On Thu, 2008-05-29 at 11:10 -0400, M. Ranganathan wrote: > 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?
How about a periodic OPTIONs request to the remote Contact in the dialog? -- Scott Lawrence tel:+1.781.229.0533;ext=162 or sip:[EMAIL PROTECTED] sipXecs project coordinator - SIPfoundry http://www.sipfoundry.org/sipXecs CTO, Voice Solutions - Bluesocket Inc. http://www.bluesocket.com/ http://www.pingtel.com/ _______________________________________________ sipx-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-dev Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-dev
