Cisco Unified Border Element Configuration Guide
Feature Information for Cisco UBE SIP Support
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Feature Information for Cisco UBE SIP Support

Table Of Contents

Feature Information for Cisco UBE SIP Support


Feature Information for Cisco UBE SIP Support

Table 1 lists the release history for this chapter.

Use Cisco Feature Navigator to find information about platform support and software image support. Cisco Feature Navigator enables you to determine which 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.


Note Table 1 lists only the Cisco IOS 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.


Table 1 Feature Information for Cisco UBE SIP Support Features 

Feature Name
Releases
Feature Information

SIP-to-SIP Basic Functionality for Session Border Controller

12.4(4)T

The SIP-to-SIP Basic Functionality for Session Border Controller (SBC) for Cisco Unified Border Element (Cisco UBE) feature provides termination and re-origination of both signaling and media between VoIP and video networks using SIP signaling in conformance with RFC 3261.

This feature uses no new or modified commands.

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).

This feature uses no new or modified commands.

SIP-to-SIP Supplementary Services for Session Border Controller

12.4(9)T,

The SIP-to-SIP Supplementary Services for Session Border Controller feature enhances terminating and re-originating signaling and media between VoIP and Video networks

This feature uses no new or modified commands.

Transparent Tunneling of QSIG and Q.931 over SIP TDM Gateway and SIP-SIP Cisco Unified Border Element

12.4(15)XZ
12.4(20)T

This feature adds support for transparent tunneling of all Q.931 messages over SIP and for the Transparent Tunneling of QSIG and Q.931 over a SIP-SIP Cisco Unified Border Element.

Transparent tunneling is accomplished by encapsulating QSIG or Q.931 messages within SIP message bodies. These messages are encapsulated using "application/qsig" or "application/x-q931" Multipurpose Internet Mail Extensions (MIME) to tunnel between SIP endpoints. Using MIME to tunnel through Cisco SIP messaging does not include any additional QSIG/Q.931 services to SIP interworking.

This feature uses no new or modified commands.

SIP Parameter Modification

12.4(15)XZ
12.4(20)T

Allows users to change the standard SIP messages sent from the Cisco SIP stack for better interworking with different SIP entities.

This feature introduces or modifies the following commands: voice class sip-profiles, voice-class sip profiles

SIP Diversion Header Enhancements

12.4(22)T

The SIP Diversion Header Enhancements feature enables time-division multiplex (TDM) gateways and Cisco Unified Communications Manager Express to populate the SIP Diversion Header with a domain name. This feature also provides choice of transparent pass through or application of address hiding to the SIP Diversion Header on Cisco UBE platforms.

This feature modifies the following commands: host-registrar, and registrar

Cisco Unified Border Element Support for Configurable Pass-through of SIP INVITE Parameters

15.0(1)M

This feature enables the Cisco Unified Border Element platform to pass through end-to-end headers at a global or dial-peer level, that are not processed or understood in a SIP trunk to SIP trunk scenario. The pass through functionality includes all or only a configured list of unsupported or non-mandatory SIP headers, and all unsupported content/MIME types.

This feature introduces the following commands: pass-thru and voice-class sip pass-thru.

SIP—Ability to Send a SIP Registration Message on a Border Element

12.4(24)T

Provides the ability to send a SIP Registration Message from Cisco Unified Border Element.

The following command was modified: credentials (SIP UA)

SIP—INFO Method for DTMF Tone Generation

12.2(11)T
12.3(2)T
12.2(8)YN
12.2(11)YV
12.2(11)T
12.2(15)T

The SIP—INFO Method for DTMF Tone Generation feature uses the Session Initiation Protocol (SIP) INFO method to generate dual-tone multifrequency (DTMF) tones on the telephony call leg. SIP methods, or request message types, request a specific action be taken by another user agent (UA) or proxy server. The SIP INFO message is sent along the signaling path of the call.

The following command was introduced: show sip-ua.

