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SIP Video with Communication Manager

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Presentation on theme: "SIP Video with Communication Manager"— Presentation transcript:

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:

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