[asterisk-dev] Jitter Buffer

zoachien at securax.org zoachien at securax.org
Tue Oct 24 14:20:26 MST 2006


And that is where forcejb comes in :)
The phones usually have a max jitter buffer of 128ms, if you do forcejb, 
you could make this bigger.

Zoa

Pavel Jezek wrote:
>
> zoachien at securax.org wrote:
>> 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,
>
> I'm testing with ci$co 7912/sip and ci$co 7920/skinny, both phones 
> using codec g729 and are connected over cdma connection (with jitter 
> ~1000ms - mainly in upload direction i.e. phone->asterisk), asterisk 
> is in the midle, sound quality is still very poor  :'(
> maybe I have bad phones for this use (maybe with too small internal 
> jitterbuffer), but I think, that proper dejittering at asterisk for 
> both legs incomming from phones should help...
> PJ
>
>
>
>
>> 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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