Rich - thanks! Glad I am not the only one seeing this :) Would be very interested in your results. No problems that I see yet with these settings.
On Fri, 18 Feb 2005 10:21:08 -0600, Rich Adamson <[EMAIL PROTECTED]> wrote: > > > That does not sound right at all. The difference between the two Time= > > > values should have been 10 (milliseconds). > > > > > > Did you reboot the Sipura after making the change? There are some values > > > in the Sipura that don't take effect until after the next reboot; I don't > > > have a clue whether this happens to be one of them. > > > > Yes - sipura was rebooted. Actually, the changes did seem to take > > affect even before the reboot (verified by call quality improvement > > and ethereal traces). > > > > So in your opinion, instead of 80, it should be a difference of 10? > > If so - then you are saying that the timestamp is in miliseconds? > > > > I am as puzzled as you - really does not seem logical, but call > > quality is finally decent and it does not seem to bother asterisk at > > all. Do you see any potential problems with this? > > I did a fair amount of experimenting this morning using a spa3000 with > g711 and g729 codecs. I'm more confused now then ever. I also used > ethereal to inspect timestamps, etc. > > spa3k(fxs) -> asterisk -> IAX(ITSP) -> pstn net -> analog phone > > The spa3k is running v2.0.13(GWg) firmware, and * is CVS-HEAD-02/13/05. > > The spa3k had a default RTP Packet Size of .030 (30 milliseconds) even > though the User Manual indicated that 20 milliseconds is the default. > Asterisk config is default at 20 milliseconds. > > I changed the spa3k rtp from .030 seconds, to .020 seconds for > consistency. Audio quality "seemed" to be better when using g711. > > Regardless of whether I used g711u or g729, the rtp timestamps were > always 160 difference between consequtive packets (as observed by > ethereal). > > Changing the spa3k rtp to .010 seconds yielded timestamps that were > always 80 difference between consequtive packets (same as you > observed). However, * -> spa3k continued to have 160 difference. > Audio quality seemed to improve another step, and the occasional > echo that we heard seemed to disappear. Pure guess is the smaller > rtp size is impacting the jitter buffer and/or echo canceller in > the spa3k. I'm going to run with these settings for a while to see > what the longer term impact/stability might be. > > Rich > > _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users