On Thu, May 29, 2008 at 12:29 PM, M. Ranganathan <[EMAIL PROTECTED]> wrote: > On Thu, May 29, 2008 at 12:15 PM, Robert Joly <[EMAIL PROTECTED]> 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? >>> >> >> 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. >> > Thanks. I guess the safest option would be to use reINVITE ( with SDP > ) as a in Dialog target refresh because not all ITSPs would > necessarily support OPTIONS.
UPDATE is another possibility I suppose but that pre-supposes support for this method. Ranga > > Ranga. > > > -- > M. Ranganathan > -- 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
