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 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
