<div dir="ltr">Ok i am ready with logs. About gist may I use private option as traces have our IPs, user</div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme <span dir="ltr"><<a href="mailto:eric@uphreak.com" target="_blank">eric@uphreak.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
Hey John,<br>
<br>
Please paste a full UNALTERED sip trace into a gist
(<a href="http://gist.github.com" target="_blank">gist.github.com</a>) from the proxy servers perspective and provide a
link so that we can see what comes in, and what goes out from both
sides.<br>
<br>
EG: ngrep -qtd any -W byline port 5060<br>
<br>
This will show us the traffic that is leaving the proxy destined for
the Freeswitch box, and what the freeswitch box sends back.<br>
<br>
Also - you can look in your browsers console log and provide the SIP
trace from there in a seperate gist, so that we can see what
opensips sends back up to your browser.<span class="HOEnZb"><font color="#888888"><br>
<br>
-Eric</font></span><div><div class="h5"><br>
<br>
<blockquote type="cite">
<div dir="ltr">
<div><span style="font-size:12.8px">Am I using correct sip.js
example? I copied it to my server and accessing it using
https: (used letsencrypt)</span></div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Thu, Jun 23, 2016 at 7:58 PM, Eric
Tamme <span dir="ltr"><<a href="mailto:eric@uphreak.com" target="_blank">eric@uphreak.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF"> 1. I would suggest
using SIP.js - <a href="https://github.com/onsip/SIP.js" target="_blank">https://github.com/onsip/SIP.js</a>
it is a much more active project that sipml5.<br>
<br>
2. Im guessing that you are not properly passing flags to
RTPEngine. If you want to have DTLS-SRTP between the
browser, and plain RTP/AVP between RTPEngine and
freeswitch, you need to "offer" rtp/avp to freeswitch, and
"answer" dtls-srtp back up to the browser.<br>
<br>
the offer to freeswitch would be: <br>
<pre> $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";
</pre>
and the answer back up to the browswer would be:<br>
<br>
<pre> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";</pre>
<br>
-Eric
<div>
<div><br>
<br>
<br>
<div>On 06/23/2016 08:20 AM, John Nash wrote:<br>
</div>
</div>
</div>
<blockquote type="cite">
<div>
<div>
<div dir="ltr">I am following <a href="http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2" target="_blank"></a><a href="http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2" target="_blank">http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2</a>
and trying to test a call
<div><br>
</div>
<div>sipml5 ----------->Opensips + rtpengine
--------> SIP end point (Freeswitch)<br>
<div><br>
</div>
<div>But I do not have any audio on both sides.
I see this error at rtpengine log "SRTP output
wanted, but no crypto suite was negotiated"<br>
</div>
</div>
<div><br>
</div>
<div>Anyone tested this scenario positive?</div>
</div>
<br>
<fieldset></fieldset>
<br>
</div>
</div>
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