Re: [SR-Users] WebRTC to PSTN call, proxied through Kamailio

Hello gents, I was trying my hands on getting a successful RTCweb call (JSsip, since Peter Dunkley mentioned that he's been using JSsip for most of the testing scenarios..) to PSTN, making my kamailio as proxy + protocol converter (sip over web-sockets to sip over udp). And yes, I've referred Carlos' config; the main problem is I get 'Bad Media Description' error in Google Chromium (Version 40.0.2214.111 m) & my SIP server even sends 200 OK, but my phone doesn't ring. To make it worse, I can see rtpengine throwing this error - "SRTCP output wanted, but no crypto suite was negotiated" BTW, I have - [root@localhost log]# openssl version OpenSSL 1.0.1j 15 Oct 2014 I even tried building kamailio & rtpengine using this openssl but in-vain. One thing that baffles me is that, apparently kamailio has started receiving RTP packets (perhaps early media) but the mobile phone hasn't ringed :-( I am attaching all possible logs & seek some guidance from the array of experts in this list. Files attached: a) tcpdump on ext. interface b) tcpdump on loopback c) syslogs d) Chromium JS logs UAC (14.98.55.38), Kamailio (125.99.186.126), SIP Server (157.238.178.153), Media Server (199.27.244.6) -- 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