the bind Command
Configuring an Inbound Dial-peer to Match the URI on SIP Calls
Configuring Media Flow Through and Flow Around
Basic Dial Plan Management
•
Dial Peer Configuration on Voice Gateway Routers
http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/dial_peer/dpeer_c.html
Basic Protocol and DTMF Interworking
•
Supported Protocol Interworking
http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gwoverview_ps5640_TSD_Products_Configuration_Guide_Chapter.html#wp1168393
Configuring Call Rate Monitoring
Perform this task to enable voice call rate monitoring on Cisco UBE.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip
4. voice call rate monitor
5. end
6. show voice call rate
DETAILED STEPS
Command or Action
Step 1 enable
Purpose
Enables privileged EXEC mode.
•
Enter your password if prompted.
Example:
Router> enable
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
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Cisco Unified Border Element Fundamentals and Basic Setup
Command or Action
Purpose
Step 2 configure terminal
Enters global configuration mode.
Example:
Router# configure terminal
Step 3 voice service voip
Enters voice service configuration mode.
Example:
Router(config)# voice service voip
Step 4 voice call rate monitor
Enables voice call rate monitoring.
Example:
Router(conf-voi-serv)# voice call rate monitor
Step 5 end
Exits voice service configuration mode and enters privileged
EXEC mode.
Example:
Router(conf-voi-serv)# end
Step 6 show voice call rate
(Optional) Displays the voice call rate information.
Example:
Router# show voice call rate
Cisco and the Cisco logo are trademarks or registered trademarks of Cisco and/or its affiliates in the U.S.
and other countries. To view a list of Cisco trademarks, go to this URL: www.cisco.com/go/trademarks.
Third-party trademarks mentioned are the property of their respective owners. The use of the word partner
does not imply a partnership relationship between Cisco and any other company. (1110R)
Any Internet Protocol (IP) addresses and phone numbers used in this document are not intended to be
actual addresses and phone numbers. Any examples, command display output, network topology diagrams,
and other figures included in the document are shown for illustrative purposes only. Any use of actual IP
addresses or phone numbers in illustrative content is unintentional and coincidental.
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
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Lawful Intercept Support
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
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IP-to-IP Gateway SIP-to-SIP Basic
Functionality
SIP-to-SIP Basic Functionality for Cisco Unified Border Element (Cisco UBE) and Cisco Unified Border
Element (Enterprise) (Cisco UBE (Enterprise)) provides termination and reorigination of both signaling
and media between VoIP and video networks using SIP signaling in conformance with RFC3261. The
SIP-to-SIP protocol interworking capabilities support the following:
•
•
•
•
•
•
•
•
Basic voice calls (Supported audio codecs include: G.711, G.729, G.728, G.726, G.723, G.722,
gsmamr nb, AAC_LD, iLBC. Video codecs: H.263, and H.264)
Calling/called name and number
DTMF relay interworking
•
•
•
•
•
•
•
•
◦ SIP RFC 2833 <-> SIP RFC 2833
◦ SIP Notify <-> SIP Notify
Interworking between SIP early-media and SIP early-media signaling
Interworking between SIP delayed-media and SIP delayed-media signaling
RADIUS call-accounting records
RSVP synchronized with call signaling
SIP-to-SIP Video calls
TCL IVR 2.0 for SIP, including media playout and digit collection (RFC 2833 DTMF relay)
T.38 fax relay and Cisco fax relay
UDP and TCP transport
Finding Feature Information, page 9
Prerequisites for IP-to-IP Gateway SIP-to-SIP Basic Functionality, page 10
Restrictions for IP-to-IP Gateway SIP-to-SIP Basic Functionality, page 10
How to Configure SIP-to-SIP Connections in a Cisco Unified Border Element Enterprise, page 10
Feature Information for IP-to-IP Gateway SIP-to-SIP Basic Functionality, page 11
Finding Feature Information
Your software release may not support all the features documented in this module. For the latest feature
information and caveats, see the release notes for your platform and software release. To find information
about the features documented in this module, and to see a list of the releases in which each feature is
supported, see the Feature Information Table at the end of this document.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
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IP-to-IP Gateway SIP-to-SIP Basic Functionality
Prerequisites for IP-to-IP Gateway SIP-to-SIP Basic Functionality
Prerequisites for IP-to-IP Gateway SIP-to-SIP Basic
Functionality
Cisco Unified Border Element
•
Cisco IOS Release 12.2(13)T3 or a later release must be installed and running on your Cisco Unified
Border Element.
Cisco Unified Border Element (Enterprise)
•
Cisco IOS XE Release 2.5 or a later release must be installed and running on your Cisco ASR 1000
Series Router.
Restrictions for IP-to-IP Gateway SIP-to-SIP Basic
Functionality
•
Connections are disabled by default in Cisco IOS images that support the Cisco UBE (Enterprise).
How to Configure SIP-to-SIP Connections in a Cisco Unified
Border Element Enterprise
To configure SIP-to-SIP connection types, perform the steps in this section.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip
4. allow-connections from-type to to-type
5. exit
DETAILED STEPS
Command or Action
Step 1 enable
Purpose
Enables privileged EXEC mode.
•
Enter your password if prompted.
Example:
Router> enable
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
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IP-to-IP Gateway SIP-to-SIP Basic Functionality
Feature Information for IP-to-IP Gateway SIP-to-SIP Basic Functionality
Command or Action
Purpose
Step 2 configure terminal
Enters global configuration mode.
Example:
Router# configure terminal
Step 3 voice service voip
Enters VoIP voice-service configuration mode.
Example:
Router(config)# voice service voip
Step 4 allow-connections from-type to to-type
Allows connections between specific types of endpoints in an Cisco
UBE. Arguments are as follows:
•
•
Example:
Router(config-voi-serv)# allow-connections
sip to sip
from-type --Type of connection. Valid values: h323, sip.
to-type --Type of connection. Valid values: h323, sip.
Note H.323-to-H.323: By default, H.323-to-H.323 connections are
disabled and POTS-to-any and any-to-POTS connections are
enabled.
Step 5 exit
Exits the current mode.
Example:
Router(config-voi-serv)# exit
Feature Information for IP-to-IP Gateway SIP-to-SIP Basic
Functionality
The following table provides release information about the feature or features described in this module.
This table lists only the software release that introduced support for a given feature in a given software
release train. Unless noted otherwise, subsequent releases of that software release train also support that
feature.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.
Cisco Unified Border Element Feature History Information.
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
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IP-to-IP Gateway SIP-to-SIP Basic Functionality
Table 1
Feature Information for SIP-to-SIP Connections in a Cisco Unified Border Element
Feature Name
Releases
Feature Information
SIP-to-SIP Basic Functionality
12.2(13)T3 12.3(7)T
This feature provides termination
and reorigination of both
signaling and media between
VoIP and video networks using
SIP signaling in conformance
with RFC3261.
The following commands were
introduced or modified: allowconnections
Cisco Unified Border Element (Enterprise) Feature History Information.
