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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