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

[Patrick Davila]

>> 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
>

Jeff,
So the first thing to improve audio quality is to switch over to a higher
quality codec like G722. What are the other higher quality codecs we can
use? Everyone connecting should make sure they're using the higher quality
codec? Is there any way to configure a stock Asterisk install to use
wideband mixing or will we have to compile our own?
Thanks again

Pat