Re: [SR-Users] Need help on WebRTC with Kamailio as proxy

Hello, I am totally struck at a point while implementing Kamailio as proxy for WebRTC enabled UAC (Jssip). I am using Google's TURN server (rfc5766-turn-server for ICE/STUN). I am able to get to the point where the SIP server sends 183 session in progress to kamailio but after that I can only see - "STUN: using this candidate" "Successful STUN binding request from .." "SRTP output wanted, but no crypto suite was negotiated" a) What should I do to resolve this issue ? b) Why is that I never get 200 OK for the INVITE ? I am attaching the logfile & configs herewith. Can somebody please help me out here... -- Warm Regds. MathuRahul _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users