Hi Rahul,
Check your record-route, it should not be static. ( maybe use advertised_address and allow double record-route for rr module ).
My 2 cents.
Le 26/01/2015 11:21, Rahul MathuR a écrit :
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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users