On Tuesday 24 October 2006 11:08, John Lange wrote: > jbenable=yes in sip.conf should tell Asterisk to dejitter incoming audio > FROM SIP channels when the receiving leg can not handle dejitter on its > own.
> One simple example demonstrates why this makes sense; in the case where > the sip channel is talking to an application inside Asterisk or > otherwise connecting to a something which doesn't have jbenable option > there is no way to activate the JB and therefore audio is jittered. > > Specifically, callers trying to record their voicemail greetings will > have jittered audio. meetme and recording apps also need this. > Setting jbenable=yes in zaptel.conf in order to dejitter sip audio is > very confusing and I know there will be a _lot_ of people wondering > about this besides me. Unless we're planning on adding jbenable-type options to voicemail, meetme, monitor/mixmonitor and local channels (as a starting point), it would seem to make a LOT more sense to keep jbenable in all voip protocol configurations... (i.e. sip, iax2, skinny, etc.) -- that's where the jitter buffer actually exists. -A. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
