Kamailio SIP Server Wiki

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The [[http://www.doubango.org|Doubango Telecom]] [[http://webrtc2sip.org/|webrtc2sip]] gateway includes an RTCWeb Breaker component.  The Doubango RTCWeb Breaker is a B2BUA.  It would be useful to have a Kamailio RTCWeb Breaker that is lighter-weight and which can be used without degrading the SIP signalling by using a B2BUA.

An RTCWeb Breaker converts SDP and media streams between those supported by WebRTC end-points and non-WebRTC end-points.  An RTCWeb Breaker does not need to transcode the media streams; G.711 is supported by most SIP UAs and is a Mandatory To Implement (MTI) codec in WebRTC, and many SIP UAs now support OPUS and VP8 too.

===== Proposed RTCWeb Breaker architecture =====
The RTCWeb Breaker should consist of two components:
  * A Kamailio control module
  * An RTCWeb Breaker daemon

Kamailio should control the RTCWeb Breaker using a light-weight TCP protocol.  SDP should be handled and manipulated by the RTCWeb daemon.

RTCWeb Breaker sessions should be initiated within Kamailio failure routes and branch-failure routes.  In parallel forking scenarios Kamailio should create distinct RTCWeb Breaker sessions in branch-failure routes enabling sessions to be successfully forked to WebRTC and non-WebRTC end-points.

{{devel:breaker2.png}}

===== Example RTCWeb Breaker signalling flow =====
{{devel:breaker.png}}

===== Scenarios ======
The RTCWeb Breaker SHOULD support:
  * Sessions from WebRTC to non-WebRTC end-points
  * Sessions from non-WebRTC to WebRTC end-points
  * Session renegotiation using re-INVITE and UPDATE
  * Early media (SDP in 183, PRACK, and UPDATE)
  * SDP offer in INVITE and answer in 180/183/200
  * SDP offer in 180/183/200 and answer in ACK
  * Parallel forking (distinct RTCWeb Breaker sessions for each branch that requires it)