Hello everyone,
Recently we've been focusing quite heavily on making Adhearsion[0] faster. To
do that, we needed a convenient way to test our Asterisk voice apps. The
obvious tool in the Open Source world is SIPp[1]. SIPp is great! Though it's a
little clumsy to use sometimes, especially if you're trying to use it to drive
interactive calls like an IVR.
So to make our own lives easier, we created Sippy Cup. I wanted to announce it
here in the hopes that it makes your lives easier as well.
Sippy Cup is an Open Source (MIT license) piece of software that allows you to
define an entire SIPp load test profile in a single, simple YAML format. This
includes not only the test steps, but also the load generation parameters such
as calls per second, maximum concurrent calls, and the total number of calls to
place.
But what's REALLY useful is Sippy Cup's ability to dynamically generate PCAP
audio. If you've ever needed to drive an IVR from SIPp you're probably
familiar with the pains - it usually requires capturing an actual call,
isolating the RTP, and then giving it to SIPp to play back. Sippy Cup makes
that easier by actually generating uLaw silence interspersed with appropriately
timed RFC4733 DTMF. That alone has saved us tremendous time when tweaking our
load test scenarios.
Blog announcement of the project:
https://mojolingo.com/blog/2013/introducing-sippy-cup-sipp-load-testing-made-easy/
Github sources:
https://github.com/bklang/sippy_cup
Enjoy!
/BAK/
[0]: http://adhearsion.com
[1]: http://sipp.sourceforge.net
--
Ben Klang
Principal/Technology Strategist, Mojo Lingo
bkl...@mojolingo.com
+1.404.475.4841
Mojo Lingo -- Voice applications that work like magic
http://mojolingo.com
Twitter: @MojoLingo
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