<p dir="ltr">The trace I posted earlier is what I see with tcpdump when attempting a call. There is no other INVITE shown in the trace: <a href="http://pastebin.com/raw/C4iymTbh">http://pastebin.com/raw/C4iymTbh</a></p>
<p dir="ltr">The trace seems to end abruptly in the middle of the SDP, so I think it could be due to packet fragmentation. </p>
<div class="gmail_quote">On 6 May 2016 4:18 pm, "Bogdan-Andrei Iancu" <<a href="mailto:bogdan@opensips.org">bogdan@opensips.org</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
<tt>So that meas the INVITE never gets to the callee ?? maybe it is
not properly routed . <br>
<br>
Do you see (with ngrep or tcpdump) the INVITE being sent out by
opensips towards callee ?<br>
<br>
Regards,<br>
</tt>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
<div>On 06.05.2016 12:56, Nabeel wrote:<br>
</div>
<blockquote type="cite">
<p dir="ltr">Hi,</p>
<p dir="ltr">Thanks for the idea about packet compression. By
'call fails to connect', I meant the call does not connect to
the callee, ie. the callee's phone does not ring after the
INVITE (despite using TURN server). </p>
<p dir="ltr">This was a public WiFi network and that was all I
could get at the time. I am using OpenSIPS version 2.1.</p>
<p dir="ltr">Nabeel</p>
<div class="gmail_quote">On 6 May 2016 9:16 am, "Bogdan-Andrei
Iancu" <<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>>
wrote:<br type="attribution">
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"> <tt>Hi,<br>
<br>
Hard to analyze a call based on the INVITE packet only :).
Still the SIP signaling does not show any ALG interference
(also not sure if the capture was done before or after the
ALG). Also, what you mean by "call fails" ?no reply,
negative reply , no audio ?<br>
<br>
Regards,<br>
</tt>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
<div>On 05.05.2016 22:35, Nabeel wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div class="gmail_extra"><br>
</div>
<div class="gmail_extra">Please check the following SIP
trace taken within a WiFi network. The call fails to
connect despite the INVITE request and using a
non-standard port. Could this be caused by SIP ALG, or
some unopened RTP port on the router?</div>
<div class="gmail_extra"><br>
</div>
<div class="gmail_extra"><a href="http://pastebin.com/raw/C4iymTbh" target="_blank">http://pastebin.com/raw/C4iymTbh</a><br>
</div>
</div>
<br>
<fieldset></fieldset>
<br>
<pre>_______________________________________________
Users mailing list
<a href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a>
<a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a>
</pre>
</blockquote>
<br>
</div>
</blockquote>
</div>
</blockquote>
<br>
</div>
</blockquote></div>