Cisco Unified Border Element (Enterprise) SIP Support Configuration Guide, Cisco IOS XE Release 3S
SIP-to-SIP Basic Functionality for Session Border Controller
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SIP-to-SIP Basic Functionality for Session Border Controller

SIP-to-SIP Basic Functionality for Session Border Controller

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. The SIP-to-SIP protocol interworking capabilities of the Cisco UBE support the following:

  • Basic voice calls (Supported audio codecs include: G.711, G.729, G.728, G.726, G.723, G.722, AAC_LD, iLBC. Video codecs: H.263, and H.264)
  • Codec transcoding
  • Calling/called name and number
  • Dual-Tone Multifrequency (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
  • Resource Reservation Protocol (RSVP) synchronized with call signaling
  • SIP-SIP Video calls
  • Tool Command Language Interactive Voice Response (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

Your software release may not support all the features documented in this module. For the latest caveats and feature information, see Bug Search Tool and 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 module.

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-to-SIP Basic Functionality for Session Border Controller

Cisco Unified Border Element

  • Cisco IOS Release 12.4(4)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.

Feature Information for SIP-to-SIP Basic Functionality for Session Border Controller

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.

ISR Feature History Information

Table 1 Feature Information for Configuring SIP-to-SIP Supplementary 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.

ASR Feature History Information

Table 2 Feature Information for Configuring SIP-to-SIP Supplementary Features

Feature Name

Releases

Feature Information

SIP-to-SIP Basic Functionality for Session Border Controller

Cisco IOS XE Release 3.1S,

Cisco IOS XE Release 3.3S

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.