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SIP Video with Communication Manager

SIP Video with Communication Manager

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SIP Video with Communication Manager

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  1. SIP Videowith 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 • 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 INVITE sip:b INVITE sip: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? • 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: •