<div dir="ltr">Thanks Peter and Tony. <div><br><div>I did the dumpcap and did the packet analysis (I had the dumpcap file as well) using wireshark and figured out that audio was processed at 5600 hz. Later I figured out that the issue was caused as I was running on 250hz kernel and not 1000hz kernel. I have changed my kernel to 1000hz now. I am still now sure whether this will solve the problem or not.</div>
<div><br></div><div>Is there any way I can monitor the call quality in future in automated way? We heavily rely on nagios of all the other monitoring. If there is any way we can alert on audio call quality using that, it will be really great. I have checked the voipmonitor, but that is also not giving the values of "clock skew" and the "clock drift" which was the actual cause my latency and call quality problem respectively.</div>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Dec 18, 2013 at 1:39 PM, Peter Olsson <span dir="ltr"><<a href="mailto:peter@olssononline.se" target="_blank">peter@olssononline.se</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">I believe you already have gotten an answer about this from Tony. As stated in an earlier reply from him:<div>
<br></div><div>"<span style="font-family:arial,sans-serif;font-size:13px">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.</span></div>
<div style="font-family:arial,sans-serif;font-size:13px"><br></div><div style="font-family:arial,sans-serif;font-size:13px">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 style="font-family:arial,sans-serif;font-size:13px"><br></div><div style="font-family:arial,sans-serif;font-size:13px">The basic idea is, unless you are using the high yield cpu version of <span>ec2</span>, you can have timing issues at any time because the <span>EC2</span> will steal the cpu at will.</div>
<div style="font-family:arial,sans-serif;font-size:13px"><br></div><div style="font-family:arial,sans-serif;font-size:13px">You should probably get some pcaps and look at the deltas and timestamps on each side of the call."</div>
<div style="font-family:arial,sans-serif;font-size:13px"><br></div></div><div><div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/12/18 Ashwin Jain <span dir="ltr"><<a href="mailto:ashwinrkjain@gmail.com" target="_blank">ashwinrkjain@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">It is running on Amazon EC2.</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Dec 18, 2013 at 1:12 PM, Peter Olsson <span dir="ltr"><<a href="mailto:peter@olssononline.se" target="_blank">peter@olssononline.se</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 soft timer is causing issues, you probably have timing issues in your platform. Are you running on a VM, or is this a real machine?</div>
<div><div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/12/17 Ashwin Jain <span dir="ltr"><<a href="mailto:ashwinrkjain@gmail.com" target="_blank">ashwinrkjain@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Changing clock to soft is causing call latency issues. Is there any other way? It works well with Asterisk with no timer.</div>
<div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Dec 17, 2013 at 2:39 PM, Peter Olsson <span dir="ltr"><<a href="mailto:peter@olssononline.se" target="_blank">peter@olssononline.se</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">Well, playback without a timer won't work for you, so stick to the soft timer.</div><div>
<div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/12/17 Ashwin Jain <span dir="ltr"><<a href="mailto:ashwinrkjain@gmail.com" target="_blank">ashwinrkjain@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">It is stored locally. <div>It works fine with "soft" timer (using this we had call latency issues), but have this issue when I use "none" as rtp-timer-name.</div>
<div class="gmail_extra">
<br><br><div class="gmail_quote">On Tue, Dec 17, 2013 at 10:00 AM, Ali Pey <span dir="ltr"><<a href="mailto:alipey@gmail.com" target="_blank">alipey@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">Is the welcome message stored locally or is it somewhere on NAS?<div><br></div><div>Regards,</div><div>
Ali Pey<br>
<div class="gmail_extra"><br><br><div class="gmail_quote"><div><div>On Sun, Dec 15, 2013 at 1:42 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>We are facing issues with the bad audio quality (audio is sounding like its recorded underwater) only for the <b>playback application</b>. There is no issue with the audio call quality (after bridging). </div>
<div><br></div><div>Audio file used with playback application is recorded at 8khz with 16 bits/sample and 1 channel. <br clear="all"><div><br></div><div>Following is the pseudo dialplan:</div><div>1. Call comes on freeswitch server.</div>
<div>2. Call answered.</div><div>3. Query database to identify destination</div><div>4. Play a welcome message audio (using playback application)</div><div>5. Start call recording</div><div>6. Bridge the call</div><div><br>
</div><div>In step 4, we are facing this issue.</div><span><font color="#888888">-- <br><div dir="ltr">Thanks and Regards,<br>Ashwin Jain<div>MetroGuild</div></div>
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<br></blockquote></div><span><font color="#888888"><br><br clear="all"><div><br></div>-- <br><div dir="ltr">Thanks and Regards,<br>Ashwin Jain<br>Director (Engineering)<div>MetroGuild</div></div>
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<br></blockquote></div><span><font color="#888888"><br><br clear="all"><div><br></div>-- <br><div dir="ltr">Thanks and Regards,<br>Ashwin Jain<br>Director (Engineering)<div>MetroGuild</div></div>
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<br></blockquote></div><span><font color="#888888"><br><br clear="all"><div><br></div>-- <br><div dir="ltr">Thanks and Regards,<br>Ashwin Jain<br>Director (Engineering)<div>MetroGuild</div></div>
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<br></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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