Also, even after setting the rtp-timer-name to none, we are facing 1-2 sec latency (earlier it was 8-10 secs). We also have another server running asterisk, so to check if it's origination provider or termination provider connectivity issue, we routed a call via asterisk. We didn't faced any latency issue with that. Is their something else I should look for? On Tue, Dec 3, 2013 at 2:07 AM, Ashwin Jain <ashwinrkjain@gmail.com> wrote:
If FS has stopped reading audio (I have set it to none), then how that is improving the latency issue? We have EC2 deployment of FS. We are using freeswitch to bridge the incoming call from voip.ms and terminating it using les.net.
On Tue, Dec 3, 2013 at 12:53 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
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.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
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