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?
Thanks!
hose
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
Hose wrote:> 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?There was a flaw in Asterisk's G.722 transcoder module that was fixed recently (on May 15, 2009), so any release made after that date should solve your problem. Upgrading to 1.6.0.10 should give you the fix (and the fix should be noted in the ChangeLog for 1.6.0.10 as well). -- 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