Table 2
Feature Information for SIP-to-SIP Connections in a Cisco Unified Border Element (Enterprise)
Feature Name
Releases
Feature Information
SIP-to-SIP Basic Functionality
Cisco IOS XE Release 2.5
This feature provides termination
and reorigination of both
signaling and media between
VoIP and video networks using
SIP signaling in conformance
with RFC3261.
The following commands were
introduced or modified: allowconnections
Cisco and the Cisco logo are trademarks or registered trademarks of Cisco and/or its affiliates in the U.S.
and other countries. To view a list of Cisco trademarks, go to this URL: www.cisco.com/go/trademarks.
Third-party trademarks mentioned are the property of their respective owners. The use of the word partner
does not imply a partnership relationship between Cisco and any other company. (1110R)
Any Internet Protocol (IP) addresses and phone numbers used in this document are not intended to be
actual addresses and phone numbers. Any examples, command display output, network topology diagrams,
and other figures included in the document are shown for illustrative purposes only. Any use of actual IP
addresses or phone numbers in illustrative content is unintentional and coincidental.
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
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SIP-to-SIP Extended Feature Functionality for
Session Border Controllers
The SIP-to-SIP Extended Feature Functionality for Session Border Controllers (SBCs) enables the SIP-toSIP functionality to conform with RFC 3261 to interoperate with SIP User Agents (UAs). The SIP-to-SIP
Extended Feature Functionality includes:
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
Call Admission Control (based on CPU, memory, and total calls)
Delayed Media Call
ENUM support
Configuring SIP Error Message Pass Through
Interoperability with Cisco Unified Communications Manager 5.0 and BroadSoft
Lawful Intercept
Media Inactivity
Modem Passthrough over VoIP, page 14
TCP and UDP interworking
Tcl scripts with SIP NOTIFY VoiceXML with SIP-to-SIP
Transport Layer Security (TLS)
Finding Feature Information, page 13
Prerequisites for SIP-to-SIP Extended Feature Functionality for Session Border Controllers, page
14
Modem Passthrough over VoIP, page 14
Feature Information for SIP-to-SIP Extended Feature Functionality for Session Border Controllers,
page 23
Finding Feature Information
Your software release may not support all the features documented in this module. For the latest feature
information and caveats, see the release notes for your platform and software release. To find information
about the features documented in this module, and to see a list of the releases in which each feature is
supported, see the Feature Information Table at the end of this document.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
13
SIP-to-SIP Extended Feature Functionality for Session Border Controllers
Prerequisites for SIP-to-SIP Extended Feature Functionality for Session Border Controllers
Prerequisites for SIP-to-SIP Extended Feature Functionality
for Session Border Controllers
Cisco Unified Border Element
•
Cisco IOS Release 12.4(6)T or a later release must be installed and running on your Cisco Unified
Border Element.
Cisco Unified Border Element (Enterprise)
•
Cisco IOS XE Release 3.1S or a later release must be installed and running on your Cisco ASR 1000
Series Router.
Modem Passthrough over VoIP
The Modem Passthrough over VoIP feature provides the transport of modem signals through a packet
network by using pulse code modulation (PCM) encoded packets.
The Modem Passthrough over VoIP feature performs the following functions:
•
•
•
•
•
Represses processing functions like compression, echo cancellation, high-pass filter, and voice activity
detection (VAD).
Issues redundant packets to protect against random packet drops.
Provides static jitter buffers of 200 milliseconds to protect against clock skew.
Discriminates modem signals from voice and fax signals, indicating the detection of the modem signal
across the connection, and placing the connection in a state that transports the signal across the
network with the least amount of distortion.
Reliably maintains a modem connection across the packet network for a long duration under normal
network conditions.
For further details, the functions of the Modem Passthrough over VoIP feature are described in the
following sections.
Modem Tone Detection
The gateway is able to detect modems at speeds up to V.90.
Passthrough Switchover
When the gateway detects a data modem, both the originating gateway and the terminating gateway roll
over to G.711. The roll over to G.711 disables the high-pass filter, disables echo cancellation, and disables
VAD. At the end of the modem call, the voice ports revert to the prior configuration and the digital signal
processor (DSP) goes back to the state before switchover. You can configure the codec by selecting the
g711alaw or g711ulaw option of the codec command.
See also the How to Configure Modem Passthrough over VoIP, page 16 section in this document.
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14
Prerequisites for the Modem Passthrough over VoIP Feature
Modem Passthrough over VoIP
Controlled Redundancy
You can enable payload redundancy so that the Modem Passthrough over VoIP switchover causes the
gateway to emit redundant packets.
Packet Size
When redundancy is enabled, 10-ms sample-sized packets are sent. When redundancy is disabled, 20-ms
sample-sized packets are sent.
Clock Slip Buffer Management
When the gateway detects a data modem, both the originating gateway and the terminating gateway switch
from dynamic jitter buffers to static jitter buffers of 200-ms depth. The switch from dynamic to static is to
compensate for Public Switched Telephone Network (PSTN) clocking differences at the originating
gateway and the terminating gateway. At the conclusion of the modem call, the voice ports revert to
dynamic jitter buffers.
The figure below illustrates the connection from the client modem to a MICA technologies modem network
access server (NAS).
Figure 2
•
•
•
•
•
•
•
•
•
Modem Passthrough Connection
Prerequisites for the Modem Passthrough over VoIP Feature, page 15
Restrictions for the Modem Passthrough over VoIP Feature, page 16
How to Configure Modem Passthrough over VoIP, page 16
Configuring Modem Passthrough over VoIP Globally, page 17
Configuring Modem Passthrough over VoIP for a Specific Dial Peer, page 18
Verifying Modem Passthrough over VoIP, page 20
Troubleshooting Tips, page 21
Monitoring and Maintaining Modem Passthrough over VoIP, page 21
Configuration Examples, page 21
Prerequisites for the Modem Passthrough over VoIP Feature
•
VoIP enabled network.
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Restrictions for the Modem Passthrough over VoIP Feature
Modem Passthrough over VoIP
•
•
Cisco IOS Release 12.1(3)T must run on the gateways for the Modem Passthrough over VoIP feature
to work.
Network suitability to pass modem traffic. The key attributes are packet loss, delay, and jitter. These
characteristics of the network can be determined by using the Cisco IOS feature Service Assurance
Agent.
Cisco Unified Border Element
•
Cisco IOS Release 12.4(6)T or a later release must be installed and running on your Cisco Unified
Border Element.
Cisco Unified Border Element (Enterprise)
•
Cisco IOS XE Release 3.3S or a later release must be installed and running on your Cisco ASR 1000
Series Router.
Restrictions for the Modem Passthrough over VoIP Feature
Cisco Unified Border Element (Enterprise)
•
Note
If call started as g729, upon modem tone (2100Hz) detection both the outgoing gateway (OGW) and
the trunking gateway (TGW) will genearate NSE packets towards peer side and up speed to g711 as
Cisco UBE(Enterprise) passes these packets to the peer side.
That OGW and TGW display the new codec, but the Cisco UBE (Enterprise) continues to show the original
codec g729 in the show commands.
