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SIP Videowith Communication Manager
Chris Kendall
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2007 Avaya Inc. All rights reserved. Avaya Proprietary
The addition of SIP to Avayas Video Telephony Solution provides theframework 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 todays H.323 video solution Highlights include:
H.323-SIP inter-working, with a common subset of telephony featuresCall admission control and video enablement policiesBandwidth management
Priority video callers Cumulative pools for audio and video Call rate negotiation
Video Conferencing (scheduled and ad-hoc)
Overview DescriptionSIP Video
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Solution view
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Solution ComponentsCM is the
Feature server - extends many CM features to SIP phonesthru 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)
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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!
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Meeting ExchangeMeeting Exchange is the existing large scale SIP conferencing platformfor CM
Version 5 supports video conferencing of up to 16 simultaneoususers.
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
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SIP Video PhonesMost Polycom & Tandberg room systems are dual-protocol H.323 andSIP
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, especiallySoftphones that can be connected to SES
Some will work as OPTIM endpoints, i.e. CM -routed and otherswill not
Avayas next generation SIP Softphone (with video) is TBA
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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
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High-level Call-flow
Edge ProxySIP phone-A
NJ CM
SIP phone-B
DR Home Proxy NJ Home Proxy
DR CM
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Why OPTIM?Using OPTIM architecture, CM becomes a central point of control forvideo 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 differentusers and systems
Dont need to dial special addresses
Dont need to think about protocols or devices Telephony features just work
Video is still as easy as making a phone call
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SIP H.323 InteroperabilityCM places particular requirements on endpoints for H.323interoperability:
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 functionalityrequired by CM
For example, In Counterpath Eyebeam (1.5) you can dial ***7469 toreveal 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
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Solution Limitations
This release has some limitations with CM as the inter-workingSIP-H.323 gateway using video endpoints.
SIP video endpoints are not supported dialing into thePolycom 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 whenconfigured as an OPTIM endpoint
Multipoint SIP room systems are not supported with CM
Future releases aim to resolve these limitations
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SIP Priority Video UsersPriority 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 thecall:
The priority status of the originating caller is preserved
If Priority video is enabled and there is an incoming call on the SIPsignaling group:
The call gets promoted to priority
SIP sub-domains may therefore be used to separate priority users from
regular users [email protected]
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SES Administration
SES Administration is effectively unchanged from existingSIP audio deployments.
Primary consideration is whether a user is mapped to amedia server extension (CM station) or not
Video calls are unmanaged when they are pure SIP anddo not route via CM.
Calling a H.323 user will still route the call through CM if themedia-server address maps are configured
This also holds true for calls involving OPTIM SIP users
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CM Admin LicensingNew 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
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CM Admin SIP Video TrunkTwo new fields for video on the SIP signaling group.
IP Video and Priority Video SIP trunk-group administration is unchanged
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CM Admin OPTIM stationsNew set type 4620SIP should be used for all non-AST SIP endpoints
IP Video should be enabled, while IP Softphone should be disabled
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Common ProblemsThe 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
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References and ResourcesSIP support in Communication Manager 4.0:
http://support.avaya.com/elmodocs2/sip/245206_7.pdf
Installing and administering SES:
http://support.avaya.com/elmodocs2/sip/03_600768_4.pdf
http://support.avaya.com/elmodocs2/sip/245206_7.pdfhttp://support.avaya.com/elmodocs2/sip/03_600768_4.pdfhttp://support.avaya.com/elmodocs2/sip/03_600768_4.pdfhttp://support.avaya.com/elmodocs2/sip/245206_7.pdf