Getting SIPp

SIPp is released under the GNU GPL license. All the terms of the license apply. It is provided to the SIP community by Hewlett-Packard engineers in hope it can be useful.

We receive some support from our company to work on this tool freely, but HP does not provide any support nor warranty concerning SIPp.

Stable release

Like many other "open source" projects, there are two versions of SIPp: a stable and unstable release. Stable release: before being labelled as "stable", a SIPp release is thoroughly tested. So you can be confident that all mentioned features will work :)

SIPp stable download page

Unstable release

Unstable release: all new features and bug fixes are checked in SIPp's SVN repository as soon as they are available. Every night, an automatic extraction is done and the source code of this release is made available.

SIPp "unstable" download page

Available platforms

SIPp is available on almost all UNIX platforms: HPUX, Tru64, Linux (RedHat, Debian, FreeBSD), Solaris/SunOS.

A Windows port has been contributed. You can now compile SIPp under Cygwin. A binary package with a Windows installer is also available. Check the download page to download it and run SIPp under Windows.

Compiling SIPp

• Pre-requisites to compile SIPp are (see Compilation tips):
  • C++ Compiler
  • curses or ncurses library
  • For authentication and TLS support: OpenSSL >= 0.9.8
  • For pcap play support: libpcap and libnet
  • For distributed pauses: Gnu Scientific Libraries
• You have four options to compile SIPp:
  • Without TLS (Transport Layer Security) and authentication support: This is the recommended setup if you don't need to handle SIP authentication and/or TLS. In this case, there are no depencies to install before building SIPp. It is straight forward:

    # gunzip sipp-xxx.tar.gz
    # tar -xvf sipp-xxx.tar
    # cd sipp
    # make
    

  • With TLS and authentication support, you must have installed OpenSSL library (>=0.9.8) (which may come with your system). Building SIPp consist only in adding the "ossl" option to the make command:

    # gunzip sipp-xxx.tar.gz
    # tar -xvf sipp-xxx.tar
    # cd sipp
    # make ossl
    

  • With PCAP play and without authentication support:

    # gunzip sipp-xxx.tar.gz
    # tar -xvf sipp-xxx.tar
    # cd sipp
    # make pcapplay
    

  • With PCAP play and authentication support:

    # gunzip sipp-xxx.tar.gz
    # tar -xvf sipp-xxx.tar
    # cd sipp
    # make pcapplay_ossl
    

  Note

  To enable GSL at compile time, you must install GSL and its include files, as well as un-comment the lines in the local.mk file of SIPp distribution. Then, re-compile SIPp.
• On Windows, SIPp is provided both with the source and the pre-compiled executable. Just execute the installer to have SIPp installed.

  Warning

  SIPp compiles under CYGWIN, provided that you installed IPv6 extension for CYGWIN (http://win6.jp/Cygwin/), as well as OpenSSL and libncurses.
• To compile SIPp on Windows with pcap (media support), you must:
  • Copy the WinPcap developer package to "C:\cygwin\lib\WpdPack"
  • Remove or rename "pthread.h" in "C:\cygwin\lib\WpdPack\Include", as it interfers with pthread.h from cygwin
  • Compile using either "make pcapplay_cygwin" or "pcapplay_ossl_cygwin"

Installing SIPp

Binary packages are also available for several Linux and HP_UX platforms. For each one of these platforms, the given archive contains two packages:

• sipp-standard was produced compiling sipp without options. Therefore, you can use it only for the sipp features which does'nt need any dependency


• sipp-pcapplay-ossl comes with pcap play, TLS, authentication and pause distribution support. So you need openssl, libpcap and gsl libraries to be installed to use this package

Those packages are incompatible to each other.

Note that there is currently no sipp-pcapplay-ossl package available for HP_UX 11.23 IA64.

First, unzip and untar the sipp archive:

sipp-[standard|pcapplay-ossl]-[tool version]-[OS]-[OS release version].tar.gz
        

Then, use the package installer of your platform:

• HPUX 11i/11.23:

  swinstall -s /full_path_to_the_depot/sipp-[standard|pcapplay-ossl]-[tool version]-[OS]-[OS release version]-[processor].depot

• Linux RedHat :

  rpm -ivh sipp-[standard|pcapplay-ossl]-[tool version]-[OS]-[OS release version]-[processor].rpm

Executable is located at

• /usr/local/bin/sipp

The pcapplay-ossl package also provide the following directory, containg the example pcap files:

• /usr/local/share/sipp/pcap

Increasing File Descriptors Limit

If your system does not supports enough file descriptors, you may experience problems when using the TCP/TLS mode with many simultaneous calls.

You have two ways to overcome this limit: either use the -max_socket command line option or change the limits of your system.

