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

[tim panton]

On 24 Dec 2005, at 17:50, Kevin P. Fleming wrote:

> 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.

So the OP's symptoms could be just a few out of order  packets, his  
SIP endpoints have
JB's so mask or fix the problems making them inaudible.
But the Monitor() application on the other hand will seek ahead,  
leaving holes (or worse) in the audio file.

Does this sound right?

T.

http://www.westhawk.co.uk/

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