On Fri, May 3, 2013 at 12:40 AM, mehroz <mehroz.ashraf85@gmail.com> wrote:
My configuration of SIP profile are:

param name="nat-options-ping" value="true"
param name="all-reg-options-ping" value="true"
param name="unregister-on-options-fail" value="true"

param name="enable-timer" value="false"
and this is because, i was experiencing video session starting up within
audio call, SDP session update once session timers completes, That was
totally unexplained, and i eventually had to disable timer, which ultimately
resolved that issue.


What left is , RTP timers, and i have
param name="rtp-timer-name" value="soft"
param name="rtp-timeout-sec" value="15"
 but it is not helpful. Channel hangs up once only a client becomes
unreachable. But if both becomes unreachable, Freeswitch is helpless!!


which logs are required to dig it in details and what approach shall be
considered?
Same thing I mentioned before:
pcap with both SIP and RTP
debug log of call not behaving properly.
-MC


--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org