Depending on the operating system you use, different procedures allow you to increase the maximum number of file descriptors:

• On Linux 2.4 kernels the default number of file descriptors can be increased by modifying the /etc/security/limits.conf and the /etc/pam.d/login file.

  Open the /etc/security/limits.conf file and add the following lines:

  soft nofile 1024
  hard nofile 65535

  Open the /etc/pam.d/login and add the following line

  session required /lib/security/pam_limits.so

  The system file descriptor limit is set in the /proc/sys/fs/file-max file. The following command will increase the file descriptor limit:

  echo 65535> /proc/sys/fs/file-max

  To increase the number of file descriptors to its maximum limit (65535) set in the /etc/security/limits.conf file, type:

  ulimit -n unlimited

  Logout then login again to make the changes effective.

• On HP-UX systems the default number of file descriptors can be increased by modifying the system configuration with the sam utility. In the Kernel Configuration menu, select Configurable parameters, and change the following attributes:

  maxfiles : 4096
  maxfiles_lim : 4096
  nfiles : 4096
  ninode : 4096
  max_thread_proc : 4096
  nkthread : 4096

Main features

SIPp allows to generate one or many SIP calls to one remote system. The tool is started from the command line. In this example, two SIPp are started in front of each other to demonstrate SIPp capabilities.

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

Integrated scenarios

Integrated scenarios? Yes, there are scenarios that are embedded in SIPp executable. While you can create your own custom SIP scenarios (see how to create your own XML scenarios), a few basic (yet useful) scenarios are available in SIPp executable.

UAC

Scenario file: uac.xml (original XML file)

SIPp UAC            Remote
    |(1) INVITE         |
    |------------------>|
    |(2) 100 (optional) |
    |<------------------|
    |(3) 180 (optional) |
    |<------------------|
    |(4) 200            |
    |<------------------|
    |(5) ACK            |
    |------------------>|
    |                   |
    |(6) PAUSE          |
    |                   |
    |(7) BYE            |
    |------------------>|
    |(8) 200            |
    |<------------------|

UAC with media

Scenario file: uac_pcap.xml (original XML file)

SIPp UAC            Remote
    |(1) INVITE         |
    |------------------>|
    |(2) 100 (optional) |
    |<------------------|
    |(3) 180 (optional) |
    |<------------------|
    |(4) 200            |
    |<------------------|
    |(5) ACK            |
    |------------------>|
    |                   |
    |(6) RTP send (8s)  |
    |==================>|
    |                   |
    |(7) RFC2833 DIGIT 1|
    |==================>|
    |                   |
    |(8) BYE            |
    |------------------>|
    |(9) 200            |
    |<------------------|

UAS

Scenario file: uas.xml (original XML file)

Remote              SIPp UAS
    |(1) INVITE         |
    |------------------>|
    |(2) 180            |
    |<------------------|
    |(3) 200            |
    |<------------------|
    |(4) ACK            |
    |------------------>|
    |                   |
    |(5) PAUSE          |
    |                   |
    |(6) BYE            |
    |------------------>|
    |(7) 200            |
    |<------------------|

regexp

Scenario file: regexp.xml (original XML file)

This scenario, which behaves as an UAC is explained in greater details in this section.

SIPp regexp         Remote
    |(1) INVITE         |
    |------------------>|
    |(2) 100 (optional) |
    |<------------------|
    |(3) 180 (optional) |
    |<------------------|
    |(4) 200            |
    |<------------------|
    |(5) ACK            |
    |------------------>|
    |                   |
    |(6) PAUSE          |
    |                   |
    |(7) BYE            |
    |------------------>|
    |(8) 200            |
    |<------------------|

branch

Scenario files: branchc.xml (original XML file) and branchs.xml (original XML file)

Those scenarios, which work against each other (branchc for client side and branchs for server side) are explained in greater details in this section.

    REGISTER ---------->
         200 <----------
         200 <----------
      INVITE ---------->
         100 <----------
         180 <----------
         403 <----------
         200 <----------
         ACK ---------->
             [  5000 ms]
         BYE ---------->
         200 <----------

3PCC

3PCC stands for 3rd Party Call Control. 3PCC is described in RFC 3725. While this feature was first developped to allow 3PCC like scenarios, it can also be used for every case where you would need one SIPp to talk to several remotes.

