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

[Kevin P. Fleming]

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.