How to Configure Modem Passthrough over VoIP
By default, modem passthrough over VoIP capability and redundancy are disabled.
Tip
You need to configure modem passthrough in both the originating gateway and the terminating gateway for
the Modem Passthrough over VoIP feature to operate. If you configure only one of the gateways in a pair,
the modem call will not connect successfully.
Redundancy can be enabled in one or both of the gateways. When only a single gateway is configured for
redundancy, the other gateway receives the packets correctly, but does not produce redundant packets.
See the following sections for the Modem Passthrough over VoIP feature. The two configuration tasks can
configure separately or together. If both are configured, the dial-peer configuration takes precedence over
the global configuration. Consequently, a call matching a particular dial-peer will first try to apply the
modem passthrough configuration on the dial-peer. Then, if a specific dial-peer is not configured, the router
will use the global configuration:
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
16
Configuring Modem Passthrough over VoIP Globally
Modem Passthrough over VoIP
Configuring Modem Passthrough over VoIP Globally
For the Modem Passthrough over VoIP feature to operate, you need to configure modem passthrough in
both the originating gateway and the terminating gateway so that the modem call matches a voip dial-peer
on the gateway.
When using the voice service voip and modem passthrough nse commands on a terminating gateway to
globally set up fax or modem passthrough with NSEs, you must also ensure that each incoming call will be
associated with a VoIP dial peer to retrieve the global fax or modem configuration. You associate calls with
dial peers by using the incoming called-number command to specify a sequence of digits that incoming
calls can match.
To configure the Modem Passthrough over VoIP feature for all the connections of a gateway, use the
following commands beginning in global configuration mode:
SUMMARY STEPS
1. enable
2. voice service voip
3. modem passthrough nse [payload-type number] codec {g711ulaw | g711alaw} [redundancy]
[maximum-sessions value]
4. exit
5. exit
DETAILED STEPS
Purpose
Command or Action
Step 1 enable
Enables privileged EXEC mode.
•
Enter your password if prompted.
Example:
Router> enable
Step 2 voice service voip
Enters voice-service configuration mode.
Configures voice service for all the connections for the gateways.
Example:
Router(config)# voice service voip
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Configuring Modem Passthrough over VoIP for a Specific Dial Peer
Modem Passthrough over VoIP
Command or Action
Step 3 modem passthrough nse [payload-type
number] codec {g711ulaw | g711alaw}
[redundancy] [maximum-sessions value]
Example:
Router(config)# Router(conf-voiserv)# modem passthrough nse payloadtype 97 codec g711alaw redundancy
maximum-sessions 3
Purpose
Configures the Modem Passthrough over VoIP feature The default
behavior is no modem passthrough.
The payload type is an optional parameter for the nse keyword. Use the
same payload-type number for both the originating gateway and the
terminating gateway. The payload-type numbercan be set from 96 to 119.
If you do not specify the payload-type number, the numberdefaults to
100. When the payload-type is 100, and you use the show runningconfigcommand, the payload-type parameter does not appear.
Use the same codec type for both the originating gateway and the
terminating gateway. g711ulaw codec is required for T1, and g711alaw
codec is required for E1.
The redundancy keyword is an optional parameter for sending redundant
packets for modem traffic.
The maximum-sessions keyword is an optional parameter for the
redundancykeyword. This parameter determines the maximum
simultaneous modem passthrough sessions with redundancy.
Step 4 exit
Exits voice-service configuration mode.
Example:
Router(conf-voi-serv)# exit
Step 5 exit
Exits global configuration mode.
Example:
Router(config)# exit
Configuring Modem Passthrough over VoIP for a Specific Dial Peer
You can configure the Modem Passthrough over VoIP feature on a specific dial peer in two ways, as
follows:
•
•
Globally in the voice-service configuration mode
Individually in the dial-peer configuration mode on a specific dial peer
The default behavior for the voice-service configuration mode is no modem passthrough. This default
behavior implies that modem passthrough is disabled for all dial peers on the gateway by default.
To enable Modem Passthrough on the VoIP dial peers on both the originating and terminating gateway,
configure modem passthrough globally or explicitly on the dial peer.
For modem passthrough to operate, you must define VoIP dial peers on both gateways to match the call, for
example, by using a destination pattern or an incoming called number. The modem passthrough parameters
associated with those dial peers then will apply to the call.
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SIP-to-SIP Extended Feature Functionality for Session Border Controllers
Modem Passthrough over VoIP
Note
When modem passthrough is configured individually for a specific dial peer, that configuration for the
specific dial peer takes precedence over the global configuration.
To configure the Modem Passthrough over VoIP feature for a specific dial peer, use the following
commands beginning in global configuration mode:
SUMMARY STEPS
1. enable
2. dial-peer voice number voip
3. modem passthrough {system | nse [payload-type number] codec {g711ulaw | g711alaw}
[redundancy]}
4. exit
5. exit
DETAILED STEPS
Command or Action
Purpose
Step 1 enable
Enables privileged EXEC mode.
•
Enter your password if prompted.
Example:
Router> enable
Step 2 dial-peer voice number voip
Enters dial-peer configuration mode.
Configures a specific dial peer in dial-peer configuration mode.
Example:
Router(config)# dial-peer voice 5
voip
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Verifying Modem Passthrough over VoIP
Modem Passthrough over VoIP
Command or Action
Step 3 modem passthrough {system | nse
[payload-type number] codec {g711ulaw |
g711alaw}[redundancy]}
Example:
Router(config-dial-peer)# modem
passthrough nse payload-type 97
codec g711alaw redundancy
Purpose
Configures the Modem Passthrough over VoIP feature for a specific dial
peer. The default behavior for the Modem Passthrough for VoIP feature in
dial-peer configuration mode is modem passthrough system. As required,
the gateway defaults to no modem passthrough.
When the system keyword is enabled, the following parameters are not
available: nse, payload-type, codec, and redundancy. Instead the values
from the global configuration are used.
The payload type is an optional parameter for the nse keyword. Use the
same payload-type number for both the originating gateway and the
terminating gateway. The payload-type numbercan be set from 96 to 119.
If you do not specify the payload-type number, the numberdefaults to 100.
When the payload-type is 100, and you use the show runningconfigcommand, the payload-type parameter does not appear.
Use the same codec type for both the originating gateway and the
terminating gateway. g711ulaw codec is required for T1, and g711alaw
codec is required for E1.
The redundancy keyword is an optional parameter for sending redundant
packets for modem traffic.
Step 4 exit
Exits dial-peer configuration mode and returns to the global configuration
mode.
Example:
Router(config-dial-peer)# exit
Step 5 exit
Exits global configuration mode.
Example:
Router(config)# exit
Verifying Modem Passthrough over VoIP
To verify that the Modem Passthrough over VoIP feature is enabled, perform the following steps:
SUMMARY STEPS
1. Enter the show runcommand to verify the configuration.
2. Enter the show dial-peer voice command to verify that Modem Passthrough over VoIP is enabled.
DETAILED STEPS
Step 1
Enter the show runcommand to verify the configuration.
