RTP&SIP
two great misunderstood protocols that should really pair up!
http://en.wikipedia.org/wiki/Real-time_Transport_Protocol
http://www.sipfoundry.org/ Open source VOIP
howto configure Click to dial on sipX
So now that we have a good look at establishing in the Session Description Protocol (SDP) as a driver for compatibility negotiation to bridge a media connection for transport-level endpoint mapping of any media payload format, how can we carry RTP across it… ?
“RTP: A Transport Protocol for Real-Time Applications”, RFC 3550, July 2003.
Note that RTP itself does not provide any mechanism to ensure timely delivery or provide other quality-of-service guarantees, but relies on lower-layer services to do so. that is why we target the use and flexibility of the lightweight SIP/SDP.
RTP is a protocol framework that is deliberately not complete. Therefore we can INVENT new uses such as this post attempts !
If both audio and video media are used in a conference, they are transmitted as separate RTP sessions. That is, separate RTP and RTCP packets are transmitted for each medium using two different UDP port pairs and/or multicast addresses. There is no direct coupling at the RTP level between the audio and video sessions, except that a user participating in both sessions should use the same distinguished (canonical) name in the RTCP packets for both so that the sessions can be associated.
An RTP mixer would not be able to combine interleaved streams of incompatible media into one stream.
receivers require a CNAME to keep track of each participant
http://bluecoat.com/
perhaps I can write a widget that takes advantage of DOjO