[Asterisk-Dev] Help Debugging Dropped Call Audio - Possibly Fixed

[Florian Overkamp]

Kevin P. Fleming wrote:
> Florian Overkamp wrote:
> 
>> Actually, shouln't the bridge happen _after_ a jitterbuffer of sorts ? 
>> In such a case the JB would return a 'no data' packet on the read and 
>> leave it to the other end to decide what to do, or simply generate an 
>> intermediate packet to cover it up ?
> 
> 
> At this time, Asterisk does not do jitter buffering for VOIP-to-VOIP 
> bridges; it's the responsibility of the VOIP endpoints to do that.

Ah, yes - you're right ofcourse - I forgot about that in my ponderings.

Best regards,
Florian