Our config is based on the example config and the WebRTC bits are based on Carlos'. I've attached the relevant parts. It's pretty heavily customized to our specific environment. The main differences are the way that we detect a video call, how we route to our backend servers, and that we send video calls directly to a registered peer and not the the backend Asterisk servers.On Thu, Feb 12, 2015 at 12:34 PM, Rahul MathuR <rahul.ultimate@gmail.com> wrote:Gentle Reminder !ThanksWarm Regds,Rahul--On Thu, Feb 12, 2015 at 12:13 AM, Rahul MathuR <rahul.ultimate@gmail.com> wrote: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:I corrected the other issues I was having and that one seemed to resolve itself.Hope that helps,MarcOn 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 versionOpenSSL 1.0.1j 15 Oct 2014I 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. interfaceb) tcpdump on loopbackc) syslogsd) Chromium JS logsUAC (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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Warm Regds.
MathuRahulWarm Regds.
MathuRahul
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