You are basically turning off the feature that allows the remote end to stop sending audio but have FS still process the stream.

The timer will allow the channel to read at the specified interval even when the other side stops sending audio.

If the other end has a clock sync issue with FS or FS does not have a reliable timing source, it could cause your problem.

If you happen to have a sipura or linksys, check the wiki for specific ways to mitigate some issues.

On Mon, Dec 2, 2013 at 6:06 AM, Ashwin Jain <ashwinrkjain@gmail.com> wrote:

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".

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