Step 2
Enter the show dial-peer voice command to verify that Modem Passthrough over VoIP is enabled.
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20
Troubleshooting Tips
Modem Passthrough over VoIP
Troubleshooting Tips
To troubleshoot the Modem Passthrough over VoIP feature, perform the following steps:
•
•
•
•
•
Make sure that you can make a voice call.
Make sure that Modem Passthrough over VoIP is configured on both the originating gateway and the
terminating gateway.
Make sure that both the originating gateway and the terminating gateway have the same named
signaling event (NSE) payload-type number.
Make sure that both the originating gateway and the terminating gateway have the same maximumsessions value when the two gateways are configured in the voice-service configuration mode.
Use the debug vtsp dsp and debug vtsp session commands to debug a problem.
Monitoring and Maintaining Modem Passthrough over VoIP
To monitor and maintain the Modem Passthrough over VoIP feature, use the following commands in
privileged EXEC mode:
Command
Router#
show call active {voice
Purpose
|
fax}[brief]
show dial-peer voice [number
summary]
Router#
|
Displays information for the active call table or
displays the voice call history table. The brief
option displays a truncated version of either option.
Displays configuration information for dial peers.
The number argument specifies a specific dial peer
from 1 to 32767. The summary option displays a
summary of all dial peers.
Configuration Examples
The following is sample configuration for the Modem Passthrough over VoIP feature:
version 12.1
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
voice service voip
modem passthrough nse codec g711ulaw redundancy maximum-session 5
!
!
resource-pool disable
!
!
!
!
!
ip subnet-zero
ip ftp source-interface Ethernet0
ip ftp username lab
ip ftp password lab
no ip domain-lookup
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SIP-to-SIP Extended Feature Functionality for Session Border Controllers
Modem Passthrough over VoIP
!
isdn switch-type primary-5ess
cns event-service server
!
!
!
!
!
mta receive maximum-recipients 0
!
!
controller T1 0
framing esf
clock source line primary
linecode b8zs
pri-group timeslots 1-24
!
controller T1 1
shutdown
clock source line secondary 1
!
controller T1 2
shutdown
!
controller T1 3
shutdown
!
!
!
interface Ethernet0
ip address 1.1.2.2 255.0.0.0
no ip route-cache
no ip mroute-cache
!
interface Serial0:23
no ip address
encapsulation ppp
ip mroute-cache
no logging event link-status
isdn switch-type primary-5ess
isdn incoming-voice modem
no peer default ip address
no fair-queue
no cdp enable
no ppp lcp fast-start
!
interface FastEthernet0
ip address 26.0.0.1 255.0.0.0
no ip route-cache
no ip mroute-cache
load-interval 30
duplex full
speed auto
no cdp enable
!
ip classless
ip route 17.18.0.0 255.255.0.0 1.1.1.1
no ip http server
!
!
!
!
voice-port 0:D
!
dial-peer voice 1 pots
incoming called-number 55511..
destination-pattern 020..
direct-inward-dial
port 0:D
prefix 020
!
dial-peer voice 2 voip
incoming called-number 020..
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SIP-to-SIP Extended Feature Functionality for Session Border Controllers
Feature Information for SIP-to-SIP Extended Feature Functionality for Session Border Controllers
destination-pattern 55511..
modem passthrough nse codec g711ulaw redundancy
session target ipv4:26.0.0.2
!
!
line con 0
exec-timeout 0 0
transport input none
line aux 0
line vty 0 4
login
!
!
end
Feature Information for SIP-to-SIP Extended Feature
Functionality for Session Border Controllers
The following table provides release information about the feature or features described in this module.
This table lists only the software release that introduced support for a given feature in a given software
release train. Unless noted otherwise, subsequent releases of that software release train also support that
feature.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.
Table 3
Feature Information for Configuring SIP-to-SIP Extended Feature Functionality for Session Border
Controllers for the Cisco Unified Border Element.
Feature Name
Releases
Feature Information
SIP-to-SIP Extended Feature
Functionality for Session Border
Controllers
12.4(6)T
The SIP-to-SIP Extended Feature
Functionality for Session Border
Controllers (SBCs) enables the
SIP-to-SIP functionality to
conform with RFC 3261 to
interoperate with SIP User
Agents (UAs).
The following commands were
introduced or modified: modem
passthrough (dial-peer);
modem passthrough (voiceservice); show call active voice
voice; show call history voice
voice; show dial-peer voice;
voice service.
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SIP-to-SIP Extended Feature Functionality for Session Border Controllers
Table 4
Feature Information for Configuring SIP-to-SIP Extended Feature Functionality for Session Border
Controllers for the Cisco Unified Border Element (Enterprise).
Feature Name
Releases
Feature Information
SIP-to-SIP Extended Feature
Functionality for Session Border
Controllers
Cisco IOS XE Release 3.1S,
The SIP-to-SIP Extended Feature
Functionality for Session Border
Controllers (SBCs) enables the
SIP-to-SIP functionality to
conform with RFC 3261 to
interoperate with SIP User
Agents (UAs).
Cisco IOS XE Release 3.3S
The following commands were
introduced or modified: modem
passthrough (dial-peer);
modem passthrough (voiceservice); show call active voice
voice; show call history voice
voice; show dial-peer voice;
voice service.
Cisco and the Cisco logo are trademarks or registered trademarks of Cisco and/or its affiliates in the U.S.
and other countries. To view a list of Cisco trademarks, go to this URL: www.cisco.com/go/trademarks.
Third-party trademarks mentioned are the property of their respective owners. The use of the word partner
does not imply a partnership relationship between Cisco and any other company. (1110R)
Any Internet Protocol (IP) addresses and phone numbers used in this document are not intended to be
actual addresses and phone numbers. Any examples, command display output, network topology diagrams,
and other figures included in the document are shown for illustrative purposes only. Any use of actual IP
addresses or phone numbers in illustrative content is unintentional and coincidental.
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
24
SIP Gateway Support for the bind Command
The Gateway Support for the bind Command feature introduces the bind command, which allows you to
configure the source IP address of signaling packets or both signaling and media packets. Befor this
feature was introduced the source address of a packet going out of a Cisco IOS gateway is not
deterministic. The session protocols and VoIP layers depended on the IP layer to give the best local
address and then used the address for the source address in signaling or media or both, even if multiple
interfaces can support a route to the destination address.
•
•
•
•
•
•
•
Finding Feature Information, page 25
Prerequisites for SIP Gateway Support for the bind Command, page 25
Information About SIP Gateway Support for the bind Command, page 26
How to Configure SIP Gateway Support for the bind Command, page 27
Verifying and Troubleshooting Tips, page 31
Configuration Examples for SIP Gateway Support for the bind Command, page 32
Feature Information for SIP Gateway Support for the bind Command, page 32
Finding Feature Information
Your software release may not support all the features documented in this module. For the latest feature
information and caveats, see the release notes for your platform and software release. To find information
about the features documented in this module, and to see a list of the releases in which each feature is
supported, see the Feature Information Table at the end of this document.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.
