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 <ms...@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.ultim...@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 > > -- Warm Regds. MathuRahul
kamailio-ws.cfg
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