RealPresence Collaboration Server (RMX) 1500/1800/2000/4000 Administrator’s Guide
Polycom®, Inc. 847
BFCP Over UDP – AS-SIP Content
In compliance with UCR 2008 Change 3, AS-SIP (Assured Services-Session Initiation Protocol)
Content flow is an implementation of SIP that utilizes SIP’s built in security features.
When using AS-SIP Content, the media line of the content channel is not sent as part of the initial SDP
Offer/Answer message sequence. The media line of the Content channel is only sent to the MCU when
an endpoint wanting to share Content initiates Content sharing. The Collaboration Server (RMX) then
sends the Content media line to all conference participants using an SDP Re-invite.
Guidelines
●AS-SIP Content is shared using Multiple Resolutions (Content Transcoding) and is not supported
in any other Content sharing mode such as H.263 Content and H.264 Cascade and SVC
Optimized Content Protocol.
●Multiple Resolutions consumes system video resources. If sufficient system video resources are
not available, a conference with AS-SIP Content enabled in its Profile, will not be created. An error:
Conference could not be created due to lack of Content DSP resources, is displayed.
●The SIP BFCP UDP application line is included in SDP Offer/Answer message sequence.
●An endpoint declaring SIP BFCP TCP is connected with video and audio but without Content. The
SIP BFCP TCP channel will not be connected.
●The following resolutions are supported with H.264 HD Content protocol. Only when H.264 HD is
selected, these resolutions are enabled for selection:
HD 720p5
HD 720p30
HD 1080p15
●Endpoints that do not support receiving H.264 Content at a resolution of HD 720p5 or greater are
considered Legacy Endpoints and will receive Content using the people video channel.
●Endpoints that do not support transmitting H.264 Content at a resolution of HD 720p5 or greater are
considered Legacy Endpoints and will transmit Content using the people video channel. Depending
on the endpoint type, these endpoints may not be able to transmit Content at all - this is dependent
on the endpoint and is not controlled by the RMX.
●A mixture of older, non AS-SIP compliant and AS-SIP compliant endpoints are supported in the same
conference and are able to share Content.
●An endpoint connecting during a Content session is immediately sent an SDP Re-invite that
includes the connect media line and will receive Content.
●An endpoint connecting after Content started and was stopped will receive the SDP Re-invite and
the content media line only after a new Content request is sent.
PREFER_IPv6 sdp-anat appears in SIP headers.
Dial in: IPv6 is the IP Version preference
Dial out: IPv6 is advertised first.
ANAT_IP_PROTOCOL System Flag Values for Dial in Dial out
Value Behavior - Dial in and Dial out