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.
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
