<div dir="ltr"><div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, May 3, 2013 at 12:40 AM, mehroz <span dir="ltr">&lt;<a href="mailto:mehroz.ashraf85@gmail.com" target="_blank">mehroz.ashraf85@gmail.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">My configuration of SIP profile are:<br>
<br>
param name=&quot;nat-options-ping&quot; value=&quot;true&quot;<br>
param name=&quot;all-reg-options-ping&quot; value=&quot;true&quot;<br>
param name=&quot;unregister-on-options-fail&quot; value=&quot;true&quot;<br>
<br>
param name=&quot;enable-timer&quot; value=&quot;false&quot;<br>
and this is because, i was experiencing video session starting up within<br>
audio call, SDP session update once session timers completes, That was<br>
totally unexplained, and i eventually had to disable timer, which ultimately<br>
resolved that issue.<br>
<br>
<br>
What left is , RTP timers, and i have<br>
param name=&quot;rtp-timer-name&quot; value=&quot;soft&quot;<br>
param name=&quot;rtp-timeout-sec&quot; value=&quot;15&quot;<br>
 but it is not helpful. Channel hangs up once only a client becomes<br>
unreachable. But if both becomes unreachable, Freeswitch is helpless!!<br>
<br>
<br>
which logs are required to dig it in details and what approach shall be<br>
considered?<br></blockquote><div>Same thing I mentioned before:<br></div><div>pcap with both SIP and RTP<br></div><div>debug log of call not behaving properly.<br></div><div>-MC<br></div></div><br clear="all"><br>-- <br>
Michael S Collins<br>Twitter: @mercutioviz<br><a href="http://www.FreeSWITCH.org" target="_blank">http://www.FreeSWITCH.org</a><br><a href="http://www.ClueCon.com" target="_blank">http://www.ClueCon.com</a><br><a href="http://www.OSTAG.org" target="_blank">http://www.OSTAG.org</a><br>
<br>
</div></div>