[Asterisk-Dev] 16 KHz audio ?

Strom Carlson stromcarlson at gmail.com
Fri Dec 17 04:28:52 MST 2004


On Fri, 17 Dec 2004 11:49:12 +0100, Eric Bart <btk-adm at byortek.com> wrote:
> I've seen nowhere a plan to improve audio quality.
> 
> It seems that the Skype success is partly due to its
> 16 KHz audio bandwidth. It gives users the feeling that
> the far party is in the same room.

I guess you could conceivably design an IAX type protocol that
operates with greater than 8khz audio, but the instant you hit
anything that uses traditional time-division multiplexing, you're
going to have to downsample that to standard 8khz mu-law or a-law
audio.  Since the vast majority of the frequency range of human speech
is below 4000 hertz, the telephone network has _always_ been designed
with a frequency response of  less than 4000 hertz.  You can ramp up
the bandwidth all you like, but I wonder at what point the telephone
set itself will hit its maximum quality.

The nice thing about mu-law companding, of course, is that it manages
to give the impression of 13 or 14 bit audio using only 8 bits of
resolution.

-- 
Strom Carlson
http://www.stromcarlson.com



More information about the asterisk-dev mailing list