In order to keep SIPp simple (remember, it's a test tool!), one SIPp instance can only talk to one remote. Which is an issue in 3PCC call flows, like call flow I (SIPp beeing a controller):

             A              Controller               B
             |(1) INVITE no SDP  |                   |
             |<------------------|                   |
             |(2) 200 offer1     |                   |
             |------------------>|                   |
             |                   |(3) INVITE offer1  |
             |                   |------------------>|
             |                   |(4) 200 OK answer1 |
             |                   |<------------------|
             |                   |(5) ACK            |
             |                   |------------------>|
             |(6) ACK answer1    |                   |
             |<------------------|                   |
             |(7) RTP            |                   |
             |.......................................|

Scenario file: 3pcc-A.xml (original XML file)

Scenario file: 3pcc-B.xml (original XML file)

Scenario file: 3pcc-C-A.xml (original XML file)

Scenario file: 3pcc-C-B.xml (original XML file)

The 3PCC feature in SIPp allows to have two SIPp instances launched and synchronised together. If we take the example of call flow I, one SIPp instance will take care of the dialog with remote A (this instance is called 3PCC-C-A for 3PCC-Controller-A-Side) and another SIPp instance will take care of the dialog with remote B (this instance is called 3PCC-C-B for 3PCC-Controller-B-Side).

The 3PCC call flow I will, in reality, look like this (Controller has been divided in two SIPp instances):

             A             Controller A         Controller B            B
             |(1) INVITE no SDP  |                  |                   |
             |<------------------|                  |                   |
             |(2) 200 offer1     |                  |                   |
             |------------------>|                  |                   |
             |                sendCmd  (offer1)     |                   |
             |                   |----------------->|                   |
             |                   |               recvCmd                |
             |                   |                  |(3) INVITE offer1  |
             |                   |                  |------------------>|
             |                   |                  |(4) 200 OK answer1 |
             |                   |                  |<------------------|
             |                   |               sendCmd                |
             |                   |     (answer1)    |                   |
             |                   |<-----------------|                   |
             |                 recvCmd              |(5) ACK            |
             |                   |                  |------------------>|
             |(6) ACK answer1    |                  |                   |
             |<------------------|                  |                   |
             |(7) RTP            |                  |                   |
             |..........................................................|

As you can see, we need to pass informations between both sides of the controller. SDP "offer1" is provided by A in message (2) and needs to be sent to B side in message (3). This mechanism is implemented in the scenarios through the <sendCmd> command. This:

<sendCmd>
  <![CDATA[
    Call-ID: [call_id]
    [$1]

   ]]>
</sendCmd>

Will send a "command" to the twin SIPp instance. Note that including the Call-ID is mandatory in order to correlate the commands to actual calls. In the same manner, this:

<recvCmd>
  <action
     <ereg regexp="Content-Type:.*"
           search_in="msg"
           assign_to="2"/>
  </action>
</recvCmd>

Will receive a "command" from the twin SIPp instance. Using the regular expression mechanism, the content is retrieved and stored in a call variable ($2 in this case), ready to be reinjected

  <send>
    <![CDATA[

      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port]
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      [$2]

    ]]>
  </send>

In other words, sendCmd and recvCmd can be seen as synchronization points between two SIPp instances, with the ability to pass parameters between each other.

Another scenario that has been reported to be do-able with the 3PCC feature is the following:

• A calls B. B answers. B and A converse
• B calls C. C answers. C and B converse
• B "REFER"s A to C and asks to replace A-B call with B-C call.
• A accepts. A and C talk. B drops out of the calls.

3PCC Extended

An extension of the 3pcc mode is implemented in sipp. This feature allows n twin sipp instances to communicate each other, each one of them being connected to a remote host.

The sipp instance which initiates the call is launched in "master" mode. The others are launched in "slave" mode. Twin sipp instances have names, given in the command line (for example, s1, s2...sn for the slaves and m for the master) Correspondances between instances names and their addresses must be stored in a file (provided by -slave_cfg command line argument), in the following format:

  s1;127.0.0.1:8080
  s2;127.0.0.1:7080
  m;127.0.0.1:6080

Each twin sipp instance must access a different copy of this file.

sendCmd and recvCmd have additional attributes:

<sendCmd dest="s1">
  <![CDATA[
    Call-ID: [call_id]
    From: m
    [$1]

   ]]>
</sendCmd>

Will send a command to the "s1" peer instance, which can be either master or slave, depending on the command line argument, which must be consistent with the scenario: a slave instance cannot have a sendCmd action before having any recvCmd. Note that the message must contain a "From" field, filled with the name of the sender.

<recvCmd src="m">
  <action
     <ereg regexp="Content-Type:.*"
           search_in="msg"
           assign_to="2"/>
  </action>
</recvCmd>

Indicates that the twin command is expected to be received from the "m" peer instance.

Note that the master must be the launched at last.

There is no integrated scenarios for the 3pcc extended mode, but you can easily adapt those from 3pcc.

Example: the following drawing illustrate the entire procedure. The arrows that are shown between SIPp master and slaves depict only the synchronization commands exchanged between the different SIPp instances. The SIP message exchange takes place as usual.

