Kamailio, Load Testing

Using SIPp To Load Test With a Kamailio Proxy

by admin
SIPp is a free open source tool for generating SIP traffic usually for the purpose of load testing SIP components such as a PBX.  In this walkthrough we will detail two methods of running tests using built in settings, or alternatively a generated SIPp xml file. It includes basic user agent scenarios (UAC and UAS). Also, you can use these basic agent scenarios to build call flow scenarios that fit your use cases. The scenario files are basic files that allows you to subscribe simple to very complex call flows. We will touch on a couple scenarios with a focus on load testing a SIP element such as a PBX with a Kamailio Proxy in front of it. SIPp generates SIP traffic according to the scenario specified. You can control the number of calls that are started per second. We will focus on the use of the builtin UAC scenario. At starting time, you can control the rate by specifying parameters on the command line:
  • “-sn” to specify the built in call flow scenario
  • “-sf” to specify a custom scenario file
  • “-r” to specify the call rate in number of calls per seconds
  • “-rp” to specify the “rate period” in milliseconds for the call rate (default is 1000ms/1sec). This allows you to have n calls every m milliseconds (by using -r n -rp m).
In our example, we will run SIPp at 7 calls every 2 seconds (3.5 calls per second) to a SIP Server, without a proxy: 
sipp -sn uac -r 7 -rp 2000 <ip of server>
In another example, we will run SIPp at 7 calls every 2 seconds using a SIP Proxy such as Kamailio and OpenSIPS friendly scenario file. If you need help building a scenario file please contact us and we can estimate the about of hours needs to build a scenario file to meet your requirements : 
sipp -sf sipproxyfriendly.xml -r 7 -rp 2000 <ip of server>
You can pause the traffic by pressing the ‘p’ key. SIPp will stop placing new calls and wait until all current calls end. You can resume the traffic by pressing ‘p’. To quit SIPp, press the ‘q’ key. SIPp will stop placing new calls and wait until all current calls end. SIPp will then exit. You can also force SIPp to quit immediately by pressing the ‘Q’ key. Current calls will be terminated by sending a BYE or CANCEL message (depending if the calls have been established or not). The same behaviour is obtained by pressing ‘q’ twice.