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? Ranga -- M. Ranganathan _______________________________________________ sipx-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-dev Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-dev
