[Freeswitch-users] rtp-timer-name variable

Hi, 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.ht... ). 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". -- Thanks and Regards, Ashwin Jain