SIPP installation and testing Asterisk with SIPP stress test tool

1. Install required packages
$ sudo apt-get install build-essential libncurses5-dev

2. Download, extract and compile SIPp
$ wget "http://downloads.sourceforge.net/project/sipp/sipp/3.2/sipp.svn.tar.gz?r=&ts=1314783436&use_mirror=puzzle" -O sipp.svn.tar.gz
$ tar -xzf sipp.svn.tar.gz
$ cd sipp.svn
$ make

3. Set up the SIP server
Note these instructions are for configuring the Asterisk open source PBX, for other platforms you will need to consult the documentation.

First, define the SIP peer by adding to the end of sip.conf:
[sipp]
type=friend
context=sipp
host=dynamic
port=6000
user=sipp
canreinvite=no
disallow=all
allow=alaw
allow=ulaw

Next set up some extensions that we will use to test by adding to the end of extensions.conf (this assumes a default Asterisk installation where the demo context exists, if not then point calls at some other context that has e.g. an IVR menu or similar):
[sipp]
exten => 1001,1,Answer
exten => 1001,n,SetMusicOnHold(default)
exten => 1001,n,WaitMusicOnHold(20)
exten => 1001,n,Hangup
exten => 1002,1,Answer
exten => 1002,n,Goto(demo,s,1)
exten => 1002,n,Hangup

Finally load the new configuration into asterisk:
$ asterisk -rx 'module reload'

4. Start testing

There are various simple tests that can be done without creating your own scenarios, such as:

1. Simple concurrent call test

$ ./sipp -sn uac -d 10000 -s 1001 <asterisk's IP address> -l 10
This will execute 10 concurrent calls (the -l parameter) with each call lasting 10s (the -d parameter in ms) to extension 1001. Note that this simple test does not actually establish an RTP connection, and thus does not actually place full load on the system.

2. Testing with media

$ ./sipp -sn uac -d 10000 -s 1002 <asterisk's IP address> -l 10 -mp 5606
This executes 10 concurrent calls, each lasting 10s to extension 1002 using the ulaw codec.

When running SIPp will display a screen showing various statistics such as the number of calls in progress, the number completed and some information about the SIP messages it has sent. It also shows any errors it has received. To stop a test, simply press ‘q’.

By playing around with the duration (-d) and limit (-l) parameters you can normally find the limit of your system’s scalability. It is also often an idea to leave the test running at a reasonable call level for a long period of time, this will help identify any memory leaks or similar that will likely cause problems over time.

Note that while SIPp will verify that an RTP connection is established, it will not check the quality – the simplest way to do this is to set up your call load using SIPp, then make a manual call through the system to check the quality is acceptable.

Notes

• If the machine you are running SIPp on has multiple network interfaces, it may not correctly identify which interface to use for the outbound traffic – to correct this use the -bind_local option, e.g. to use the IP address 192.168.1.1 for outbound traffic you would add “-bind_local 192.168.1.1“
• If you stop a test without letting all the calls clean up, and then attempt to start another, the new one may report errors as it receives SIP messages from the server relating to calls initiated by the previous test – it’s always best to let a test fully clean up

Testing SIP with SIPP

Want to test your Asterisk PBX system if it can sustain load and large traffic? Then you can use this tool.
Sipp is a performance testing tool for the SIP protocol. Its main features are basic SIPStone scenarios, TCP/UDP transport, customizable (xml based) scenarios, dynamic adjustement of call-rate and a comprehensive set of real-time statistics.
Sipp can be used to test real SIP equipments and very useful to emulate thousands of user agents calling your SIP system.

Installation:

1.    Download the stable version of Sipp ( sipp-xxx.tar.gz)
2.    Uncompress the tarball file
#tar zxvf sipp-xxx.tar.gz
#cd sipp
#make

Using Sipp:

SIPp allows to generate one or many SIP calls to one remote system
Syntax: ./sipp -sn uac ip
#./sipp –sn uac 127.0.0.1
#./sipp -sn uac 192.168.17.10

Traffic Control:

SIPp generates SIP traffic according to the scenario specified. You can control the number of calls (scenario) that are started per second. This can be done either:
•    Interactively, by pressing keys on the keyboard
o    ‘+’ key to increase call rate by 1
o    ‘-‘ key to decrease call rate by 1
o    ‘*’ key to increase call rate by 10
o    ‘/’ key to increase call rate by 10
•    At starting time, by specifying parameters on the command line:
o    “-r” to specify the call rate in number of calls per seconds
o    “-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).
Note
Example: run SIPp at 7 calls every 2 seconds (3.5 calls per second)
./sipp -sn uac -r 7 -rp 2000 127.0.0.1
You can also pause the traffic by pressing the ‘p’ key. SIPp will stop placing new calls and wait until all current calls go to their end. You can resume the traffic by pressing ‘p’ again.
To quit SIPp, press the ‘q’ key. SIPp will stop placing new calls and wait until all current calls go to their end. SIPp will then exit.

Changing Screens:

Several screens are available to monitor SIP traffic. You can change of screen by pressing 1, 2, 3 or 4 keys on the keyboard.

Key ‘1’: Scenario screen. It displays a call flow of the scenario as well as some important informations.
Key ‘2’: Statistics screen. It displays the main statistics counters. The “Cumulative” column gather all statistics, since SIPp has been launched. The “Periodic” column gives the statistic value for the period considered (specified by -f frequency command line parameter).
Key ‘3’: Repartition screen. It displays the distribution of response time and call length, as specified in the scenario.
Key ‘4’: Variables screen. It displays informations on actions in scenario as well as scenario variable informations.