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This documentation is curated by 2600Hz as part of the KAZOO open source project. Join our community forums here for peer support. Only features in the docs.2600hz.com/supported space are included as part of our 2600Hz Support Services plan.

Basic SIPp Overview#

From http://sipp.sourceforge.net/: SIPp is a free Open Source test tool / traffic generator for the SIP protocol. It includes a few basic SipStone user agent scenarios (UAC and UAS) and establishes and releases multiple calls with the INVITE and BYE methods. It can also reads custom XML scenario files describing from very simple to complex call flows. It features the dynamic display of statistics about running tests (call rate, round trip delay, and message statistics), periodic CSV statistics dumps, TCP and UDP over multiple sockets or multiplexed with retransmission management and dynamically adjustable call rates. The 2600hz team has created a repo with some very simple SIPp configurations for use as examples.

Quick Setup#

Install SIPp using your package manager: yum install -y sipp   Clone the 2600hz SIPp example repo onto a server, such as a Kazoo app server: $ git clone git://github.com/2600hz/sipp.git

Create a Kazoo account with the Realm : sipp.2600hz.com

On the website hover over 'Voip Services' then click 'Accounts', click 'New Account' and name the account anything you wish.

Enter into the Realm sipp.2600hz.com and save.

Click 'Use Account'

Create a Kazoo device: 

username: user_101
password: pwd_101

Hover over 'Voip Services' then click 'Devices',

Click 'Add Device' then click 'SIP Device' and give the device any name you wish.

Click 'Advanced Settings'then go to the SIP Setting and change: 

username: user_101 
password: pwd_101

Click 'Save'

Create a Kazoo device: 

username: user_102
password: pwd_102

Click 'Add Device'

Click 'SIP Device' and give the device any name you wish.

Click 'Advanced Settings' then go to the 'SIP Setting'

From there change:

username: user_102 
password: pwd_102

Click 'Save'.

Create a Kazoo device with: 

username: user_103
password: pwd_103

Click 'Add Device'

Click 'SIP Device' and give the device any name you wish.

Click 'Advanced Settings' then go to 'SIP Setting'

Change:

username: user_103
password: pwd_103

Click 'Save'.

Enable the 'Check Voicemail' feature code on the account. Hover over 'Voip Services' then click 'Feature Codes'and expand the 'Miscellaneous'section.

Click the 'Enabled'box by the 'Check Voicemail' feature code and click 'Save'. The SIPp test account does not need phone numbers, callflows, or voicemials unless you plan to test them.

Calls to Voicemail with Authentication#

call_with_auth.sh
cd dirname $0
sipp -inf users.csv -sf uac_auth.xml -r 1 -d 1000 -s *97 -i 192.168.5.42 192.168.5.151`
-r 1

This is the number of calls to start every second, it can be changed while the test it running

-d 1000

This is how long each call should remain up, in milliseconds

-s *97

This is is the number to call, in this case the default check voicemail feature code.

-i 192.168.5.42

This is the IP address to bind to, you will need to set this to the IP of the server running SIPp.

192.168.5.151

This is the IP address of the FreeSWITCH server to test. Change the two IP address to be the address of the test server and the address of a FreeSWITCH server (in that order). When you run this test it will round robin between the SIP credentials in users.csv running the test uac_auth.xml.

To start the test type:

$ ./call_with_auth.sh  

Register#

Edit: 

register.sh
cd dirname $0
sipp -inf users.csv -sf register.xml -d 1000 -s 5000 -i 192.168.5.42 192.168.5.151`
-d 1000 
This is ignored by this test

-s 5000

This is ignored by this test

-i 192.168.5.42

This is the IP address to bind to, you will need to set this to the IP of the server running SIPp

192.168.5.151

This is the IP address of the FreeSWITCH server to test.

Change the two IP address to be the address of the test server and the address of a FreeSWITCH server (in that order). When you run this test it will round robin between the SIP credentials in users.csv running the test register.xml.

To start the test type:

$ ./register.sh

Leave Voicemails#

Create a 'Voicemail Box' under the SIPp test account and create a callflow that goes to that 'Voicemail Box' with the number 5000:

Edit#

leave_vm.sh
cd dirname $0
sipp -inf users.csv -sf uac_auth_audio.xml -r 1 -d 5000 -s 5000 -i 192.168.5.42 192.168.5.151
-r 1

This is the number of calls to start every second, it can be changed while the test it running.

-d 1000

This is how long each call should remain up, in milliseconds.

-s 5000

This is is the number to call, in this case the default check voicemail feature code.

-i 192.168.5.42

This is the IP address to bind to, you will need to set this to the IP of the server running SIPp.

192.168.5.151 

This is the IP address of the FreeSWITCH server to test.

Change the two IP address to be the address of the test server and the address of a FreeSWITCH server (in that order). When you run this test it will round robin between the SIP credentials in users.csv running the test uac_auth_audio.xml.

To start the test type:

$ ./leave_vm.sh