Hi Sandeep, You can see how to implement support of Session timer (RFC 4028 ) in SIP. Thanks, Jitendra Singh Bhadoriya From: pjsip [mailto:pjsip-bounces@xxxxxxxxxxxxxxx] On Behalf Of Sandeep Karanth Sent: 23 January 2013 10:05 AM To: pjsip list Subject: Dangling Calls Hi all, Could some one clarify this to me? Suppose I have a UDP call established between Caller A (Some 3rd party app) and Callee B (app which plays a file into call using pjsua-lib). Consider that B expects call tear down to be initiated always from A, i.e A will send BYE to B. Now if the initial call set up has happened and media flows and then in the the middle of the call suppose the Caller A crashes then at the B's end the call would exist forever right! Is there any mechanism in case of UDP in pjsua-lib that B will get to know that the other end doesn't exist anymore and hence the call should be terminated? Any pointers on this would be appreciated. Regards, Sandeep -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20130201/15dc81e2/attachment-0001.html>
