[Freeswitch-users] Automated testing

[Henry Huang]

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 at 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 This page lists mostly outdated and closed-source packages, so not very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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