[Freeswitch-users] rtp-timer-name variable

[Ashwin Jain]

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.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".

-- 
Thanks and Regards,
Ashwin Jain
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