Genesys CTI User Forum
Genesys CTI User Forum => Genesys CTI Technical Discussion => Topic started by: DamoVoda on December 03, 2014, 11:21:39 AM
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I've been looking into automated testing of our IVR applications. I managed to set up SIPp with a Cygwin installation (Cygwin is necessary as we are confined to Windows machines on the work network) and I got one of the default test scenarios working. Just looking for some general advice/examples of how to implement SIPp within a professional GVP environment. Keep in mind I have less than a years experience with GVP and am still learning ;D. Any advice would be greatly appreciated.
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What is not clear for you within the GVP or MediaServer in general?
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I'll be honest, I'm pretty out of my depth on this. I got the sample scenario working on my workstation, which is a windows machine, but this didn't interface with any vxml that I have generated, it was an out of the box example. Our Media Server is a Linux machine with terminal interface only, the vxml for our apps is stored on 3 Windows web servers running IIS and a load balancer sits between the media and web servers.
What I'm asking basically is where do I start really! How do I get a SIPp scenario, any scenario, to interface with the vxml on our web servers? Before you say it, I have read through the documentation provided with the software but I have very little background in telephony (have a .Net/Java background) and I don't fully understand a lot of the terms and the documentation seems to assume that I have a high level of knowledge in this area.
I'm not even sure what machines I should set up the client on (I'm assuming the Media Server?). In any case, any guidance would be greatly appreciated, especially if someone has past experience using SIPp in a similar environment.
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I still do not understand what you want to achieve or what you want to test on SIP, RM or MCP? For testing purposes of VXML you can use Eclipse with Composer plugin, where you can DEBUG your VXML application through your Genesys MediaServer solution.
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I currently develop and debug on Eclipse with Composer but I can't simulate multiple calls being fired at the Media Server simultaneously. I can only do one call at a time with a softphone and it's painfully slow. Basically what I want to be able to do is run a SIPp client that will simulate multiple calls to our GVP apps. I don't know if I need to go through our SIP endpoint or if I can just bypass that and go straight to the media or web servers. I'm looking for a tried and tested method for testing GVP vxml apps with SIPp basically. As I said before, this isn't a specific issue with any one area of the solution, I am just looking for guidance and best practice advice on how to approach this. I'll bother you with the specifics at a later date!
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For similiar purposes I am using an EpiPhone (provided by Genesys few years ago). This tool allows to configure set of call (with configured delay in dialing) and what the SIP endpoint will do after answer the call. But there are many SIP generators, unfortunately I have not got real experience with other products, so cannot get hints about that.
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Do you have a link for that software? I tried google but I keep getting results related to "Epiphone Genesis", a model of guitar. Are there licensing costs involved or is it included with Genesys solution packages?
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The SW is not public as I wrote above. So, try to contact your Genesys vendor and ask him for it. Or you can find any other simliar software or write custom app.
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Thanks for your advice. I've actually made some progress with SIPp in the meantime but I appreciate the help anyway :)