[Asterisk-Users] Sipura g729 call quality to PSTN

Pedro traci.asterisk at gmail.com
Fri Feb 18 15:10:20 MST 2005


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 <radamson at routers.com> 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
> 
>



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