[Asterisk-Users] one way choppy sound problem !

Steven Critchfield critch at basesys.com
Fri Jan 2 11:59:48 MST 2004


On Fri, 2004-01-02 at 12:46, 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!

Not that I know of. Sorry.
-- 
Steven Critchfield  <critch at basesys.com>




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