[asterisk-dev] Jitter Buffer

[Nic Bellamy]

John Lange wrote:

>On Fri, 2006-10-20 at 18:52 +0300, Zoa wrote:
>  
>
>>The thing that should be dejittered is the audio coming from asterisk 
>>going to the PSTN.
>>    
>>
>
>I suppose thats another way of looking at it. It makes the logic very
>complex though.
>
>If I understand what you are saying; putting jbenable=yes in sip.conf
>causes Asterisk to dejitter outgoing audio on the sip channel if the
>source of the audio can cause jitter?
>
>Therefore, in order to dejitter audio coming from a SIP device before it
>goes out the Zap channel I would need to put "jbenable=yes" in
>zapata.conf?
>  
>
Precisely.

There are a couple of channel flags that control (a) whether a channel 
can accept jitter - AST_CHAN_TP_WANTSJITTER, and (b) whether it creates 
it - AST_CHAN_TP_CREATESJITTER.

These flags, plus jbforce/jbenabled in the configuration, are used to 
determine whether the jitterbuffer is used, and in which directions.

Cheers,
    Nic.

-- 
Nic Bellamy,
Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/