Thanks guys !
I did further investigation of the Chrome logs and found that... (this is really interesting), even though I disabled Video; still JSsip was sending video information in the m & a lines. The fact that I was trying to call PSTN number made it mandatory to set video port to '0' in 183 and 200. However, JSsip was not happy with that and cribbed about codec-formats not being present, ergo "Bad Media Description".
Marc, Could you please share your config so that I'd be sure my kamailio & rtpengine side is in proper shape.
P.S. I am attaching mine here.
On Wed, Feb 11, 2015 at 8:58 PM, Marc Soda msoda@coredial.com wrote:
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
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