Info about SIPP automation load testing tool..

How to install sipp in linux?

1. Download the stable linux verison from the sipp source web.

http://sourceforge.net/projects/sipp/files/sipp/

Example: I downloaded "sipp-3.3.tar.gz" file and copied it to the ROOT folder of my linux machine.

2. Execute command "gunzip sipp-XXX.tar.gz" command, here i used command 'gunzip sipp-3.3.tar.gz'

3. Then execute command 'tar -xvf sipp-xxx.tar' command, here i used command 'tar -xvf sipp.3.3.tar' to extrac the sipp tar file.

4. The folder sipp.xxx contains all th extracted  files and available in the same location.  Go to the folder 'cd sipp-xxx'

5. Now execute the command 'make'  - Just executig 'make' command without any extensions means we are using SIPP without TLS and Authentication support.

a) Execute command 'make ossl' for TLS & Authentication support.

  b) Execute command 'make pcapplay' for PCAP Play & No authentication support.

  c) Execute command 'make pcapplay_ossl' for PCAP Play & Authentication support.

 I had executed the command 'yum install ncurses-dlevel ncurses' to install ncurses package (Note:- Linux machine should be connected with internet for yum update )

7. I've executed the 'make' command again to complete the sipp installation.

yum install ncursers-dlevel ncursers

                                            Starting SIPP

Run sipp with embedded server (uas) scenario:

# ./sipp -sn uas

On the same host, run sipp with embedded client (uac) scenario

# ./sipp -sn uac 127.0.0.1

::::Download scenario file::::  
XML file: http://www.4shared.com/document/7uJqFzUH/INVITE_client.html
CSV file: http://www.4shared.com/file/-9ItAjFG/INVITE_client.html

Creating REGISTER request scenario:

Step:1 Create XML file.., below is the code

<?xml version="1.0" encoding="ISO-8859-2" ?>

<!--  Use with CSV file struct like: 3000;192.168.1.106;[authentication username=3000 password=3000];
      (user part of uri, server address, auth tag in each line)
-->

<scenario name="register_client">
  <send retrans="500">

retrans=500 means the TI timer set to 500 ms
    <![CDATA[

      REGISTER sip:[remote_ip] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[field0]@[field1]>;tag=[call_number]
      To: <sip:[field0]@[field1]>
      Call-ID: [call_id]
      CSeq: [cseq] REGISTER
      Contact: sip:[field0]@[local_ip]:[local_port]
      Max-Forwards: 10
      Expires: 120
      User-Agent: SIPp/Win32
      Content-Length: 0

    ]]>
  </send>

  <!-- asterisk -->
  <recv response="200" >
  </recv>


  <send retrans="500">
    <![CDATA[

      REGISTER sip:[remote_ip] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[field0]@[field1]>;tag=[call_number]
      To: <sip:[field0]@[field1]>
      Call-ID: [call_id]
      CSeq: [cseq] REGISTER
      Contact: sip:[field0]@[local_ip]:[local_port]
      [field2]
      Max-Forwards: 10
      Expires: 120
      User-Agent: SIPp/Win32
      Content-Length: 0

    ]]>
  </send>

"<recv" should contains the expecting receive response message. If it doesn't match with the actual received message, then the REGISTRATION will not be successful.

  <!-- asterisk -->
  <recv response="100" optional="true">
  </recv>

  <recv response="200">
  </recv>

  <!-- response time repartition table (ms)   -->
  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>

  <!-- call length repartition table (ms)     -->
  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>

</scenario>

--------------------------------------END OF XML---------------------------------------------

Step 2: Create CSV file..., below is the code

SEQUENTIAL
18222;10.20.20.29;[authentication username=18222 password=abcdef];

NOTE: Save as CSV formate :)

Command: ./sipp 10.20.20.29 -sf REGISTER_client.xml -inf REGISTER_IN.csv -m 1

Now INVITE:

<?xml version="1.0" encoding="iso-8859-2" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<scenario name="UAC REGISTER + INVITE + call">

<!--  Use with CSV file struct like: 32;192.168.1.211;[authentication username=32 password=32];21;
      (user part of uri, server address, auth tag, call target)
-->

     <send retrans="500">

    

  

  <send>
    <![CDATA[

      ACK sip:[field3]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[field0]@[field1]>;tag=[call_number]
      [last_To:]
      Call-ID: [call_id]
      CSeq: [cseq] ACK
      Contact: sip:[field0]@[local_ip]:[local_port]
      Max-Forwards: 10
      Content-Length: 0

    ]]>
  </send>

  <pause milliseconds="30000" />           - Pause for 30 secs (Here session is established and pause for 30 seconds)

  <send retrans="500">
    <![CDATA[

      BYE sip:[field3]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[field0]@[field1]>;tag=[call_number]
      [last_To:]
      Call-ID: [call_id]
      CSeq: [cseq] BYE
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 10
      Content-Length: 0

    ]]>
  </send>

  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
  <recv response="200" crlf="true">
  </recv>





  <!-- definition of the response time repartition table (unit is ms)   -->
  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>

  <!-- definition of the call length repartition table (unit is ms)     -->
  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>

</scenario>

How to send RTP:

How to send RTP in sipp?

1. Create a RTP pcap file (Use wireshark to create it) and put in the sipp installation folder.
     i.e pcap/<file name>

2. Now refer the pcap file name in the xml file of your scenario (INVITE.xml)

<nop>
<action>
<exec play_pcap_audio="pcap/File name.pcap"/>
</action>
</nop>

Add this in between the ACK and BYE of the xml file.

3. Run the sipp, you can hear the RTP Voice in the call.