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