Support for PAID, PPID, Privacy, PCPID, and PAURI Headers on the Cisco UBE

12.4(22)YB
15.0(1)M

This feature enables Cisco UBE platforms to support:

P-Preferred Identity (PPID), P-Asserted Identity (PAID), Privacy, P-Called Party Identity (PCPID), in INVITE messages

Translation of PAID headers to PPID headers and vice versa

Translation of From: or RPID headers to PAID or PPID headers and vice versa

Configuration and/or pass through of privacy header values

PCPID header to route INVITE messages

Multiple PAURI headers in the response messages (200 OK) it receives to REGISTER messages

P-Preferred Identity and P-Asserted Identity Headers

The following commands were introduced: call-route p-called-party-id, privacy-policy, random-contact, random-request-uri validate, voice-class sip call-route p-called-party-id, voice-class sip privacy-policy, voice-class sip random-contact, and voice-class sip random-request-uri validate.

Configuring Selective Filtering of Outgoing Provisional Response on the Cisco UBE

12.4(22)YB
15.0(1)M

This feature adds support on Cisco UBE for selective filtering of outgoing provisional responses, including "180-Alerting" and "183-Session In Progress" responses. Selective filtering can be further based on the availability of media information in the received provisional response.

The following commands were introduced or modified: block and voice-class sip block.

RFC 4040-Based Clear Channel Codec Negotiation for SIP Calls

15.0(1)XA
15.1(1)T

This feature adds support for RFC 4040-based clear channel codec Negotiation for SIP calls.

The following commands were modified: encap clear-channel standard and voice-class sip encap clear-channel

Support for Expires Timer Reset on Receiving or Sending SIP 183 Message

15.0(1)XA
15.1(1)T

This feature enables support for resetting the Expires timer upon receipt of SIP 183 messages on Cisco Unified Communications Manager Express (Cisco Unified CME), a Cisco IOS voice gateway, or a Cisco Unified Border Element (Cisco UBE).

The following commands were introduced or modified: reset timer expires and voice-class sip reset timer expires.

Support for Multiple Registrars on SIP Trunks

15.0(1)XA
15.1(1)T

This feature provides support for multiple registrars on SIP trunks on Cisco IOS SIP TDM gateways, Cisco Unified CME, and Cisco UBEs. This feature allows for a redundant registrar for each SIP trunk and enables registrar redundancy across multiple service providers.

This feature includes the following new or modified commands: credentials, localhost, registrar, voice-class sip localhost.

Cisco UBE Support for generating Out-of-dialog SIP OPTIONS Ping messages to monitor SIP Servers

15.1(1)T

This feature provides option to configure the error response code when a dial peer is busied out because of an Out-of-Dialog OPTIONS ping failure.

The following commands were introduced or modified in this release: error-code-override options-keepalive failure, voice-class sip error-code-override options-keepalive failure.

Configuring Support for SIP 181 Call is Being Forwarded Message

12.2(13)T

This feature allows users to configure support for SIP 181 Call is Being Forwarded messages either globally or on a specific dial-peer.

This feature includes the following new or modified commands: block, map resp-code, voice-class sip block, voice-class sip map resp-code.

SIP—Configurable Hostname in Locally Generated SIP Headers

12.4(2)T

This feature allows you to configure the hostname in locally generated SIP headers in global and dial-peer-specific configuration modes.

The following commands were introduced or modified: localhost dns and voice-class sip localhost dns

SIP—Core SIP Technology Enhancements

12.2(13)T
12.2(15)T

Compliance to RFC 2543-bis-04 adds enhanced SIP support and ensures smooth interoperability and compatibility with multiple vendors.

The following commands were modified: debug ccsip messages, show sip-ua map, show sip-ua statistics, and.

SIP—Enhanced 180 Provisional Response Handling

12.2(11)T
12.2(8)YN
12.2(15)T
12.2(11)YV
12.2(11)T

The Session Initiation Protocol (SIP) Enhanced 180 Provisional Response Handling feature provides the ability to enable or disable early media cut-through on Cisco IOS gateways for SIP 180 response messages.

The following commands were introduced or modified: disable-early-media 180 and show sip-ua status.

SIP—Session Timer Support

12.2(8)YN
12.2(15)T
12.2(11)YV
12.2(11)T
12.3(2)T

The SIP Session Timer Support feature adds the capability to periodically refresh Session Initiation Protocol (SIP) sessions by sending repeated INVITE requests. The repeated INVITE requests, or re-INVITEs, are sent during an active call leg to allow user agents (UAs) or proxies to determine the status of a SIP session.

The following commands were introduced or modified: min-se (SIP) and show sip-ua min-se.


Any Internet Protocol (IP) addresses used in this document are not intended to be actual addresses. Any examples, command display output, and figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses in illustrative content is unintentional and coincidental.

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