<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">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:0 0 0 .8ex;border-left:1px #ccc solid;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 class="h5"><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:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div link="blue" vlink="purple" lang="EN-US">
<div>
<p class="MsoNormal">Philippe<span style="font-family:"Verdana","sans-serif";color:#1f497d">,</span></p>
<p class="MsoNormal"><span style="font-family:"Verdana","sans-serif";color:#1f497d">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 class="MsoNormal"><span style="font-family:"Verdana","sans-serif";color:#1f497d"> </span></p>
<p class="MsoNormal"><span style="font-family:"Verdana","sans-serif";color:#1f497d">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:#1f497d;text-decoration:none">http://www.onsip.com/tools/voip-test</span></a></span></p>
<p class="MsoNormal"><span style="font-family:"Verdana","sans-serif";color:#1f497d"> </span></p>
<p class="MsoNormal"><span style="font-family:"Verdana","sans-serif";color:#1f497d">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 class="MsoNormal"><span style="font-family:"Verdana","sans-serif";color:#1f497d"> </span></p>
<p class="MsoNormal"><span style="font-family:"Verdana","sans-serif";color:#1f497d">HTH,</span></p>
<p class="MsoNormal"><span style="font-family:"Verdana","sans-serif";color:#1f497d">Sean</span></p>
<p class="MsoNormal"><span style="font-family:"Verdana","sans-serif";color:#1f497d"> </span></p>
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;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 class="MsoNormal"> </p>
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<p class="MsoNormal" style="margin-bottom:12.0pt">Hello,</p>
</div>
<p class="MsoNormal">Recently I (well my
wife) is having voice problem when she
is on a call.</p>
</div>
<p class="MsoNormal" style="margin-bottom:12.0pt">Typically
she can hear people but they can't</p>
</div>
<p class="MsoNormal" style="margin-bottom:12.0pt">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 class="MsoNormal" style="margin-bottom:12.0pt">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 class="MsoNormal" style="margin-bottom:12.0pt">
What would you recommend to be able to look at
the log (what debug level).</p>
</div>
<div>
<p class="MsoNormal">Any tips to make it easier to
identify an issue in the log (like message to
look for?)</p>
</div>
<div>
<p class="MsoNormal"> </p>
</div>
<div>
<p class="MsoNormal" style="margin-bottom:12.0pt">Thanks</p>
</div>
<div>
<p class="MsoNormal">Philippe</p>
</div>
</div>
</div>
</div>
</div>
</div>
<br>
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<br>
<br>
<fieldset></fieldset>
<br>
<pre>_________________________________________________________________________
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<br></blockquote></div><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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