<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Hi afshin,<div><br></div><div>I changed to the linux-image-rt (real time preemption patch) and so far the calls have been good. A bulk of the calls will happen tomorrow so I will have more of and idea if it worked. Looking at it now its set to 1000HZ so I hope this solves the issue.</div><div><br></div><div>Ash.</div><div><br></div><div><br></div><div><br><div><div>On 08/08/2010, at 5:36 AM, afshin afzali wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">Hi Ash,<br><br>Did you change your kernel to -preempt (10.04 Lucid Lynx) ?<br><br>-- afshin<br><br><div class="gmail_quote">On Fri, Aug 6, 2010 at 7:27 PM, Ash <span dir="ltr"><<a href="mailto:ash@archerdrive.com">ash@archerdrive.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">I think I may have solved my own problem... I am running on Ubuntu (I know its not recommended) and my kernel timer was set to 100hz. Running time_test 1000 1000 it would avg from 997 to 1300 when I kept running it. I have now set the kernel to be 1000hz and it now avg around 1008-1014 on multiple tests.<br>
<br>
Unfortunately its not one of those always happens issues so I will have to see how it goes.<br>
<div><div></div><div class="h5"><br>
<br>
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<br>
On 06/08/2010, at 6:38 PM, Ash wrote:<br>
<br>
> Hi All,<br>
><br>
> I am running freeswitch 1.0.6 and some users have reported that they will be on a call and the audio will go silent for 3 seconds. This doesn't appear to be happening to all users. So far, I have done the following:<br>
><br>
> - Internal and External profiles set <param name="rtp-autofix-timing" value="false" /> ( I was getting the RTP warning about ptime being 30 and all phones are set to 20ms)<br>
> - Set the phone to use PCMA and made sure its PCMA through to the provider<br>
> - We have installed a SIP Proxy at the other location so the voip phones all go via the SIP Proxy as we thought it might be a NAT issue.<br>
> - Disabled VAD and CNG on the phone<br>
> - The remote phones are connected via a DSL service but this appears to be rock solid and averages response times between 10ms and 40ms.<br>
><br>
> I thought it might be something with VAD, is VAD disabled by default or should I add to the internal and external profiles<br>
> <param name="vad" value="none"/ ><br>
><br>
><br>
> Has anyone else come across something like this before? Maybe some other ways I can investigate what is going wrong?<br>
><br>
> Any help would be much appreciated.<br>
><br>
> Ash.<br>
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