Cisco Cisco IOS Software Release 12.4(4)T Données agrégées
Product Bulletin
© 2009 Cisco Systems, Inc. All rights reserved. This document is Cisco Public Information.
Page 31 of 299
Additional Information
:
Product Management Contact:
Jay Chokshi,
3.3.2) Session Initiation Protocol (SIP) Enhancements
Cisco is consistently leading the development of Session Initiation Protocol (SIP). This is part of
IOS that runs on all routers in the Integrated Services Router (ISR) portfolio. This is also a key
development for the unified communications solution for service providers, Enterprises, SMBs and
small branch offices that provide voice, data, voicemail, Automated-Attendant, video, and security
capabilities.
Cisco is consistently leading the development of Session Initiation Protocol (SIP). This is part of
IOS that runs on all routers in the Integrated Services Router (ISR) portfolio. This is also a key
development for the unified communications solution for service providers, Enterprises, SMBs and
small branch offices that provide voice, data, voicemail, Automated-Attendant, video, and security
capabilities.
In this current release, core components include the following:
●
RSVP Preconditions (RFC3312) for TDM Gateway and Cisco Unified Communications
Manager Express. It extends negotiation of RSVP CAC/QoS across CUCM clusters*,
Gateways, CUCME and CUBE
Manager Express. It extends negotiation of RSVP CAC/QoS across CUCM clusters*,
Gateways, CUCME and CUBE
●
Audio RSVP enhancements to support RE-INVITE or 302-Response based supplementary
services on gateways
services on gateways
●
RSVP support on the SIP trunk of SCCP-CUCME
●
SIP SRTP Fallback to Non-secure RTP and SRTP over sip: scheme for CUBE:
This feature extends the existing SRTP fallback on the SIP-TDM gateway to interoperate with the
SRTP fallback method of CUCM on SIP trunk. It adds the CUCM interoperable SRTP fallback
support to SIP-SIP and SIP-H323 call-flow of CUBE. This is supported on CUBE for the following
call flows—EO-EO, DO-DO, FS-EO, EO-FS, SS-DO:
SRTP fallback method of CUCM on SIP trunk. It adds the CUCM interoperable SRTP fallback
support to SIP-SIP and SIP-H323 call-flow of CUBE. This is supported on CUBE for the following
call flows—EO-EO, DO-DO, FS-EO, EO-FS, SS-DO:
●
SIP Diversion Header Enhancements
●
SIP History INFO (RFC 4244): Many services that SIP is anticipated to support, require the
ability to determine why and how the call arrived at a specific application. SIP History-Info
header provides a standard mechanism for capturing the request history information to
enable a wide variety of services for networks and end-users. The History-Info header
provides a building block for development of new services.
ability to determine why and how the call arrived at a specific application. SIP History-Info
header provides a standard mechanism for capturing the request history information to
enable a wide variety of services for networks and end-users. The History-Info header
provides a building block for development of new services.
●
SIP Multicast Music on Hold: When the IP-Phone puts a call on hold, the CUCM will ask the
MOH server to stream the RTP packets on a pre-configured multicast address. The CM will
also send mid-call Invite with Send-Only attribute and multicast address to the IOS SIP
gateway to listen on that multicast address.
MOH server to stream the RTP packets on a pre-configured multicast address. The CM will
also send mid-call Invite with Send-Only attribute and multicast address to the IOS SIP
gateway to listen on that multicast address.
*Need the correct version of CUCM
Additional Information
:
Product Management Contact:
David Sauerhaft,