Hi, We're building a call app based on SEMS and started out building on top of AmB2AB logic. So the first call leg (caller) is being setup and after playing an announcement the second call leg (callee) is set up. The audio is relayed in this case. As an optimization we've added some business rules that decides whether the the caller and callee trunks will receive a reinvite with each other's SDP so - after a successful re-invite - the audio will no longer be relayed by SEMS but send between the trunks directly. We've got this all working. There's just one problem we're facing. For a brief moment when both caller en callee sessions are setup they're both using a MediaSession to relay audio (announcement, early media, etc.). After a successful re-invite we will call stopMediaProcessing() in both caller and callee session to remove this MediaSession. This seems to work for the callee session, but it does not for the caller session. When the call hits the "dead_rtp_timeout" limit an onRtpTimeout callback is made and the caller session is disconnected by SEMS. The MediaSession of the caller leg is - for some reason - still active and monitoring the RTP stream. Is there anybody that knows how we can properly remove the MediaSession from the caller session? Or maybe keep the MediaSession alive, but prevent the auto-disconnect when the dead_rtp_timeout kicks in? We've already thought of the obvious solution to set the dead_rtp_timeout configuration parameter to 0, but this way we would lose this functionality in general. But if no "real" solution is found we might end up using that. Any help is appreciated! Best regards, Tom van der Geer
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