> > I have the following situation that can result in a resource leak > > Call setup between music on hold ( or some long playing > stream that does not hang up ) and caller who calls in via the ITSP. > Remote end drops call without bye. > > > I have to do some liveness testing and destroy the stream. > The question is how to do this? > > Initially, I thought RTP would be good enough but then > potentially the phone is not sending any RTP. Can I rely > upon the phone sending RTCP? > I would like to use RTCP receiver reports for liveness > testing of the remote endpoint and hang up automatically if > I dont see any RTCP reports from one or more of the > participants of the call. Is this a sound strategy? >
Although mandatory, it is not uncommon to find SIP endpoints that do not support RTCP. I would therefore claim that relying on RYCP will not yield the desired results. Have you looked into adding session timers support to your B2BUA (RFC4028). Session timers can work even if the other end does not support them. _______________________________________________ sipx-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-dev Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-dev
