> When we enabled jitterbuffer the sound quality seemed to improve but we > noticed some problems: > > (a) sometime we would get only one-way audio; > (b) other times we would experience no audio in one direction for between 1 > and 4 seconds and then things would seem to work fine; > (c) some times users reported a "clipped" and almost "half duplex" sound > quality as the flow of the conversation shifted back and forth. > > We also noticed some wingnut values for Lag and Jitter such as: > Lag: -65476ms > Jitter: 12897799ms > > PSTN gateway is "CVS-04/20/04-01:11:29 " > Client machine is "CVS-HEAD-06/02/04-07:56:41" > > Searching the Asterisk bug lists shows some significant fixes (1696, 1643). > > Q2: Is jitterbuf working well enough to try again? > > Q3: Any other suggestions for improving voice quality with IAX links?
A google search of the asterisk-cvs list indicates there has been several iax changes in the last several months. Iax2 with gsm is working very well between * systems using the current cvs Head. I was told specifically by Mark to include jitterbuffer=no in the iax.conf, but with no explanation as to why. Although I'm not a programmer, causual browsing of the source code would seem to suggest that some sort of dynamic jitter buffer function is in use and attempts to over-ride it might not be a reasonable thing to do. I'd suggest bumping both systems up to current cvs Head, add the statement, and eval the result. _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
