On Sun, 29 Aug 2004, Andrew Kohlsmith wrote:
> Also, is are logs of problem conversations already in progress any use to you? > You nailed down the "dead audio after 65535ms" problem but every now and > again (very very rare) we will have a conversation where the incoming audio > goes totally dead for about 2-4 seconds and then continues just fine. This > occurs usually several minutes into the conversation, and I've never seen it > occur twice in a conversation. Logs of parts of a call are fine. The jitter buffer makes all its decisions about dejittering based on the timestamps of incoming frames. There a fundamental expectation that the sending side is correctly stamping each frame - 20msec, 40msec etc etc. The problem is that the sending side doesn't always do that. Sometimes for one reason or another the stamps "jump". The receiver has no way of telling that the sender mangled the timestamps, and assumes that the packets with the new stamps have been delayed, or arrived early, or whatever. Either way, the jitter buffer does its thing and unknowingly makes things worse. Unfortunately, this is why you can still be better off without it - but the problem really needs to be fixed by fixing the timestamp generation on the sender. Steve _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users