[asterisk-users] g.722 + loudness

[Steve Underwood]

Hose wrote:
> Hi,                                                                                                                                                                   
>                                                                                                                                                                       
> We've been running g.722 in asterisk 1.6.09 for awhile now, with a PRI                                                                                                
> and numerous sip clients.  Internal sip to sip and sip to pri (and
> vice versa) work fine between g.722 and ulaw - the transcoding is                                                                   
> acceptable.                                      
>                                                            
> The only time it fails is when we utilize a meetme conference bridge.
> With a Polycom IP 6000 + a call over the PRI, the person calling in over                  
> the PRI sounds distorted when they're barely talking at a normal volume.       
> Anything over a normal volume results in terrible clipping.  Bringing                                                                                                 
> the volume down on the Polycom either via software settings or the                                                                                                    
> actual volume keys doesn't stop the distortion, so that points to a                                                                                                   
> problem with asterisk (the volume can be very loud, barely audible, but
> you can still hear the clipping occuring).  By clipping, I mean the
> static that happens when you have a signal that's too loud.
>
> The thing is, when you call directly into the Polycom over the PRI, it's
> fine.  This ONLY happens during a conference call with g.722, though
> this might be because asterisk is negotiating a ulaw connection when
> called direct from the PRI - is there a way to check what codec it's
> negotiated during the call?
>
> I have a feeling that the issue is between transcoding of ulaw to g.722
> and it's too loud during the transcoding - anyway to adjust the levels?
>   
I'm not sure in which version of Asterisk it was fixed, but there was a 
6dB gain error in the G.722 codec until fairly recently. You are 
probably hitting that problem.

Steve