Hello Daniel ,

I'm Also Having the Same doubt on g729 Codec,

I'm  using RTPproxy with Nathelper ,
OpenSER and RTP proxy does  media signaling when the Call is Established,

My main question is
Is RTP proxy support the G729, with OpenSER,

With out using the Transcoder ( Asterisk ) How can OpenSER signals the G729 Codec.

On 2/15/07, Daniel-Constantin Mierla <daniel@voice-system.ro> wrote:
Hello,

you need a transcoder in the middle. OpenSER does only signaling, so it
is not able to transcode. Asterisk, for example, does.

Cheers,
Daniel

On 02/15/07 10:57, tusker keg wrote:
> Howdy
>
> I have a situation  I hope you guys will help me out with
>
> I am receiving  call from a VOIP  peer  (SIP Call) and the peer can
> only send them as G711.
> I need to redirect to call to another voip peer over the wan and due
> to bandwidth considerations I need to translate the codec to g729.
>
> Any ideas on how to do this,
>
> Sample config file will be help
>
> Regards
>
> ./Tusker
>
> _______________________________________________
> Users mailing list
> Users@openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users
>

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