[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
>
>
>


I found this link:
http://www.voip-info.org/wiki/view/Asterisk+codecs

So every client that connects to the conference would have to be
configured to use whatever codec we wind up using.