[Freeswitch-users] Solved: Jitter problems

[Vazir]

I don't think this value can be ignored, it's just adopted 
to that value. G729 should be 20 by standard... What codec 
did you use?

On 7 May 2007 21:09, Ivan C Myrvold wrote:
> It turned out that SIPURA SPA-2000 has a bug (which also
> affects PAP2). But easily fixed:
>
> Use your browser to open the SIP configuration on the
> SIPURA SPA-2000. Change "RTP Packet Size" from 0.030 to
> the new value 0.020
>
> Now I have perfect audio both ways with Freeswitch.
>
> Why didn't I have the problem with Asterisk? Apparently
> Asterisk ignores the value set by the SIPURA.
>
> Ivan
>
> Den 6. mai. 2007 kl. 13:11 skrev Ivan C Myrvold:
> > I am trying to get Freeswitch to work with my SIP
> > provider, IP24, but I always get very bad sound in one
> > direction.
> > I made a wireshark trace, and looked at the jitter
> > graph it produced.
> >
> > I made an incoming call from my mobile phone to
> > Freeswitch via SIP provider IP24, PCMA codec:
> >
> > 1. http://www.myrvold.org/freeswitch/me2ip24.jpg , from
> > freeswitch-
> >
> > >ip24
> >
> > 2. http://www.myrvold.org/freeswitch/ip242me.jpg , from
> > ip24-
> >
> > >freeswitch
> >
> > There is a big difference in jitter between the two
> > graphs, and the speech quality is very bad
> > ip24->freeswitch (2).
> >
> > What can I do to fix the speech quality?
> > Asterisk and OpenPBX (CallWeaver) have both excellent
> > audio quality in calls through my SIP provider IP24.
> >
> >
> > Here is a voip graph of the SIP call:
> > http://www.myrvold.org/ freeswitch/graph.jpg
> >
> > Ivan
> >
> >
> >
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