<div dir="ltr"><div><div>Regarding my comment about not finding what has changed, I didn't mean to imply it was something in Freeswitch. <br><br></div>I am actually fairly certain it is not FS since it was working fine for weeks on the same version.<br>
<br></div><br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On 25 April 2013 13:15, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</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"><br><div class="gmail_extra"><br><br><div class="gmail_quote"><div class="im">On Wed, Apr 24, 2013 at 6:05 PM, Philippe Le Toquin <span dir="ltr"><<a href="mailto:philippe@ppmt.org" target="_blank">philippe@ppmt.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div bgcolor="#CCCCCC" text="#000000">
OK I installed pcapsipdump but it is killing my poor Guruplug.<br>
<br>
CPU is at 99% so the call are going through but there is a big
delay. So it is not going to help!<br></div></blockquote></div><div>Bummer! <br></div><div class="im"><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div bgcolor="#CCCCCC" text="#000000">
<br>
What bugs is that up to 2/3 weeks ago everything was fine and
suddenly the cutout started.<br>
I can't find out what can have change that would cause this.<br></div></blockquote></div><div>This is why I mentioned in another thread that I would like to get people more familiar with git bisect. You can manually use git bisect to find out which commit broke things for you. Check out Mitch Capper's handy wiki entry:<br>
<a href="http://wiki.freeswitch.org/wiki/Git_Tips#Git_Bisect_-_Tracking_down_breaks_and_bugs_extremely_fast_with_git" target="_blank">http://wiki.freeswitch.org/wiki/Git_Tips#Git_Bisect_-_Tracking_down_breaks_and_bugs_extremely_fast_with_git</a><span class="HOEnZb"><font color="#888888"><br>
<br></font></span></div><span class="HOEnZb"><font color="#888888"><div>-MC <br></div></font></span><div><div class="h5"><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div bgcolor="#CCCCCC" text="#000000">
<br>
/Philippe<br>
<br>
<div>On 13-04-23 11:41 PM, Michael Collins
wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div>
<div>
<div>pcapsipdump ftw!<br>
<br>
</div>
Check it out:<br>
<a href="http://wiki.freeswitch.org/wiki/Packet_Capture#pcapsipdump" target="_blank">http://wiki.freeswitch.org/wiki/Packet_Capture#pcapsipdump</a><br>
<br>
</div>
It gets pcaps for you and separates each call into its own
file with both signaling and media. Find the pcap with a bad
phone call and open that one up in Wireshark for analysis.<br>
<br>
</div>
-MC<br>
</div>
<div class="gmail_extra">
<br>
<br>
<div class="gmail_quote">On Tue, Apr 23, 2013 at 6:30 PM,
Philippe Le Toquin <span dir="ltr"><<a href="mailto:philippe@ppmt.org" target="_blank">philippe@ppmt.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div bgcolor="#CCCCCC" text="#000000"> Hi Sean and Michel<br>
<br>
Thanks for your input.<br>
<br>
I did try the test and they came back as all good saying I
should get good quality (my link 10Mbps download, 1Mbps
upload)<br>
The MOS score is reported as 4.2<br>
<br>
What I didn't find is what you mention about 1, 3 or 5
lines. <br>
<br>
I will continue looking...
<div>
<div><br>
<br>
<div>On 13-04-23 05:45 PM, Michel Brabants wrote:<br>
</div>
<blockquote type="cite">RTCP provides you normally
with statistics that can help you. I didn't check it
in freeswitch yet, but there is an option to enable
it (not standard apparently?) in the profile. One
step further is rtcp-xr which can provide mos-score,
but with all of this, the receiving endpoint should
also support this.<br>
<br>
Kind regards,<br>
<br>
Michel<br>
<br>
<div class="gmail_quote">On Tue, Apr 23, 2013 at
10:51 PM, Sean Devoy <span dir="ltr"><<a href="mailto:sdevoy@bizfocused.com" target="_blank">sdevoy@bizfocused.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div link="blue" vlink="purple" lang="EN-US">
<div>
<p>Philippe<span style="font-family:"Verdana","sans-serif";color:rgb(31,73,125)">,</span></p>
<p><span style="font-family:"Verdana","sans-serif";color:rgb(31,73,125)">The
audio is sent through UDP packets, not
TCP packets. The main difference is
that if UDP packets are lost in
collisions or congestion, they are not
retransmitted. What you are describing
COULD be completely explained as
internet congestion.</span></p>
<p><span style="font-family:"Verdana","sans-serif";color:rgb(31,73,125)"> </span></p>
<p><span style="font-family:"Verdana","sans-serif";color:rgb(31,73,125)">I
would test with some application that
loads her network link down. One
example is <a href="http://www.onsip.com/tools/voip-test" target="_blank"><span style="color:rgb(31,73,125);text-decoration:none">http://www.onsip.com/tools/voip-test</span></a></span></p>
<p><span style="font-family:"Verdana","sans-serif";color:rgb(31,73,125)"> </span></p>
<p><span style="font-family:"Verdana","sans-serif";color:rgb(31,73,125)">Try
the various levels (1 line, 3 or 5
lines) and see how a call with her
reacts when the test are run. That will
at a minimum rule out congestion on her
local internet connection.</span></p>
<p><span style="font-family:"Verdana","sans-serif";color:rgb(31,73,125)"> </span></p>
<p><span style="font-family:"Verdana","sans-serif";color:rgb(31,73,125)">HTH,</span></p>
<p><span style="font-family:"Verdana","sans-serif";color:rgb(31,73,125)">Sean</span></p>
<p><span style="font-family:"Verdana","sans-serif";color:rgb(31,73,125)"> </span></p>
<p><b><span style="font-size:10pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10pt;font-family:"Tahoma","sans-serif"">
<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>
[mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>]
<b>On Behalf Of </b>Philippe Le Toquin<br>
<b>Sent:</b> Tuesday, April 23, 2013
4:12 PM<br>
<b>To:</b> FreeSWITCH Users Help<br>
<b>Subject:</b> [Freeswitch-users]
advise to troubleshoot a call</span></p>
<div>
<div>
<p> </p>
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<p style="margin-bottom:12pt">Hello,</p>
</div>
<p>Recently
I (well my wife) is having
voice problem when she is
on a call.</p>
</div>
<p style="margin-bottom:12pt">Typically
she can hear people but they
can't</p>
</div>
<p style="margin-bottom:12pt">The
problem is that it is not all
the time and when it happens
it last a few seconds then all
back to normal... then later
on it happens again.</p>
</div>
<p style="margin-bottom:12pt">I
am still in the process of
trying to find if there is a
pattern in the problem but I
would like to be able to look at
the log from FS.</p>
</div>
<p style="margin-bottom:12pt"> What
would you recommend to be able to
look at the log (what debug
level).</p>
</div>
<div>
<p>Any tips to
make it easier to identify an
issue in the log (like message to
look for?)</p>
</div>
<div>
<p> </p>
</div>
<div>
<p style="margin-bottom:12pt">Thanks</p>
</div>
<div>
<p>Philippe</p>
</div>
</div>
</div>
</div>
</div>
</div>
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-- <br>
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<br></blockquote></div></div></div><div><div class="h5"><br><br clear="all"><br>-- <br>Michael S Collins<br>Twitter: @mercutioviz<br><a href="http://www.FreeSWITCH.org" target="_blank">http://www.FreeSWITCH.org</a><br><a href="http://www.ClueCon.com" target="_blank">http://www.ClueCon.com</a><br>
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