<div dir="ltr">As I already explained. When timer-name set to none, it does blocking reads on the audio vs timed ones. If the other end has a skew in clock freq with the FS then it will timeout erroneously. Having blocking reads ensures you only read audio when there is actually some there. Basically this timer-name set to none makes FS behave just like asterisk does by default.<div>
<br></div><div>Keep in mind we do not support virtual servers beyond friendly advice because it quickly turns into time-consuming project to help people configure their virtual envs.</div><div><br></div><div>The basic idea is, unless you are using the high yield cpu version of ec2, you can have timing issues at any time because the EC2 will steal the cpu at will.</div>
<div><br></div><div>You should probably get some pcaps and look at the deltas and timestamps on each side of the call.</div><div><br></div><div><br></div><div><br></div><div><br><div><br></div><div><br></div><div><br></div>
</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Dec 2, 2013 at 2:43 PM, Ashwin Jain <span dir="ltr"><<a href="mailto:ashwinrkjain@gmail.com" target="_blank">ashwinrkjain@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Also, even after setting the rtp-timer-name to none, we are facing 1-2 sec latency (earlier it was 8-10 secs). <div>
<br></div><div>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.</div>
<div><br></div><div>Is their something else I should look for?</div><div><div class="h5"><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Dec 3, 2013 at 2:07 AM, Ashwin Jain <span dir="ltr"><<a href="mailto:ashwinrkjain@gmail.com" target="_blank">ashwinrkjain@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">If FS has stopped reading audio (I have set it to none), then how that is improving the latency issue?<div>
We have EC2 deployment of FS. We are using freeswitch to bridge the incoming call from <a href="http://voip.ms" target="_blank">voip.ms</a> and terminating it using <a href="http://les.net" target="_blank">les.net</a>.</div>
</div><div class="gmail_extra"><div><div><br><br><div class="gmail_quote">On Tue, Dec 3, 2013 at 12:53 AM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">You are basically turning off the feature that allows the remote end to stop sending audio but have FS still process the stream.<div>
The timer will allow the channel to read at the specified interval even when the other side stops sending audio.</div>
<div><br></div><div>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.</div><div><br></div><div>If you happen to have a sipura or linksys, check the wiki for specific ways to mitigate some issues.</div>
<div><br></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote"><div><div>On Mon, Dec 2, 2013 at 6:06 AM, Ashwin Jain <span dir="ltr"><<a href="mailto:ashwinrkjain@gmail.com" target="_blank">ashwinrkjain@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr">Hi,<div><br></div><div>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 (<a href="http://lists.freeswitch.org/pipermail/freeswitch-users/2011-August/075616.html" target="_blank">http://lists.freeswitch.org/pipermail/freeswitch-users/2011-August/075616.html</a>). </div>
<div><br></div><div>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".<span><font color="#888888"><br clear="all">
<div><br></div>-- <br><div dir="ltr">Thanks and Regards,<br>
Ashwin Jain<br></div>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br></div></div><div dir="ltr"><div><div>Thanks and Regards,<br>Ashwin Jain<br></div></div>Director (Engineering)<div>MetroGuild</div></div>
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</blockquote></div><br><br clear="all"><div><br></div>-- <br><div dir="ltr">Thanks and Regards,<br>Ashwin Jain<br></div>
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