[Asterisk-Dev] Digium's G.729A codec problem

[Jacky]

Hi, Daniel,

I also try your code based Intel's IPP, The console show many Notice
information like,
  "Dropping extra frame of G.729 since we already have a VAD frame at the end"
and I find out from maillist discuss about this, so I add SDP info to
force SIP UA to use G.729A, but still the same result as Digium's
G.729A.

Sometime can hear voice each other, but rest time almost no voice when
SIP UA use VAD feature.

Many SIP UA can't disable VAD, ex. Grandstream, MOSA 3704 SIP gateway,
So I still get "Drop Notice informaion".

If I wish let the G.729 codec module is compatible with G.729B VAD,
what do I  must to modify Asterisk code, or your implementation code?

Best regards,






On Wed, 02 Mar 2005 11:39:19 +0000, Daniel Pocock
<daniel at readytechnology.co.uk> wrote:
> 
> The Digium implementation is closed source :( unlike the rest of
> Asterisk, so you probably won't be able to troubleshoot this yourself.
> 
> Try the open source implementation and let me know if you have the same
> problem:
> 
>    http://www.readytechnology.co.uk/open/g729
> 
> Make sure you are watching the Asterisk console when you try making
> calls, my code spits out error messages if packets are the wrong size,
> etc.  This will give you some clues.
> 
> Regards,
> 
> Daniel
> 
> Jacky wrote:
> 
> >Hi, all,
> >
> >I have buy 5 Digium's G.729A codec(it just support G.729A license)
> >When I  calll with 2 SIP UA that support G.729A and G.729B, its rtp frame
> >have some problem when softswitch with Asterisk.
> >
> >The voice frame have been drop, so sometime I can't hear voice.
> >
> >If I want to fix the problem when softswitch G.729A and G.729B codec.
> >What source code I must to modify ?
> >Or some people have finished the issue, Could you show me how to do?
> >
> >
> >
> >
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-- 
Jacky