[asterisk-dev] Jitter Buffer

[Andrew Kohlsmith]

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.