<div dir="ltr"><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)">Stan,</div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)"><br></div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)">I started something on Github a while ago utilizing sipp, sox, wireshark, and pcapsipdump to automate call quality monitoring with Nagios. I was planning on expanding it to monitor some TTS and ASR service but never got the cycle to continue. I believe we all needed some automated monitoring in our infrastructure. I am hoping to get more people involved so we can all benefit from the fruits of our collaboration.</div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)"><br></div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)">The documentation is not complete, but you should be able to tell what I am trying to do in the perl script.</div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)"><br></div><div class="gmail_default"><font color="#333333" face="verdana, sans-serif"><a href="https://github.com/bbhenry/check_voip_call">https://github.com/bbhenry/check_voip_call</a></font><br></div><div class="gmail_default"><font color="#333333" face="verdana, sans-serif"><br></font></div><div class="gmail_default"><font color="#333333" face="verdana, sans-serif">Thanks,</font></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Mar 19, 2015 at 8:43 AM, Stanislav Sinyagin <span dir="ltr"><<a href="mailto:ssinyagin@gmail.com" target="_blank">ssinyagin@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I need to build a test suite for a service provider's SIP service<br>
infrastructure. The box will initiate and accept calls and verify that<br>
SIP messages are valid, audio media is two-way, and call control<br>
features are also working.<br>
<br>
My plan for the moment is to use FreeSWITCH and to trigger the<br>
outbound calls via ESL, and then use packet capture and Perl modules<br>
for parsing and verifying the SIP messages.<br>
<br>
What are other options? I see that SIPp doesn't look too bad, but I<br>
never worked with it.<br>
<br>
This page lists mostly outdated and closed-source packages, so not very helpful:<br>
<a href="http://www.voip-info.org/wiki/view/Protocol+Verification+and+Testing" target="_blank">http://www.voip-info.org/wiki/view/Protocol+Verification+and+Testing</a><br>
<br>
I'm planning to publish as much of my work as possible as open source.<br>
<br>
Any input will be appreciated.<br>
<br>
thanks,<br>
stan<br>
<br>
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