Prerequisites for SIP Gateway Support for the bind Command
Cisco Unified Border Element
•
Cisco IOS Release 12.2(8)T or a later release must be installed and running on your Cisco Unified
Border Element.
Cisco Unified Border Element (Enterprise)
•
Cisco IOS XE Release 2.5 or a later release must be installed and running on your Cisco ASR 1000
Series Router.
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25
SIP Gateway Support for the bind Command
Information About SIP Gateway Support for the bind Command
Information About SIP Gateway Support for the bind
Command
Prior to the Gateway Support for the bind Command feature the source address of a packet going out of the
gateway was never deterministic. That is, the session protocols and VoIP layers always depended on the IP
layer to give the best local address . The best local address was then used as the source address (the address
showing where the SIP request came from) for signaling and media packets. Using this nondeterministic
address occasionally caused confusion for firewall applications, because a firewall could not be configured
with an exact address and would take action on several different source address packets.
The bind interface command allows you to configure a specific interface’s IP address as the source IP
address of signaling and media packets. The address that goes out on the packet is bound to the IP address
of the interface specified with the bind command. Packets that are not destined to the bound address are
discarded.
When you do not specify a bind address, or if the interface is down, the IP layer still provides the best local
address.
With the bind command, SIP signaling and media paths can advertise the same source IP address on the
gateway for certain applications, even if the paths use different addresses to reach the source. This
eliminates confusion for firewall applications that, Without the binding, may have taken action on several
different source address packets.
The table below lists the results of the bind command based on the state of the interface.
Table 5
Command functions for the bind command based on the state of the interface
Interface State
Result Using Bind Command
A bind interface is shut down, or its IP Address is changed,
or the physical cable is pulled while SIP calls are active
The call becomes a one-way call with media flowing in only
one direction. It flows from the gateway where the change or
shutdown took place to the gateway where no change occurred.
Thus, the gateway with the status change no longer receives
media.
The call is then disconnected, but the disconnected message is
not understood by the gateway with the status change, and the
call is still assumed to be active.
No Shutdown-- With no active calls.
The TCP and UDP socket listeners are initially closed. (Socket
listeners receive datagrams addressed to the socket.)
Then the sockets are opened and bound to the IP address set by
the bind command.
The sockets accept packets destined for the bound address only.
No Shutdown-- With active calls.
The TCP and UDP socket listeners are initially closed.
Then the sockets are opened to listen to any IP address.
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Setting the Bind Address
How to Configure SIP Gateway Support for the bind Command
Interface State
Result Using Bind Command
Shutdown-- With or without active calls.
The TCP and User Datagram Protocol (UDP) socket listeners
are initially closed. (Socket listeners receive datagrams
addressed to the socket.)
Then the sockets are opened to listen to any IP address.
If the outgoing gateway has the bind command enabled and
has an active call, the call becomes a one-way call with media
flowing from the outgoing gateway to the terminating gateway.
The Bound interface’s IP address is removed
The TCP and UDP socket listeners are initially closed.
Then the sockets are opened to listen to any address, because
the IP address has been removed.
A message stating that the IP address has been deleted from
SIP bound interface is displayed.
If the outgoing gateway has the bindcommand enabled and has
an active call, the call becomes a one-way call with media
flowing from the outgoing gateway to the terminating gateway.
The physical cable is pulled on the bound port, or the
Interface layer goes down
The TCP and UDP socket listeners are initially closed.
Then the sockets are opened and bound to listen to any address.
When the pulled cable is replaced, the result is as documented
for no shutdowninterfaces.
Note
If there are active calls, the bind command will not take effect if it is issued for the first time or if it is
issued while another bind command is in effect. A message is displayed reminding you that there are active
calls and that the bind command change cannot take effect.
How to Configure SIP Gateway Support for the bind Command
•
•
Setting the Bind Address, page 27
Setting a Source IP Address for Signaling and Media Packets, page 28
Setting the Bind Address
To set the bind address, complete the task in this section.
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Setting a Source IP Address for Signaling and Media Packets
How to Configure SIP Gateway Support for the bind Command
SUMMARY STEPS
1. enable
2. configure terminal
3. dial-peer voice number voip
4. session target ipv4: destination-address
5. exit
DETAILED STEPS
Command or Action
Step 1 enable
Purpose
Enables privileged EXEC mode.
•
Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal
Enters global configuration mode.
Example:
Router# configure terminal
Step 3 dial-peer voice number voip
Enters dial peer configuration mode to configure a VoIP dial-peer.
Example:
Router(config)# dial-peer voice 2 voip
Step 4 session target ipv4: destination-address
Specifies a network-specific address for a dial peer.
•
Example:
•
Router(config-dial-peer)# session target
ipv4: 172.16.43.3
Step 5 exit
This command must be set to the bind address of the receiving
gateway before using the bind command.
ipv4 :destination-address: Sets the IP address of the dial peer. A
valid IP address is in this format: xxx.xxx.xxx.xxx.
Exits dial peer voice configuration mode.
Example:
Router(config-dial-peer)# exit
Setting a Source IP Address for Signaling and Media Packets
SIP configuration mode starts from voice-service VoIP configuration mode. When the router is in SIP
configuration mode, several options are available, including the bind command. To enable this feature,
review the prerequisites to make sure your network is compliant, and then complete the task in this section.
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SIP Gateway Support for the bind Command
How to Configure SIP Gateway Support for the bind Command
•
•
•
•
•
Endure you have Cisco IOS XE Release 2.5 or a later release installed and running on your Cisco ASR
1000 Series Router.
Ensure that the gateway has voice functionality that is configurable for SIP.
Establish a working IP network.
Configure VoIP.
Set the bind address prior to using the bind command.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip
4. sip
5. session transport {udp | tcp}
6. bind {control | all} source-interface interface-id
7. default {bind|rel1xx|session-transport|url}
8. exit
DETAILED STEPS
Command or Action
Purpose
Step 1 enable
Enables privileged EXEC mode.
•
Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal
Enters global configuration mode.
Example:
Router# configure terminal
Step 3 voice service voip
Enters voice-service configuration mode
Example:
Router(config)# voice service voip
Step 4 sip
Enters the SIP configuration mode.
Example:
Router(config-voi-srv)# sip
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SIP Gateway Support for the bind Command
How to Configure SIP Gateway Support for the bind Command
Command or Action
Step 5 session transport {udp | tcp}
Example:
Purpose
(Optional) Sets the session transport type for the SIP user agent.
•
•
Router(conf-serv-sip)# session
transport udp
Step 6 bind {control | all} source-interface
interface-id
The default is UDP.
The transport protocol (udp or tcp) specified with the session
transportcommand, and the protocol specified with the transport
command, must be identical.
Sets a source address for signaling and media packets.
Example:
•
•
•
Router(conf-serv-sip)# bind all
source- interface fastethernet
•
control : Binds SIP signaling packets.
all : Binds SIP signaling packets and media packets.
source-interface : Specifies an interface as the source address of SIP
packets.
interface-id argument specifies the type of interface:
◦
◦
◦
◦
◦
◦
◦
◦
◦
◦
◦
◦
◦
◦
◦
Step 7 default {bind|rel1xx|session-transport|
url}
Async
BVI
CTunnel
Dialer
Ethernet
FastEthernet
Lex
Loopback
Multilink
Null
Serial
Tunnel
Vif
Virtual-Template
Virtual-TokenRing
(Optional) Resets the default value of a SIP command.
