[Freeswitch-users] rtp-timer-name variable

[Anthony Minessale]

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 at 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".
>
> --
> Thanks and Regards,
> Ashwin Jain
>
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-- 
Anthony Minessale II

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