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