•
Example:
•
Router(conf-serv-sip)# bind
•
•
bind-- Configures the source address of signaling and media packets to a
specific interface’s IP address
rel1xx --Enables all SIP provisional responses (other than 100 Trying) to
be sent reliably to the remote SIP endpoint
session-transport --Configures the underlying transport layer protocol
for SIP messages to TCP or UDP
url --Configures URLs to either the SIP or TEL format for your voip sip
calls
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Verifying a Bound IP Address
Verifying and Troubleshooting Tips
Command or Action
Purpose
Step 8 exit
Exits the current configuration mode.
Example:
Router(conf-serv-sip)# exit
Verifying and Troubleshooting Tips
Two show commands verify the correct settings for the bind command. The first enables you to verify a
bound IP address. The second indicates the status of bind (enabled or disabled):
•
•
•
•
Verifying a Bound IP Address, page 31
Verifying Bind Status, page 31
Verifying a Bound IP Address, page 31
Verifying Bind Status, page 31
Verifying a Bound IP Address
The following examples show output for the show ip socketcommand, indicating that the bind address of
the receiving gateway is set:
Router# show ip socket
Proto Remote Port Local Port In Out Stat TTY OutputIF
17 0.0.0.0 0 --any-- 2517 0 0 9 0
17 --listen-- 172.18.192.204 1698 0 0 1 0
17 0.0.0.0 0 172.18.192.204 67 0 0 489 0
17 0.0.0.0 0 172.18.192.204 5060 0 0 A1 0
Verifying Bind Status
The following example shows output for the show sip-ua statuscommand, indicating that bind is enabled.
Router# show sip-ua status
SIP User Agent Status
SIP User Agent for UDP : ENABLED
SIP User Agent for TCP : ENABLED
SIP User Agent bind status(signaling): ENABLED 172.18.192.204
SIP User Agent bind status(media): ENABLED 172.18.192.204
SIP max-forwards : 6
SIP DNS SRV version: 1 (rfc 2052)
To troubleshoot this feature, perform the following:
•
Use the debug ccsip all command to enable all SIP debugging capabilities, or use one of the following
SIP debug commands:
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SIP Gateway Support for the bind Command Example
Configuration Examples for SIP Gateway Support for the bind Command
•
•
◦ debug ccsip calls
◦ debug ccsip error
◦ debug ccsip events
◦ debug ccsip messages
◦ debug ccsip states
Use the show ip socketcommand to display IP socket information.
Use the show sip-ua statuscommand to verify if binding is enabled. See the show sip-ua
statuscommand for details.
Configuration Examples for SIP Gateway Support for the bind
Command
•
SIP Gateway Support for the bind Command Example, page 32
SIP Gateway Support for the bind Command Example
This section shows partial output from the show running-config command, indicating that bind is
functional on receiving router 172.18.192.204.
ip subnet-zero
ip ftp source-interface Ethernet0
!
voice service voip
sip
bind all source-interface FastEthernet0
!
interface FastEthernet0
ip address 172.18.192.204 255.255.255.0
duplex auto
speed auto
fair-queue 64 256 1000
ip rsvp bandwidth 75000 100
!!
Feature Information for SIP Gateway Support for the bind
Command
The following table provides release information about the feature or features described in this module.
This table lists only the software release that introduced support for a given feature in a given software
release train. Unless noted otherwise, subsequent releases of that software release train also support that
feature.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.
Feature History Table entry for the Cisco Unified Border Element.
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
32
SIP Gateway Support for the bind Command
Table 6
Feature Information for SIP: Gateway Support for the bind Command
Feature Name
Releases
Feature Information
SIP: Gateway Support for the
bind Command
12.2(8)T, 12.3(2)T, 12.2(11)T,
12.2(15)T
In Cisco IOS XE Release 2.5,
This feature was introduced on
the Cisco ASR 1000 Series
Routers.
The following commands were
introduced or modified: bindand
sip.
Feature History Table entry for the Cisco Unified Border Element (Enterprise) .
Table 7
Feature Information for SIP: Gateway Support for the bind Command
Feature Name
Releases
Feature Information
SIP: Gateway Support for the
bind Command
Cisco IOS XE Release 2.5
In Cisco IOS XE Release 2.5,
This feature was introduced on
the Cisco ASR 1000 Series
Routers.
The following commands were
introduced or modified: bindand
sip.
Cisco and the Cisco logo are trademarks or registered trademarks of Cisco and/or its affiliates in the U.S.
and other countries. To view a list of Cisco trademarks, go to this URL: www.cisco.com/go/trademarks.
Third-party trademarks mentioned are the property of their respective owners. The use of the word partner
does not imply a partnership relationship between Cisco and any other company. (1110R)
Any Internet Protocol (IP) addresses and phone numbers used in this document are not intended to be
actual addresses and phone numbers. Any examples, command display output, network topology diagrams,
and other figures included in the document are shown for illustrative purposes only. Any use of actual IP
addresses or phone numbers in illustrative content is unintentional and coincidental.
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
33
SIP Gateway Support for the bind Command Example
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
34
SIP Video Calls with Flow Around Media
The SIP Video Calls with Flow Around Media feature provides the ability to have a SIP video call where
the media flows around the Cisco Unified Border Element (Cisco UBE) and the Cisco Unified Border
Element (Enterprise) platform. Previous support was only for call scenarios where the media flowed
through the Cisco UBE.
•
•
•
•
•
Finding Feature Information, page 35
Prerequisites for SIP Video Calls with Flow Around Media, page 35
Restrictions for SIP Video Calls with Flow Around Media, page 35
How to Configure Support for SIP Video Calls with Flow Around Media, page 36
Feature Information for Support for SIP Video Calls with Flow Around Media, page 36
Finding Feature Information
Your software release may not support all the features documented in this module. For the latest feature
information and caveats, see the release notes for your platform and software release. To find information
about the features documented in this module, and to see a list of the releases in which each feature is
supported, see the Feature Information Table at the end of this document.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.
Prerequisites for SIP Video Calls with Flow Around Media
Cisco Unified Border Element
•
Cisco IOS Release 12.4(15)XZ or a later release must be installed and running on your Cisco Unified
Border Element.
Cisco Unified Border Element (Enterprise)
•
Cisco IOS XE Release 3.1S or a later release must be installed and running on your Cisco ASR 1000
Series Router.
Restrictions for SIP Video Calls with Flow Around Media
•
Media flow-around for Delayed-Offer to Early-Offer audio and video calls is not supported.
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35
SIP Video Calls with Flow Around Media
How to Configure Support for SIP Video Calls with Flow Around Media
How to Configure Support for SIP Video Calls with Flow
Around Media
To enable this feature use the mediacommand in dial peer, voice class, or voice service configuration
mode. For detailed information on the use of this command, see the Cisco IOS Voice Command Reference
at the following URL: http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_book.html
Feature Information for Support for SIP Video Calls with Flow
Around Media
The following table provides release information about the feature or features described in this module.
