[Asterisk-Dev] new sip jitterbuffer?

[Steve Kann]

Roy Sigurd Karlsbakk wrote:

>>> btw
>>>
>>> if having the following setup, will the IAX2 jitterbuffer have any 
>>> effect on jitter introduced on the SIP connection?
>>>
>>> SIP ATA (internet) Asterisk1 chan_sip <-> Asterisk1 chan_iax2 (gigE) 
>>> Asterisk2 chan_iax2 <-> chan_zap
>>>
>>>
>>
>> Assuming that you're using 1.2 or CVS head and default settings, yes.
>
>
> This is 1.0.x, but I guess this works similarly on 1.0 as with 1.2?
>
>> The jitterbuffer(s) would be totally disabled on asterisk1 during the 
>> bridge, and the IAX2 JB on asterisk2 would see and compensate for all 
>> jitter introduced on packets flowing to the right, and the SIP ATA, 
>> in theory, would see all jitter introducted on packets flowing left; 
>> assuming it too, has a JB, it could compensate.
>
>
> nice. ATA has JB, yes...

The paragraph above doesn't apply to version 1.0; 1.0 doesn't 
automatically disable the jitterbuffer when VoIP calls are bridged like 
1.2 does. Also, the 1.0 jitterbuffer doesn't detect lost packets or call 
for PLC, and it effectively does silenct substitution when growing.

-SteveK