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. :) Our Test Result:
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