<div dir="ltr">Our config is based on the example config and the WebRTC bits are based on <span style="font-size:12.8000001907349px">Carlos'. </span>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.<div class="gmail_extra"><br><div class="gmail_quote">On Thu, Feb 12, 2015 at 12:34 PM, Rahul MathuR <span dir="ltr"><<a href="mailto:rahul.ultimate@gmail.com" target="_blank">rahul.ultimate@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Gentle Reminder !<div><br></div><div>Thanks</div><div><br></div><div>Warm Regds,</div><div>Rahul</div></div><div class="gmail_extra"><div><div class="h5"><br><div class="gmail_quote">On Thu, Feb 12, 2015 at 12:13 AM, Rahul MathuR <span dir="ltr"><<a href="mailto:rahul.ultimate@gmail.com" target="_blank">rahul.ultimate@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Thanks guys !<div><br></div><div>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.</div><div>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".</div><div><br></div><div>Marc,</div><div>Could you please share your config so that I'd be sure my kamailio & rtpengine side is in proper shape.</div><div><br></div><div><br></div><div>P.S. I am attaching mine here.</div></div><div class="gmail_extra"><div><div><br><div class="gmail_quote">On Wed, Feb 11, 2015 at 8:58 PM, Marc Soda <span dir="ltr"><<a href="mailto:msoda@coredial.com" target="_blank">msoda@coredial.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">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.<div><br></div><div>I'm not sure about the media description error, however, the crypto error is probably not a real issue. Richard explained it here:</div><div><br></div><div><a href="http://lists.sip-router.org/pipermail/sr-users/2014-December/086271.html" target="_blank">http://lists.sip-router.org/pipermail/sr-users/2014-December/086271.html</a></div><div><br></div><div>I corrected the other issues I was having and that one seemed to resolve itself.</div><div><br></div><div>Hope that helps,</div><div>Marc<br><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Tue, Feb 10, 2015 at 12:01 PM, Rahul MathuR <span dir="ltr"><<a href="mailto:rahul.ultimate@gmail.com" target="_blank">rahul.ultimate@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div><div><div dir="ltr">Hello gents,<div><br></div><div>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).</div><div>And yes, I've referred Carlos' config; the main problem is I get 'Bad Media Description' error in Google Chromium (<span style="color:rgb(48,57,66);font-family:'Segoe UI',Tahoma,sans-serif;font-size:12px">Version 40.0.2214.111 m</span>) & my SIP server even sends 200 OK, but my phone doesn't ring. To make it worse, I can see rtpengine throwing this error - </div><div>"SRTCP output wanted, but no crypto suite was negotiated"</div><div><br></div><div>BTW, I have - </div><div><div>[root@localhost log]# openssl version</div><div>OpenSSL 1.0.1j 15 Oct 2014</div></div><div><br></div><div><div>I even tried building kamailio & rtpengine using this openssl but in-vain.</div><div>One thing that baffles me is that, apparently kamailio has started receiving RTP packets (perhaps early media) but the mobile phone hasn't ringed :-(</div><div><br></div><div>I am attaching all possible logs & seek some guidance from the array of experts in this list.</div><div><br></div><div>Files attached:</div><div>a) tcpdump on ext. interface</div><div>b) tcpdump on loopback</div><div>c) syslogs</div><div>d) Chromium JS logs</div><div><br></div><div>UAC (14.98.55.38), Kamailio (125.99.186.126), SIP Server (157.238.178.153), Media Server (199.27.244.6)</div><span><font color="#888888"><div><br></div><div><br></div><div><br></div>-- <br><div>Warm Regds.<br>MathuRahul<br></div>
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