This table lists only the software release that introduced support for a given feature in a given software
release train. Unless noted otherwise, subsequent releases of that software release train also support that
feature.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.
Feature History Table entry for the Cisco Unified Border Element.
Table 8
Feature Information for SIP Video Calls with Flow Around Media
Feature Name
Releases
Feature Information
SIP Video Calls with Flow
Around Media
12.4(15)XZ 12.4(20)T
This feature provides the
capability for media packets to
pass directly between endpoints
without the intervention of the
Cisco UBE.
The following command was
modified by this feature: media
Feature History Table entry for the Cisco Unified Border Element (Enterprise).
Table 9
Feature Information for SIP Video Calls with Flow Around Media
Feature Name
Releases
Feature Information
SIP Video Calls with Flow
Around Media
Cisco IOS XE Release 3.1S
This feature provides the
capability for media packets to
pass directly between endpoints
without the intervention of the
Cisco UBE.
The following command was
modified by this feature: media
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
36
SIP Video Calls with Flow Around Media
Cisco and the Cisco logo are trademarks or registered trademarks of Cisco and/or its affiliates in the U.S.
and other countries. To view a list of Cisco trademarks, go to this URL: www.cisco.com/go/trademarks.
Third-party trademarks mentioned are the property of their respective owners. The use of the word partner
does not imply a partnership relationship between Cisco and any other company. (1110R)
Any Internet Protocol (IP) addresses and phone numbers used in this document are not intended to be
actual addresses and phone numbers. Any examples, command display output, network topology diagrams,
and other figures included in the document are shown for illustrative purposes only. Any use of actual IP
addresses or phone numbers in illustrative content is unintentional and coincidental.
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
37
SIP Video Calls with Flow Around Media
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
38
Additional References
The following sections provide references related to the Cisco Unified Border Element (Enterprise)
Configuration Guide.
•
•
•
•
•
Related Documents, page 39
Standards, page 40
MIBs, page 40
RFCs, page 41
Technical Assistance, page 42
Related Documents
Related Topic
Document Title
Cisco IOS commands
Cisco IOS Master Commands List, All Releases
Cisco IOS Voice commands
Cisco IOS Voice Command Reference
Cisco IOS Voice Configuration Library
For more information about Cisco IOS voice
features, including feature documents, and
troubleshooting information--at
http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/
cisco_ios_voice_configuration_library_glossary/
vcl.htm
Cisco IOS Release 15.0
Cisco IOS Release 15.0 Configuration Guides
Cisco IOS Release 12.2
Cisco IOS Voice, Video, and Fax Configuration
Guide, Release 12.2
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
39
Additional References
Standards
Related Topic
internet Low Bitrate Codec (iLBC) Documents
Document Title
•
Codecs section of the Dial Peer Configuration
on Voice Gateway Routers Guide
http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/
dial_peer/ dp_ovrvw.html
•
Dial Peer Features and Configuration section
of the Dial Peer Configuration on Voice
Gateway Routers Guide
http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/
dial_peer/ dp_confg.html
Related Application Guides
•
•
•
Troubleshooting and Debugging guides
•
Cisco Unified Communications Manager and
Cisco IOS Interoperability Guide
Cisco IOS SIP Configuration Guide
Cisco Unified Communications Manager
(CallManager) Programming Guides
Cisco IOS Debug Command Reference,
Release 12.4 at
http://www.cisco.com/en/US/docs/ios/debug/
command/reference/db_book.html
•
•
Troubleshooting and Debugging VoIP Call
Basics at http://www.cisco.com/en/US/tech/
tk1077/technologies_tech_
note09186a0080094045.shtml
VoIP Debug Commands at
http://www.cisco.com/en/US/docs/routers/access/
1700/1750/software/configuration/guide/
debug.html
Standards
Standard
Title
ITU-T G.711
--
MIBs
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40
Additional References
RFCs
MIB
•
•
•
•
•
•
•
•
•
•
MIBs Link
CISCO-PROCESS MIB
CISCO-MEMORY-POOL-MIB
CISCO-SIP-UA-MIB
DIAL-CONTROL-MIB
CISCO-VOICE-DIAL-CONTROL-MIB
CISCO-DSP-MGMT-MIB
IF-MIB
IP-TAP-MIB
TAP2-MIB
USER-CONNECTION-TAP-MIB
To locate and download MIBs for selected
platforms, Cisco IOS XE software releases, and
feature sets, use Cisco MIB Locator found at the
following URL:
http://www.cisco.com/go/mibs
RFCs
RFC
Title
RFC 1889
RTP: A Transport Protocol for Real-Time
Applications
RFC 2131
Dynamic Host Configuration Protocol
RFC 2132
DHCP Options and BOOTP Vendor Extensions
RFC 2198
RTP Payload for Redundant Audio Data
RFC 2327
SDP: Session Description Protocol
RFC 2543
SIP: Session Initiation Protocol
RFC 2543-bis-04
SIP: Session Initiation Protocol, draft-ietf-siprfc2543bis-04.txt
RFC 2782
A DNS RR for Specifying the Location of Services
(DNS SRV)
RFC 2833
RTP Payload for DTMF Digits, Telephony Tones
and Telephony Signals
RFC 3203
DHCP reconfigure extension
RFC 3261
SIP: Session Initiation Protocol
RFC 3262
Reliability of Provisional Responses in Session
Initiation Protocol (SIP)
RFC 3323
A Privacy Mechanism for the Session Initiation
Protocol (SIP)
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
41
Additional References
Technical Assistance
RFC
Title
RFC 3325
Private Extensions to the Session Initiation
Protocol (SIP) for Asserted Identity within Trusted
Networks
RFC 3515
The Session Initiation Protocol (SIP) Refer Method
RFC 3361
Dynamic Host Configuration Protocol (DHCP-forIPv4) Option for Session Initiation Protocol (SIP)
Servers
RFC 3455
Private Header (P-Header) Extensions to the
Session Initiation Protocol (SIP) for the 3rdGeneration Partnership Project (3GPP)
RFC 3608
Session Initiation Protocol (SIP) Extension Header
Field for Service Route Discovery During
Registration
RFC 3711
The Secure Real-time Transport Protocol (SRTP)
RFC 3925
Vendor-Identifying Vendor Options for Dynamic
Host Configuration Protocol version 4 (DHCPv4)
Technical Assistance
Description
Link
The Cisco Support website provides extensive
http://www.cisco.com/cisco/web/support/
online resources, including documentation and tools index.html
for troubleshooting and resolving technical issues
with Cisco products and technologies.
To receive security and technical information about
your products, you can subscribe to various
services, such as the Product Alert Tool (accessed
from Field Notices), the Cisco Technical Services
Newsletter, and Really Simple Syndication (RSS)
Feeds.
Access to most tools on the Cisco Support website
requires a Cisco.com user ID and password.
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
42
Glossary
AMR-NB --Adaptive Multi Rate codec - Narrow Band.
