[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
>>>> _______________________________________________
>>>> --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
>>> _______________________________________________
>>> --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
>>
>> _______________________________________________
>> --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
> _______________________________________________
> --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
More information about the asterisk-dev
mailing list