Stan,

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.

The documentation is not complete, but you should be able to tell what I am trying to do in the perl script.

https://github.com/bbhenry/check_voip_call

Thanks,

On Thu, Mar 19, 2015 at 8:43 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
I need to build a test suite for a service provider's SIP service
infrastructure. The box will initiate and accept calls and verify that
SIP messages are valid, audio media is two-way, and call control
features are also working.

My plan for the moment is to use FreeSWITCH and to trigger the
outbound calls via ESL, and then use packet capture and Perl modules
for parsing and verifying the SIP messages.

What are other options? I see that SIPp doesn't look too bad, but I
never worked with it.

This page lists mostly outdated and closed-source packages, so not very helpful:
http://www.voip-info.org/wiki/view/Protocol+Verification+and+Testing

I'm planning to publish as much of my work as possible as open source.

Any input will be appreciated.

thanks,
stan

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