[Freeswitch-users] Solved: Jitter problems
Ivan C Myrvold
ivan at myrvold.org
Mon May 7 09:09:06 PDT 2007
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.
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-
> 2. http://www.myrvold.org/freeswitch/ip242me.jpg , from ip24-
> 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/
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> Freeswitch-users at lists.freeswitch.org
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