[asterisk-users] Improving audio bitrate for all callers in aconference room for a podcast

[Jeff Brower]

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