[Asterisk-Users] Re: codec translation hints

[Rich Adamson]

> 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?

If you're talking about a relatively small * system, codec translation
is not a big deal. From the * CLI, do a "show translation" to get a
rough idea what kind of consumption happens between various codecs.
 
> 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?

Depends on what you're trying to do with asterisk. If its a dedicated
corporate pbx with no interfaces to the internet, the codec selection 
is likely to be governed by what the sip phones can support.

If you're trying to use * as a soho system, its likely governed by
a combination of what your itsp will support and what phones you
are using, combined with available braodband bandwidth.

You'll find most of these items mentioned/discussed in the wiki. Take
a look.

Most popular codecs are g711, g729, and gsm (for iax). For the small
cost of a g729 license, dump five or ten into your system and add to
them later if actually needed.