I have started to play around with webRtc. My first work is to implement websocket support in our existing SIP server of our company. Which is going public right now as named as SIP.ANYFIREWALL.COM. One of our company's webrtc support is allready running on cloud for TURN/STUN or as Relay server. This can be found on TURN.ANYFIREWALL.COM .
So I have implemented the Web SIP support (WebSocket with SIP protocol support) and seen some interesting output when I tested it sipml5.org 's webrtc client. I feel pleasure to say that it's working fine when I tested some Video call.
So feel free to test your websocket SIP client with our Web SIP server when it goes public. :)
