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SIP-to-SIP Basic Functionality for Session Border Controller

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SIP-to-SIP Basic Functionality for Session Border Controller

Table Of Contents

SIP-to-SIP Basic Functionality for Session Border Controller

Prerequisites for 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) and for Cisco Unified Border Element (Cisco UBE) feature provides termination and reorigination 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 are: G.711, G.729, G.726, G.728, G.722, G.723, 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

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.