[asterisk-dev] g722 question

Steve Underwood steveu at coppice.org
Mon Dec 7 08:52:15 CST 2009

On 12/07/2009 10:13 PM, Saúl Ibarra wrote:
> 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.
I think most of these projects are using my implementation of G.722. 
Early versions had this 14<->16 bit issue. The current one doesn't. 
People with the problem really ought to upgrade. The right thing is for 
the G.722 clipping point to be 32767/-32768.


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