[Freeswitch-users] Poor audio quality due to rtp timing issues

[Garey Arrington]

When making inbound and outbound calls I am getting slightly less than acceptable audio quality after a few seconds.  The machine is a 64 bit Centos 5.5 vm running on Xen with kernel version: 2.6.18-194.el5

I am getting warnings like with each call:

2013-03-21 04:22:54.655968 [WARNING] mod_sofia.c:1274 Asynchronous PTIME not supported, changing our end from 30 to 20
and
2013-03-21 04:22:54.056083 [WARNING] switch_time.c:578 Increasing global timer resolution to 10ms to handle interval 30


I have tried the fix that I have consistently found for this issue which is to add this to external.xml:
<param name="rtp-autofix-timing" value="false"/>

But this makes the audio quality so bad you can hardly make out anything.

Can someone point me in a better direction for a solution to this problem?
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