WebRTC and SIP Over WebSockets
draft-ietf-sipcore-sip-websocket defines a way to use WebSockets formally as a transport for SIP. It is not possible to simply view the WebSocket as a tunnel and pass SIP messages through them. Specfically, SIP user agents and proxies must behave slightly differently when WebSockets is used instead of UDP or TCP, and the draft specifies what this means in practice. There is a branch in reSIProcate attempting to implement this behavior.
reSIProcate and WebRTC
- WebRTC specifies that ICE/STUN/TURN support is mandatory in user agents/end-points. The reTurn server project and the reTurn client libraries from reSIProcate can fulfil this requirement.
- WebRTC requires some mechanism for finding peers and initiating calls. SIP over WebSockets, interacting with a repro proxy server can fulfill this task.
Current support for WebSockets
- There is a branch in the repository
- There has been discussion in various threads on the resiprocate-devel mailing list
- and some test results/observations
- Using options like EnableFlowTokens seems to be helpful
WebRTC capable browsers
- The SIP client must be hosted on a web server such as Apache