Cisco Unified Border Element H.323-to- SIP Internetworking Configuration Guide,

Cisco Unified Border Element H.323-to- SIP Internetworking Configuration Guide,
Cisco Unified Border Element H.323-toSIP Internetworking Configuration Guide,
Cisco IOS Release 12.4T
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CONTENTS
Cisco Unified Border Element H.323-to-SIP Interworking 1
Finding Feature Information 1
Cisco Unified Border Element H.323-to-SIP Interworking Features 1
Session Border Controller Enhancements 3
Prerequisites for Session Border Controller Enhancements 3
Feature Information for Session Border Controller Enhancements 4
Additional References 7
Related Documents 7
Standards 8
MIBs 8
RFCs 9
Technical Assistance 10
Glossary 11
Cisco Unified Border Element H.323-to-SIP Internetworking Configuration Guide, Cisco IOS Release 12.4T
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Contents
Cisco Unified Border Element H.323-to-SIP Internetworking Configuration Guide, Cisco IOS Release 12.4T
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Cisco Unified Border Element H.323-to-SIP
Interworking
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.
Note
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
Cisco Unified Border Element H.323-to-SIP Interworking Features, page 1
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 H.323-to-SIP Interworking
Features
This chapter contains the following configuration topics:
Cisco UBE Prerequisites and Restrictions
•
•
Prerequisites for Cisco Unified Border Element (Enterprise)
Restrictions for Cisco Unified Border Element (Enterprise)
Cisco Unified Border Element H.323-to-SIP Internetworking Configuration Guide, Cisco IOS Release 12.4T
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Cisco Unified Border Element H.323-to-SIP Interworking
H.323-SIP Protocol Handling and Supplementary Services
•
H.323-to-SIP Supplementary Feature Interworking for Session Border Controller
◦
H323-to-SIP Back-to-Back Support
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 H.323-to-SIP Internetworking Configuration Guide, Cisco IOS Release 12.4T
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Session Border Controller Enhancements
The H.323-SIP and SIP-SIP Supplementary Services, Transcoding Optimization, and Firewall Integration
feature provides enhancements for H.323-to-SIP Supplementary Services--specifically, mapping ECS to
ReINVITE and ECS to REFER and for SIP-to-SIP Supplementary Services mapping ReINVITE to
ReINVITE on the Cisco Session Border Controller. This feature also provides enhancements in
Transcoding Performance and support for Universal Transcoding Support as well as integration of Cisco
IOS Firewall with the Session Border Controller. Feature highlights include the following:
•
•
•
•
•
•
•
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•
•
•
•
•
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
TCP and UDP interworking
Tcl scripts with SIP NOTIFY VoiceXML with SIP-to-SIP
Transport Layer Security (TLS)
Prerequisites for Session Border Controller Enhancements, page 3
Feature Information for Session Border Controller Enhancements, page 4
Prerequisites for Session Border Controller Enhancements
Cisco Unified Border Element
•
Cisco IOS Release 12.4(11)XW 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.
Cisco Unified Border Element H.323-to-SIP Internetworking Configuration Guide, Cisco IOS Release 12.4T
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Session Border Controller Enhancements
Feature Information for Session Border Controller Enhancements
Feature Information for Session Border Controller
Enhancements
GUID-704DF1EE-C5CE-4FB6-BA02-547A7DE369ED2 Feature History Table entry for the Cisco Unified
Border Element.
Table 1
Feature Name
Feature Information for Configuring Session Border Controller Enhancements for H.323-to-SIP and
SIP-to-SIP Supplementary Services, Transcoding Optimization, and Firewall Integration
Releases
Session Border Controller
12.4(11)XW 12.4(20)T
Enhancements for H.323-SIP and
SIP-SIP Supplementary Services,
Transcoding Optimization, and
Firewall Integration
Feature Information
The H.323-to-SIP and SIP-to-SIP
Supplementary Services,
Transcoding Optimization, and
Firewall Integration feature
provides enhancements for H.
323-SIP Supplementary Services
on the Cisco Session Border
Controller. This feature also
provides enhancements in
Transcoding Performance and
support for Universal
Transcoding Support as well as
integration of Cisco IOS Firewall
with the Session Border
Controller.
GUID-EC201279-0F00-428F-913B-F49E325D87E82 Feature History Table entry for the Cisco Unified
Border Element (Enterprise).
Cisco Unified Border Element H.323-to-SIP Internetworking Configuration Guide, Cisco IOS Release 12.4T
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Session Border Controller Enhancements
Table 2
Feature Information for Configuring Session Border Controller Enhancements for H.323-to-SIP and
SIP-to-SIP Supplementary Services, Transcoding Optimization, and Firewall Integration
Feature Name
Releases
Session Border Controller
Cisco IOS XE Release 3.1S
Enhancements for H.323-SIP and
SIP-SIP Supplementary Services,
Transcoding Optimization, and
Firewall Integration
Feature Information
The H.323-to-SIP and SIP-to-SIP
Supplementary Services,
Transcoding Optimization, and
Firewall Integration feature
provides enhancements for H.
323-SIP Supplementary Services
on the Cisco Session Border
Controller. This feature also
provides enhancements in
Transcoding Performance and
support for Universal
Transcoding Support as well as
integration of Cisco IOS Firewall
with the Session Border
Controller.
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 H.323-to-SIP Internetworking Configuration Guide, Cisco IOS Release 12.4T
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Session Border Controller Enhancements
Cisco Unified Border Element H.323-to-SIP Internetworking Configuration Guide, Cisco IOS Release 12.4T
6
Additional References
The following sections provide references related to the Cisco Unified Border Element (Enterprise)
Configuration Guide.
•
•
•
•
•
Related Documents, page 7
Standards, page 8
MIBs, page 8
RFCs, page 9
Technical Assistance, page 10
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 H.323-to-SIP Internetworking Configuration Guide, Cisco IOS Release 12.4T
7
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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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)
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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 H.323-to-SIP Internetworking Configuration Guide, Cisco IOS Release 12.4T
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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
Cisco Unified Border Element H.323-to-SIP Internetworking Configuration Guide, Cisco IOS Release 12.4T
11
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
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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.
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Glossary
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