[Asterisk-Dev] High-Bandwidth codecs (again) G.722.1

[asterisk at ntplx.net]

I won't requote the whole email....but I have G.722 running now supporting
Grandstream BT phones. I don't have a codec translator working yet so G.722
only works between phones. G.711 is still used for other calls.

Note this is G.722, not G.722.1 or G.722.2. I'll look at the polycom stuff
for G.722.1 it seems like it might be a good solution. The grandstream is
not a very good phone so I can't hear much improvement on G.722 vs. G.711.

The simple answer for Asterisk on high sample rate (ie 16Khz) audio
is don't worry about it....G.722 (original) was designed for 64K ISDN
so in it's natural form we pass it off just like G.711 64K...no problem.
I have G.722 file format working fine too because it uses the same
setup as G.711. I have not looked at making lower bandwith (56K, 48K, etc)
verions working as I don't have a better test phone.

As for what do to with 16k audio....downsample/upsample. If someone
is not using the native G.722 codec then who cares if the codec translates
down to a 8Khz format. At least for now it would work, 16Khz SLin audio
can be added later (much later) if ever needed. It would only buy you
higher quality voicemail/MOH/meetme...not very important at this time.

If two phones use G.722 at 64K then asterisk does not have to do anything
special to pass the audio data.

  Andrew


Quoting John Todd <jtodd at loligo.com>:

>
> There has been discussion here about the complexities of adding
> high-bandwidth codecs to Asterisk which use Asterisk's ability to
> insert/examine/redirect media flows.