[Asterisk-Users] one way choppy sound problem !

[Andres]

On Friday 02 January 2004 13:46, Nicolas Gudino wrote:
> Hi Steven,
>
> On Fri, 2004-01-02 at 14:55, Steven Critchfield wrote:
> > What is the ping times between your 2 asterisk servers? In the archive I
> > have documented before that IAX jitter buffer sometimes has problems on
> > short ping time links. At the time we where on a private T1 with 4ms
> > ping times. We re enabled our jitter buffer now that we are on a DSL
> > connection and our ping time is between 56 and 70 ms.
>
> The ping time is about 35 ms, one server is on ADSL and the other a T1.
> I tried with different jitter buffer settings, but I really don't know
> how to tune them.  I also tried disabling jitter buffers. I even tried
> using a sip call directly, without using IAX2 (so no jitter buffers
> apply, at least no iax jitter buffers), always with the same result:
> choppy sound from sip to pstn and perfect sound from pstn to sip. Using
> alaw or ulaw the choppiness is tolerable, with other codecs is prety
> bad. Are there any documents on how to tune jitter buffers? Thanks!

Are your "rxgain"  and "txgain" values different than zero in zapata.conf?  If 
so then repeat your calls setting them to "0" and see if it helps.