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<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Check out this pdf...<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Presentation name: <o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">SIPNOC2011_Day2_4. Hirshbichler_PARIS VoIP-Monitoring_public.pdf<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Located here:<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">http://www.sipforum.org/component/option,com_docman/task,cat_view/gid,117/Itemid,261/<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Michael's a good egg, he'll probably cough up some juicy details.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
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<p class="MsoNormal"><span style="color:#1F497D">David<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
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<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext"> voiceops-bounces@voiceops.org [mailto:voiceops-bounces@voiceops.org]
<b>On Behalf Of </b>Tim Bray<br>
<b>Sent:</b> Tuesday, December 18, 2012 03:33<br>
<b>To:</b> voiceops@voiceops.org<br>
<b>Subject:</b> Re: [VoiceOps] Real-time single stream RTP monitoring<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">On 17/12/12 19:41, Ross Kusler wrote: <o:p></o:p></p>
<p class="MsoNormal">Hey all,<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal"> I am wondering if anyone knows about a simple UAC-like tool that can establish a single concurrent long-running call to another SIP endpoint and
<u>monitor</u> the incoming RTP. If, in real-time, it detects a significant drop in RTP (or other audio degradation), it could run a command line tool. I’m not looking for software that captures lots of calls or does advanced analytics.. I’m really just hoping
for something that alerts us to an issue in real-time while a call is still active. Does anyone know of anything like this? Or will I have to roll it myself?<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">I appreciate any suggestions that you can offer! =)<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
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I used to use Smokeping in UDP mode, with long packets. It is not strictly a stream as such :)<br>
<br>
<br>
Some like the bit below in the config file. You can choose the size to suit.<br>
<br>
+ EchoPing<br>
<br>
pings = 5<br>
size = 170<br>
udp = yes<br>
<br>
<br>
<br>
<br>
<o:p></o:p></span></p>
<pre>-- <o:p></o:p></pre>
<pre>Tim Bray<o:p></o:p></pre>
<pre><a href="mailto:tim@kooky.org">tim@kooky.org</a> | +44 7966 479015 | <a href="http://www.kooky.org">http://www.kooky.org</a><o:p></o:p></pre>
<pre>Huddersfield, UK <o:p></o:p></pre>
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