I was facing lot of audio quality issues and turning off sip jitter buffer didn't solved anything. I set rtp-timer-name = none (it was set to soft earlier) and it seemed to solved the problem (http://lists.freeswitch.org/pipermail/freeswitch-users/2011-August/075616.html). 

Can someone explain what was the issue and how it got fixed? I wasn't able to find any documentation related to "rtp-timer-name".
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