We are in the middle of designing a similar solution with Kamailio and rtpengine and after some initial problems things are going really well.  I can tell you that we ended up going with SIPjs over JSSip and it handled a lot of the weird browser specific issues we were having.

I'm not sure about the media description error, however, the crypto error is probably not a real issue.  Richard explained it here:

http://lists.sip-router.org/pipermail/sr-users/2014-December/086271.html

I corrected the other issues I was having and that one seemed to resolve itself.

Hope that helps,
Marc

On Tue, Feb 10, 2015 at 12:01 PM, Rahul MathuR <rahul.ultimate@gmail.com> wrote:
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

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