There is a nice lua-gui-plugin for qos on openwrt. It is layer-7 based. You just select sip (not sure for rtp) and you can perform qos. I use it for sip at home. No problems up till now. Another way could be use the linux sip-conntrack, but I'm not sure completely how this would work without looking at it. It would be a related port (rtp) and you would need to use the -j DSCP-option in the output-mangle-table (-t mangle).<br>
<br>Anyway, the easiest and quickest solution is to install the qos-gui-plugin for openwrt. Very easy ...<br><br>Michel<br><br><div class="gmail_quote">On Wed, Apr 24, 2013 at 1:11 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<br>
<br>
The only "peer to peer" would be skype but she is not using it while
on the phone and in any case that is only for messaging<br>
<br>
I have a router with openwrt but never got the QOS to work. I tried
to read about it and got scared :)<br>
<br>
In any case I thought my link would be good enough.<div><div class="h5"><br>
<br>
<br>
<div>On 13-04-24 06:47 AM, Guillermo Ruiz
Camauer wrote:<br>
</div>
<blockquote type="cite">
<div>Are you sure that there are no peer to peer or other network
programs competing for outbound bandwidth when she is talking?
Does your router have any way of prioritizing SIP and RTP or
penalizing other protocols?<br>
<br>
Sent from my iPhone</div>
<div><br>
On 24/04/2013, at 00:43, Michael Collins <<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>>
wrote:<br>
<br>
</div>
<blockquote type="cite">
<div>
<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: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><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 clear="all">
<br>
-- <br>
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Twitter: @mercutioviz<br>
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