Bfcp Over Udp – As-Sip Content - Polycom RealPresence RMX 1500 Administrator's Manual

ANAT_IP_PROTOCOL System Flag Values for Dial in Dial out
Value
PREFER_IPv6
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 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.
Polycom®, Inc.
Behavior - Dial in and Dial out
sdp-anat appears in SIP headers.
Dial in: IPv6 is the IP Version preference
Dial out: IPv6 is advertised first.
Ultra Secure Mode
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