[asterisk-users] Asterisk 1.4 Conference with G.722

[TienSen Chong]

I haven't tried the app_conference yet. I want to know if the conference is
consisting of 3 users with G.722, does the app_conference perform
transcoding? If it is not, then app_conference will solve the issue of
having conference consists of only G.722 user since no transcoding is
needed. Is my understanding correct?

Regards,
chong


On 4/26/07, Thomas Kenyon <digium at sanguinarius.co.uk> wrote:
>
> TienSen Chong wrote:
> > Hi all,
> >
> > I am having problem with conference call (meetme feature) using G.722
> > phone. G.722 phone to phone is working fine. I suspect this is due to
> > the fact that Asterisk 1.4 only support G.722 passthrough.
> >
> This will be the case, Meetme transcodes the audio (to slin iirc), where
> it mixes it.
>
> > Any ideas how this problem can be fixed.
> >
> Have you tried using app_conference?
> To be honest, I don't know how you would be able to have more than 2
> people in a call without some transcoding going on.
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