WebRTC and SIP Over WebSockets

Background

WebSockets is a mechanism for creating sockets from a web browser (typically running Javascript) to a server.

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.

WebRTC is related to WebSockets, but it is not the same thing. WebRTC specifies a way for a browser to act as an RTC endpoint, but not specifically as a SIP endpoint. There are SIP implementations written in Javascript that use the WebSocket transport to create WebRTC sessions, and a correctly adapted repro proxy server should be able to interact with such clients.

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

Related links

Javascript SIP clients for WebRTC capable browsers

• sipML5
• JsSIP

WebRTC capable browsers

• Chrome 25 beta
• Firefox nightly build

Other components

• The SIP client must be hosted on a web server such as Apache