[Asterisk-Users] Re: codec translation hints

[snacktime]

On Apr 8, 2005 12:32 PM, snacktime <snacktime at gmail.com> wrote:
> On Apr 8, 2005 12:22 PM, snacktime <snacktime at gmail.com> wrote:
> > So far it seems that the major thing affecting voice quality on my *
> > box is codec translation.   How much cpu is required to translate even
> > a single channel without getting static like sounds or other obvious
> > translation issues?  I know this probably depends on the codecs
> > involved, but are there any general guidelines to follow?
> >
> > Chris
> >
> 
> One more question.  I've been trying to figure out the best
> combination of codecs to use.    So far it seems that g729 is the low
> bandwidth codec most widely supported.  gsm seems to be supported by
> providers but not by sip devices.  g726 the opposite.  I'm thinking it
> might be worth it to just pay digium to license g729 and record all
> our own voice prompts.  Having the g729 license will enable us to
> record files in g729 format correct?
>

Ok so it looks like you can't create g729 files since it's not meant
to be an audio compression format.  So it looks like there is no way
to use a single codec on all ends if you need a low bandwidth codec,
and need to support sip devices like the sipura ata's.

Chris
 
> Chris
>