[asterisk-dev] Jitter Buffer

[zoachien at securax.org]

We should not dejitter when possible, we should dejitter when we need to 
because the receiving endpoint cannot do dejittering.
Sip to iax should not dejitter, as the endpoints should dejitter, the 
reason for this is that if we do dejittering in the middle, and another 
time on the endpoints, the total delay because of dejittering will be 
bigger than needed.

A configuration file for applications might make sense, for example if 
you do recordings because of line quality checks, you don't want the 
jitter buffer to be active, as you want the real thing with the gaps 
etc, but if you want recordings for monitoring people, you will want it 
because it will be nicer to listen to.

Zoa


Pavel Jezek wrote:
> Zoa, I think, that we should dejjitter when possible, always _before_ 
> any processing within asterisk (sending to some app, bridging to 
> channel, etc.),
> is there any reason, why (in your example) dejjitter only sip-zap and 
> not sip-iax?
> PJ
>
>
>
> zoachien at securax.org wrote:
>>
>>> 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.
>>>   
>>
>> I disagree with this, it's the only way you can choose when to have 
>> dejittering and when not.
>> You might for example want it for sip to zap but not from sip to iax, 
>> thus if you set it in sip, you will have dejittering for sip in your 
>> suggestion.
>>
>> Zoa
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