[Asterisk-Users] G.723.1 codec

Andrei Koulik
Sat, 17 Jan 2004 03:49:51 -0000

I solve it for h323 in follow way:
1. Exclude all codecs except g723.1 from h323.conf:

   disallow=ULAW
   allow=g723.1

2. Add format_g723 module (http://www.agk.nnov.ru/format_g723.c.gz)
   into project

3. convert all wav and gsm sound into g723 format (use lbccodec from
    g723_1 demo package, don't ask me where you can download it)

4. set maxsilence=0 in voicemail.conf to suppress conversion into pcm
    format for silence detection.

And it works fine for me.
But where are some bugs in h323 module:
* not supported g7231 without sound detection (simple to fix).
* sometime data transfer (rtp traffic) begins before negotiation
  complete and first packet is going in g711 codec and channel going
  down (not yet reviewed).

if will any question regards format_g723 module send mail to:
f723 >< agk.nnov.ru



Friday, January 16, 2004, 10:30:41 PM, Dan Tusa wrote:

DT> Hi,

DT> Want to do some experiments with the G.723 codecs - where can I download
the
DT> 723 source code for Asterisk?

DT> I know there are some ongoing discussion regarding patents and license
fees
DT> for the g.723 but I have some hardware on which I only have the 723 and
need
DT> to test it privately.

DT> Thanks!
DT> Dan

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--
Andrei Koulik.
System administrator, Sandy Info Ltd. (ISP), Nizhny Novgorod, Russia