Allow header --Lists the set of methods supported by the UA generating the message.
bind -- In SIP, configuring the source address for signaling and media packets to the IP address of a
specific interface.
call --In SIP, a call consists of all participants in a conference invited by a common source. A SIP call is
identified by a globally unique call identifier. A point-to-point IP telephony conversation maps into a
single SIP call.
call leg --A logical connection between the router and another endpoint.
CLI --command-line interface.
Content-Type header --Specifies the media type of the message body.
CSeq header --Serves as a way to identify and order transactions. It consists of a sequence number and a
method. It uniquely identifies transactions and differentiates between new requests and request
retransmissions.
delta --An incremental value. In this case, the delta is the difference between the current time and the time
when the response occurred. dial peer--An addressable call endpoint.
dial peer --An addressable call endpoint.
DNS --Domain Name System. Used to translate H.323 IDs, URLs, or e-mail IDs to IP addresses. DNS is
also used to assist in locating remote gatekeepers and to reverse-map raw IP addresses to host names of
administrative domains.
DNS SRV --Domain Name System Server. Used to locate servers for a given service.
DSP --Digital Signal Processor.
DTMF --dual-tone multifrequency. Use of two simultaneous voice-band tones for dialing (such as touchtone).
EFXS --IP phone virtual voice ports.
FQDN --fully qualified domain name. Complete domain name including the host portion; for example,
serverA.companyA.com .
FXS --analog telephone voice ports.
gateway --A gateway allows SIP or H.323 terminals to communicate with terminals configured to other
protocols by converting protocols. A gateway is the point where a circuit-switched call is encoded and
repackaged into IP packets.
H.323 --An International Telecommunication Union (ITU-T) standard that describes packet-based video,
audio, and data conferencing. H.323 is an umbrella standard that describes the architecture of the
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43
Glossary
conferencing system and refers to a set of other standards (H.245, H.225.0, and Q.931) to describe its
actual protocol.
iLBC --internet Low Bitrate Codec.
INVITE--A SIP message that initiates a SIP session. It indicates that a user is invited to participate,
provides a session description, indicates the type of media, and provides insight regarding the capabilities
of the called and calling parties.
IP-- Internet Protocol. A connectionless protocol that operates at the network layer (Layer 3) of the OSI
model. IP provides features for addressing, type-of-service specification, fragmentation and reassemble,
and security. Defined in RFC 791. This protocol works with TCP and is usually identified as TCP/IP. See
TCP/IP.
ISDN --Integrated Services Digital Network.
Minimum Timer --Configured minimum value for session interval accepted by SIP elements (proxy,
UAC, UAS). This value helps minimize the processing load from numerous INVITE requests.
Min-SE --Minimum Session Expiration. The minimum value for session expiration.
multicast --A process of transmitting PDUs from one source to many destinations. The actual mechanism
(that is, IP multicast, multi-unicast, and so forth) for this process might be different for LAN technologies.
originator --User agent that initiates the transfer or Refer request with the recipient.
PDU --protocol data units. Used by bridges to transfer connectivity information.
PER --Packed Encoding Rule.
proxy --A SIP UAC or UAS that forwards requests and responses on behalf of another SIP UAC or UAS.
proxy server --An intermediary program that acts as both a server and a client for the purpose of making
requests on behalf of other clients. Requests are serviced internally or by passing them on, possibly after
translation, to other servers. A proxy interprets and, if necessary, rewrites a request message before
forwarding it.
recipient --User agent that receives the Refer request from the originator and is transferred to the final
recipient.
redirect server --A server that accepts a SIP request, maps the address into zero or more new addresses,
and returns these addresses to the client. It does not initiate its own SIP request or accept calls.
re-INVITE --An INVITE request sent during an active call leg.
Request URI --Request Uniform Resource Identifier. It can be a SIP or general URL and indicates the
user or service to which the request is being addressed.
RFC --Request For Comments.
RTP --Real-Time Transport Protocol (RFC 1889)
SCCP --Skinny Client Control Protocol.
SDP--Session Description Protocol. Messages containing capabilities information that are exchanged
between gateways.
session --A SIP session is a set of multimedia senders and receivers and the data streams flowing between
the senders and receivers. A SIP multimedia conference is an example of a session. The called party can
be invited several times by different calls to the same session.
session expiration --The time at which an element considers the call timed out if no successful INVITE
transaction occurs first.
session interval --The largest amount of time that can occur between INVITE requests in a call before a
call is timed out. The session interval is conveyed in the Session-Expires header. The UAS obtains this
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
44
Glossary
value from the Session-Expires header of a 2xx INVITE response that it sends. Proxies and UACs
determine this value from the Session-Expires header in a 2xx INVITE response they receive.
SIP --Session Initiation Protocol. An application-layer protocol originally developed by the Multiparty
Multimedia Session Control (MMUSIC) working group of the Internet Engineering Task Force (IETF).
Their goal was to equip platforms to signal the setup of voice and multimedia calls over IP networks. SIP
features are compliant with IETF RFC 2543, published in March 1999.
SIP URL --Session Initiation Protocol Uniform Resource Locator. Used in SIP messages to indicate the
originator, recipient, and destination of the SIP request. Takes the basic form of user@host , where user is
a name or telephone number, and host is a domain name or network address.
SPI --service provider interface.
socket listener -- Software provided by a socket client to receives datagrams addressed to the socket.
stateful proxy --A proxy in keepalive mode that remembers incoming and outgoing requests.
TCP --Transmission Control Protocol. Connection-oriented transport layer protocol that provides reliable
full-duplex data transmissions. TCP is part of the TCP/IP protocol stack. See also TCP/IP and IP.
TDM --time-division multiplexing.
UA --user agent. A combination of UAS and UAC that initiates and receives calls. See UASand UAC.
UAC --user agent client. A client application that initiates a SIP request.
UAS --user agent server. A server application that contacts the user when a SIP request is received and
then returns a response on behalf of the user. The response accepts, rejects, or redirects the request.
UDP -- User Datagram Protocol. Connectionless transport layer protocol in the TCP/IP protocol stack.
UDP is a simple protocol that exchanges datagrams without acknowledgments or guaranteed delivery,
requiring that error processing and retransmission be handled by other protocols. UDP is defined in
RFC-768.
URI --Uniform Resource Identifier. Takes a form similar to an e-mail address. It indicates the user’s SIP
identity and is used for redirection of SIP messages.
URL --Universal Resource Locator. Standard address of any resource on the Internet that is part of the
World Wide Web (WWW).
User Agent --A combination of UAS and UAC that initiates and receives calls. See UAS and UAC.
VFC --Voice Feature Card.
VoIP --Voice over IP. The ability to carry normal telephone-style voice over an IP-based Internet with
POTS-like functionality, reliability, and voice quality. VoIP is a blanket term that generally refers to the
Cisco standards-based approach (for example, H.323) to IP voice traffic.
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
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Glossary
Cisco Unified Border Element Fundamentals and Basic Setup Configuration Guide, Cisco IOS Release 12.4T
46
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