some devices couldn't handle timer re-INVITE correctly. They will always reopen rtp stream and wouldn't check SDP version and number. And too short timer value may rejected by other sip devices. Start a timer at UA2 to query media stream couter will be more reliable. regards, Gang On Mon, Nov 14, 2011 at 9:31 PM, Jacob Holmgaard <jh at danphone.com> wrote: > Hi, > Functionality to handle this kind of problems is already part of the SIP > protocol. > Try looking into "Session Timers". It's described in RFC 4028 "Session > Timers in the Session Initiation Protocol (SIP)" > > Regards > Jacob Holmgaard > > Den 14-11-2011 13:50, Sandeep Karanth skrev: > > Hi all, > ? ? ? ? Could anyone suggest a solution to overcome the following problem, > 1) A softphone (UA_1) is used to call a Pjsip based passive user agent (UA_2 > which only accepts the call and records it) > 2) UA_1 calls UA_2 and media session is established > 3) Now UA_1 application crashes mid-way without sending a BYE to UA_2 > > This situation would cause recording to continue indefinitely,as the > recording UA waits ?for BYE to be sent from UA_1 for the session to > terminate and end the recording. I do not wish to add a timer to end a call > after a fixed duration like the way its done in pjsua_app.c. Is there any > way I can detect this deadlock and break out of it in my appllication.Does > pjsip have any callback to detect a situation like this.Am using the pjsip > version 1.12.Any help on this would be great for me. > -- > Regards, > Sandeep > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >
