[asterisk-dev] g722 question

[Saúl Ibarra]

Hi,

I'm forwarding this mail from the VUC regarding g722. As Asterisk a
largely used VoIP platform ;) we also need to know how Asterisk
handles this:

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.


-- 
/Saúl
http://saghul.net | http://sipdoc.net