1. SIP Video with Communication Manager
Chris Kendall

2. Overview Description SIP Video
The addition of SIP to Avaya’s Video Telephony Solution provides the framework for the future of enterprise video conferencing
Leveraging existing IETF open standards
Ensuring higher levels of interoperability with third parties
Maintaining a sophisticated feature set that is consistent and compatible with today’s H.323 video solution
Highlights include:
H.323-SIP inter-working, with a common subset of telephony features
Call admission control and video enablement policies
Bandwidth management
Priority video callers
Cumulative pools for audio and video
Call rate negotiation
Video Conferencing (scheduled and ad-hoc)

3. Solution view

4. Solution Components CM is the SES contains the following components
Feature server - extends many CM features to SIP phones thru OPTIM architecture
Gateway – interworks SIP to all other protocols (H.323, ISDN, DCP, analog, etc..)
Back-to-back user-agent (B2BUA)
SES contains the following components
Proxy
Registrar
Event Server
Personal Profile Manager (PPM)
Location service (database)

5. Solution Components (continued)
Supported SIP Adjuncts
Modular Messaging
Voice Portal
Meeting Exchange (version 5 with Video)
Supported SIP Endpoints
Avaya SIP Softphone (AST)
46xx SIP
96xx SPARK phone (AST)
Toshiba SIP phone (AST)
One-X mobile edition (CHAMP)
Cisco and other third-party SIP phones
SIP Video phones
NEW!
NEW!

6. Meeting Exchange
Meeting Exchange is the existing large scale SIP conferencing platform for CM
Version 5 supports video conferencing of up to 16 simultaneous users.
Meet-me, scheduled and ad-hoc conferences may be used
Configured as a video-enabled trunk on CM
Video-bridge configuration must be set for ad-hoc conferencing

7. SIP Video Phones
Most Polycom & Tandberg room systems are dual-protocol H.323 and SIP
Tandberg T150 (v4.1) has been tested extensively with CM in SIP
For presentations and collaboration, H.323 is still recommended
There are many third party SIP phones out in the market, especially Softphones that can be connected to SES
Some will work as OPTIM endpoints, i.e. “CM-routed” and others will not
Avaya’s next generation SIP Softphone (with video) is TBA

8. Jargon Buster Lots of jargon and acronyms to digest…
SIP: Session Initiation Protocol
SDP: Session Description Protocol
Dialog: A conversation/communication
Session: Media (audio/video/data) between two entities
AST: Advanced SIP Telephony
SUSHI: Toshiba SIP Phone (supporting AST)
B2BUA: Back-to-Back User Agent
OPTIM/OPS: Off PBX/Premises Station
OATS: Origination And Termination Service

9. High-level Call-flow SIP phone-A Edge Proxy SIP phone-B INVITE sip:b
DR Home Proxy
NJ Home Proxy
INVITE sip:b
INVITE sip:b
INVITE sip:b
INVITE sip:b
DR CM
NJ CM

10. Why OPTIM? Video is still as easy as making a phone call
Using OPTIM architecture, CM becomes a central point of control for video policy
BWM, CAC, video and conferencing capabilities
As a B2BUA, CM will modify the SDP to enforce these policies
Allow or deny video
Restrict call-rates
Allow/deny use of shared resources
As a SIP-H.323 gateway, CM bridges the protocol gap between different users and systems
Don’t need to dial special addresses
Don’t need to think about protocols or devices
Telephony features “just work”
Video is still as easy as making a phone call

11. SIP – H.323 Interoperability
CM places particular requirements on endpoints for H.323 interoperability:
RFC3890, for enterprise bandwidth management
SIP endpoints to present their full capabilities in the SDP
Supporting asymmetric SDP payload types is also required
Some SIP endpoints can be configured to support the functionality required by CM
For example, In Counterpath Eyebeam (1.5) you can dial ***7469 to reveal a hidden config UI and set the following options:
Rtp:media:send_bandwidth_modifier = 1
Media:sdp:specify_all_codecs_in_offer_answer = 1
Media:sdp:force_describe_well_known_codecs = 1

12. Solution Limitations
This release has some limitations with CM as the inter-working SIP-H.323 gateway using video endpoints.
SIP video endpoints are not supported dialing into the Polycom MGC (either H.323 or SIP)
SIP video endpoints cannot view H.239 streams (e.g. Polycom “People plus Content”)
Polycom SIP firmware is not currently supported when configured as an OPTIM endpoint
Multipoint SIP room systems are not supported with CM
Future releases aim to resolve these limitations

13. SIP – Priority Video Users
Priority video is not signaled over SIP trunks
So it works a bit differently
If Priority video is enabled and the SIP signaling group is terminating the call:
The priority status of the originating caller is preserved
If Priority video is enabled and there is an incoming call on the SIP signaling group:
The call gets promoted to priority
SIP sub-domains may therefore be used to separate priority users from regular users

14. SES Administration
SES Administration is effectively unchanged from existing SIP audio deployments.
Primary consideration is whether a user is mapped to a media server extension (CM station) or not
Video calls are unmanaged when they are “pure SIP” and do not route via CM.
Calling a H.323 user will still route the call through CM if the media-server address maps are configured
This also holds true for calls involving OPTIM SIP users

15. CM Admin – Licensing New Feature to be enabled in license file.
Multimedia IP SIP Trunking? (FEAT_MMIP_SIP)
“Multimedia IP SIP Trunking?” can then be enabled in customer options

16. CM Admin – SIP Video Trunk
Two new fields for video on the SIP signaling group.
‘IP Video’ and ‘Priority Video’
SIP trunk-group administration is unchanged

17. CM Admin – OPTIM stations
New set type ‘4620SIP’ – should be used for all non-AST SIP endpoints
‘IP Video’ should be enabled, while ‘IP Softphone’ should be disabled

18. Common Problems
“The call fails to connect when calling from a SIP endpoint”
If response is “403 forbidden (Denial 11)”, check COR/COS, ip-network-region domain
If response is “482 Loop detected”, check dial plan (UDP/AAR/EXT)
If response is “488 Not acceptable”, check audio codec is supported
“I get no video when calling from a SIP endpoint”
Check licensing and capacities
Check signaling group, network-region and codec-set admin
Check station admin - ensure IP-direct audio enabled for station form page 2
Check bandwidth is available
Check endpoint enabled for video, and is offering bandwidth
Due to protocol complexity, SIP users will not get video dialing into the H.323 MGC via CM
“My SIP endpoint wont register”
If response is “401 Unauthorized”, check password
If response is “404 User not found”, check username and domain are correct,
check that the SES configuration has been saved/updated
More items in the AVTS configuration checklist…

19. References and Resources
SIP support in Communication Manager 4.0:
Installing and administering SES: