[Asterisk-Dev] How to trigger PLC in a codec in SIP channels

[Steve Kann]

Gouri Johannsen wrote:

> At 07:00 PM 3/31/2005, you wrote:
>
>> I
>> On Mar 31, 2005, at 7:48 PM, Gouri Johannsen wrote:
>>
>>> I know that Jitter buffer and PLC implementation is not yet done in 
>>> the SIP/RTP channels. But I was wondering is there an easy way to 
>>> indicate to the codec that a frame did not arrive on time with the 
>>> current implementation (i.e. without JB?)
>>
>>
>> If you don't have a jitterbuffer, how can you know when a frame 
>> didn't arrive on time?
>
>
> Some entity that keeps time should figure out that if a frame arrived 
> every 20ms or not. When a frame doesn't arrive it can set a flag to 
> say that no frame arrived and call the translator. You can call this a 
> rudimentary JB, if you will. Currently looks like * simply doesn't do 
> anything when a frame doesn't arrive. Still there is the concept of 
> timing and expectation of a frame every 20ms, right?


Nope. Frames come in to the channel driver, the channel driver sends 
them up to the core. Nothing expects frames in any particular amount of 
time. (and 20ms isn't always the proper interval anyway).

> This triggering of no frame can also be used for better handling of 
> silent frames.

Could be, but it's probably not the best solution to this problem. The 
best solution would be for apps to use "ast_generator" stuff instead of 
always expecting incoming frames to be present.



>>> Along the same lines, anybody has an idea when the new jitterbuffer 
>>> implementation will be ready for SIP?
>>
>>
>> There's a preliminary patch in mantis now.
>
>
> Thanks, I will check it out.
>
> Gouri.
>
>
>> -SteveK
>>
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