Table Of Contents

SIP-to-SIP Basic Functionality for Session Border Controller

Prerequisites

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

Prerequisites

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.