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

[Kevin P. Fleming]

SteveK wrote:

> On the other hand, in the case of Monitor()'ed calls, for ideal call  
> recording quality, you'd want a jitterbuffer somewhere between the  
> packets being received and being written to disk, but asterisk  doesn't 
> have that.   You could do that by enabling the jb for VoIP- >VoIP calls 
> when Monitor() is active, which would add latency to  these calls, or 
> some other way, which would require some more code to  implement.

Using the new MixMonitor infrastructure, it would be relatively easy to 
put a jitterbuffer in between the frames being copied from the channels 
and them being mixed/written.