asterisk users - Apr 2010 - Improving audio bitrate for all callers in a conference room for a podcast

Hello,
As a podcaster I use Asterisk extensively and often have several people in
a conference room. We'll record the calls via a SIP phone connected to a
sound mixer. Is there an easy way to bump up the audio bitrate for all
callers connected to the Asterisk server and improve the general sound
quality? The server is not used much outside of recording the podcast.
We're not opposed to compiling Asterisk ourselves to get the results
we'd
like.

Any help is appreciated.
Thanks

Pat Davila

-- 
http://tllts.org/ - The Linux Link Tech Show
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Pat-
> As a podcaster I use Asterisk extensively and often have several people in
> a conference room. We'll record the calls via a SIP phone connected to
a
> sound mixer. Is there an easy way to bump up the audio bitrate for all
> callers connected to the Asterisk server and improve the general sound
> quality? The server is not used much outside of recording the podcast.
> We're not opposed to compiling Asterisk ourselves to get the results
we'd
> like.
Let me understand first:  the SIP phone doing the recording is not one of the
people on the conference?  It's in
monitor mode, for recording purposes only?

If that's the case, then you can't achieve audio quality higher than the
individual conference node channels
themselves -- sort of a 'lowest common denominator' situation.  If you
could get all nodes using a wideband codec (say
G722), and if Asterisk supports wideband mixing and recording (i.e. everything
done at 16 kHz sampling rate), then you
might be able to do it.

-Jeff