[asterisk-users] Improving audio bitrate for all callers in a conference 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
>>
>
> 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?

Another possibility might be G711.1.

> Everyone connecting should make sure they're using the higher quality
> codec?

Yes.  If a few don't and a few do then you would have a couple of issues:

  -transcoding has to take place prior to mixing, so
   whatever SIP software you're using has to correctly
   handle negotiation and call setup and one of the
   software components in your setup has to do actual
   transcoding work on RTP (voice data) packets.  Are
   you using Asterisk for the conferencing function?
   Or only recording?  If the latter then who/what
   does the conferencing (mixing) ?

  -if you were to use Asterisk for transcoding, I'm
   not sure how Asterisk would handle that.  It
   could downsample the wideband nodes, then you get
   no audio quality improvement , or it could upsample
   the G711 (or other nodes) and your recording would
   sound better when the wideband nodes are talking

> Is there any way to configure a stock Asterisk install to use
> wideband mixing or will we have to compile our own?

Not sure!

-Jeff