[Asterisk-Dev] Implementing ITU G.723.1 Annex A

[Sergio Serrano Revuelto]

Thanks

-----Mensaje original-----
De: asterisk-dev-admin at lists.digium.com
[mailto:asterisk-dev-admin at lists.digium.com]En nombre de Vinod Sankar
Enviado el: viernes, 07 de marzo de 2003 15:39
Para: asterisk-dev at lists.digium.com
Asunto: Re: [Asterisk-Dev] Implementing ITU G.723.1 Annex A


I think you can get the full source code from University de Sherbrooke,
once you have paid the royalty fees, if you are planning to use it in a
commercial application. Sipro Lab Telecom is the authorized
agent (?) for licensing G723.1 standard. You can ask them for more details.

Nathalie Beaudoin
Marketing and Licensing Manager
Sipro Lab Telecom
E-mail: nathalieb at sipro.com
Tel: 514-737-5874 x232
www.sipro.com

You can also look out the open source codec speex, which is a
royalty free implementation (with some differences) of G723.1. It is
not interoprable with G723.1 though. (www.speex.org)

Vinod Sankar

On Fri, Mar 07, 2003 at 12:58:57PM +0100, Sergio Serrano Revuelto wrote:
- Hello,
- 	I'm trying to implement the Annex A of the codec G.723.1 of the ITU. On
- having started implementing I realize that the code in the file
- codec_g723_1.c does call to fuciones that are defined of different form in
- the Annex A and in the Annex B. someone has implemented well this codec?
-
- Thanks
-
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- Asterisk-Dev at lists.digium.com
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