On Mon, May 01, 2006 at 12:16:18PM +0400, Jean-Michel Hiver spake thusly: > > >This is only an issue if your SIP phone has a poor/nonexistent jitter > >buffer. > > I agree with that. Asterisk should just forward any RTP immediately and > let endpoints handle the jitter buffer - unless asterisk is the endpoint > itself (e.g. with phones plugged in its fxs ports).
That makes sense if asterisk is just serving as a gateway, passing on audio to other machines, but if it's processing the audio on its own, that's not so good - it'd mess up recordings, for one. -- Jon-o Addleman - http://redowl.dyndns.org _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
