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<tt>Any chance you could provide a pcap capture of the call?<br>
<br>
Best regards,<br>
</tt>
<pre class="moz-signature" cols="72">Răzvan Crainea
OpenSIPS Solutions
<a class="moz-txt-link-abbreviated" href="http://www.opensips-solutions.com">www.opensips-solutions.com</a></pre>
<div class="moz-cite-prefix">On 08/20/2015 01:45 PM, Герман Чичикин
wrote:<br>
</div>
<blockquote cite="mid:1440067559.818370798@f379.i.mail.ru"
type="cite">
Hi, All!<br>
<br>
Having trouble with testing webrtc with a rtpengin. I use
opensips.cfg from opensips.org (Documentation -> Tutorials
-> WebSocket Transport using OpenSIPS), opensips v2.1, clients
- sipml5, jssip demo webphones or my own script in various
combinations. Clients register without problems. When a client
calls another, the callee joyfully calls. But when I pick up the
phone call is dropped, session ends and no any media. The caller
receives a "bye" from opensips. <br>
Anybody can tell in what direction to dig?<br>
<br>
Best regards, Herman.<br>
<br>
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</pre>
</blockquote>
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