asterisk users - Dec 2009 - g722 question

Hello,

I am working with several SIP projects that use g722, or are trying to
do so, with pjsip library.

According to pjsip team's interpretation of g722, it works with 14bits
PCM for input/output, so pjsip basically 'converts'  the audio sample
from 16 bits to 14 when encoding and vice-versa. Some implementations
don't do 16<->14 bits conversion, so when pjmedia talks to one of
those the over-driven audio problems appear.

What we need to know is what's the most used implementation: 14<->16
bits conversion or not.

Any pointers to help clear this up? We'd really like to see more
g722-capable SIP clients for our own conference on ZipDX.

Regards,

Randy
http://vuc.me

Randy R wrote:
> I am working with several SIP projects that use g722, or are trying to
> do so, with pjsip library.
> 
> According to pjsip team's interpretation of g722, it works with 14bits
> PCM for input/output, so pjsip basically 'converts'  the audio
sample
> from 16 bits to 14 when encoding and vice-versa. Some implementations
> don't do 16<->14 bits conversion, so when pjmedia talks to one of
> those the over-driven audio problems appear.
> 
> What we need to know is what's the most used implementation:
14<->16
> bits conversion or not.
> 
> Any pointers to help clear this up? We'd really like to see more
> g722-capable SIP clients for our own conference on ZipDX.
As far as I am aware, for ITU-T compliance the codec only cares about 14
significant bits, but the reference source code needs those 14 bits in
the *top* 14 bits of each 16-bit word that it supplies/produces. The
Asterisk implementation does not do any bit-shifting or masking at all,
and seems to interoperate with quite a few endpoints just fine, so
presumably that means it's the correct implementation :-)

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpfleming at digium.com
Check us out at www.digium.com & www.asterisk.org