[Asterisk-Dev] new jitterbuffer in 1.2?

[Steve Kann]

Mike Taht wrote:

>I've been testing it on calls from the US to our china office, where I
>typically get 10% packet loss or worse. Works pretty good during US
>off-hours, not so well during China business hours (where I bet packet
>loss is much more bursty). It works a heck of a lot better than the
>previous code.... this is on ulaw.
>
>I tried to get speex to work for the first time connecting these two,
>and I got nothing but noise. That doesn't mean anything by itself, I'd
>never tried speex before on anything.
>  
>

I think that presently, if you're call is coming in via IAX, and being 
terminated to a zap channel (for example), then PLC won't be applied, 
because the ulaw<->pcm translator is not being used.. That's something 
that, I suppose, would need to be added somewhere (maybe to chan_zap?).

Not sure what the speex issue is, but you can try GSM, as in that case, 
you'll use GSM<->PCM<->ulaw conversions, and PLC will be applied in the 
first translation..

-SteveK



>
>On Wed, 23 Feb 2005 07:18:59 -0500, Andrew Kohlsmith
><akohlsmith-asterisk at benshaw.com> wrote:
>  
>
>>On February 23, 2005 12:49 am, rsenykoff at harrislogic.com wrote:
>>    
>>
>>>Please please please give us the wonderful jitterbuffer and Packet Loss
>>>Concealment.
>>>      
>>>
>>Are you testing it?  The only way it's gonna get in is if it's been well
>>tested.  Jerjer (Nufone) even has a test server for terminating calls to PSTN
>>with it if you want to use it (you'll need an account with him).
>>
>>Also, Please please please turn off HTML when posting to the mailing lists.
>>Save everyone some bandwidth!
>>
>>-A.
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