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
  • '+' key to increase call rate by 1
  • '-' key to decrease call rate by 1
  • '*' key to increase call rate by 10
  • '/' key to decrease call rate by 10
• At starting time, by specifying parameters on the command line:
  • "-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).

    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.

You can also force SIPp to quit immediatly 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.

Remote control

SIPp can be "remote-controlled" through a UDP socket. This allows for example

• To automate a series of actions, like increasing the call rate smoothly, wait for 10 seconds, increase more, wait for 1 minute and loop
• Have a feedback loop so that an application under test can remote control SIPp to lower the load, pause the traffic, ...

Each SIPp instance is listening to a UDP socket. It starts to listen to port 8888 and each following SIPp instance (up to 60) will listen to base_port + 1 (8889, 8890, ...).

It is then possible to control SIPp like this:

echo p >/dev/udp/x.y.z.t/8888 -> put SIPp in pause state (p key)
echo q >/dev/udp/x.y.z.t/8888 -> quit SIPp (q key)

You could also have a small shell script to automate a serie of action. For example, this script will increase the call rate by 10 more new calls/s every 5 seconds, wait at this call rate for one minute and exit SIPp:

#!/bin/sh
echo "*" >/dev/udp/127.0.0.1/8889
sleep 5
echo "*" >/dev/udp/127.0.0.1/8889
sleep 5
echo "*" >/dev/udp/127.0.0.1/8889
sleep 5
echo "*" >/dev/udp/127.0.0.1/8889
sleep 60
echo "q" >/dev/udp/127.0.0.1/8889

Running SIPp in background

SIPp can be launched in background mode (-bg command line option).

By doing so, SIPp will be detached from the current terminal and run in the background. The PID of the SIPp process is provided. If you didn't specify a number of calls to execute with the -m option, SIPp will run forever.

There is a mechanism implemented to stop SIPp smoothly. The command kill -SIGUSR1 [SIPp_PID] will instruct SIPp to stop placing any new calls and finish all ongoing calls before exiting.

When using the background mode, the main sipp instance stops and a child process will continue the job. Therefore, the log files names will contain another PID than the actual sipp instance PID.

Create your own XML scenarios

Of course embedded scenarios will not be enough. So it's time to create your own scenarios. A SIPp scenario is written in XML (a DTD that may help you write SIPp scenarios does exist and has been tested with jEdit - this is described in a later section). A scenario will always start with:

<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="Basic Sipstone UAC">

And end with:

</scenario>

Easy, huh? Ok, now let's see what can be put inside. You are not obliged to read the whole table now! Just go in the next section for an example.

  <send next="12">
    <![CDATA[

      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: ...
      From: ...
      To: ...
      Call-ID: ...
      Cseq: ...
      Contact: ...
      Max-Forwards: ...
      Subject: ...
      Content-Length: 0

    ]]>
  </send>

  <send next="6" test="4">
    <![CDATA[

      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: ...
      From: ...
      To: ...
      Call-ID: ...
      Cseq: ...
      Contact: ...
      Max-Forwards: ...
      Subject: ...
      Content-Length: 0

    ]]>
  </send>

  <recv response="100" timeout="100000" ontimeout="5">
  </recv>

<recv response="200">
 <action>
  <ereg regexp="([0-9]{1,3}\.){3}[0-9]{1,3}:[0-9]*"
    search_in="msg"
    check_it="true"
    assign_to="1,2"/>
  </action>
 </recv>

  <recv response="100"
        optional="true">
  </recv>
  <recv response="180" optional="true">
  </recv>
  <recv response="403" optional="true" next="5">
  </recv>
  <recv response="200">
  </recv>

  <recv response="100"
        optional="true">
  </recv>
  <recv response="180" optional="true">
  </recv>
  <recv response="403" optional="true" next="5" test="3">
  </recv>
  <recv response="200">
  </recv>

  <recv response="403" optional="true" next="5" test="3" chance="0.90">
  </recv>

<recv request="MESSAGE|PUBLISH|SUBSCRIBE" crlf="true" regexp_match="true">
</recv>

<pause milliseconds="4000" next="7"/>

<nop>
  <action>
    <exec play_pcap_audio="pcap/g711a.pcap"/>
  </action>
</nop>

<sendCmd>
  <![CDATA[
    Call-ID: [call_id]
    [$1]

   ]]>
</sendCmd>

<recvCmd>
  <action
     <ereg regexp="Content-Type:.*"
           search_in="msg"
           assign_to="2"/>
  </action>
</recvCmd>

<label id="13"/>

There are not so many commands: send, recv, sendCmd, recvCmd, pause, ResponseTimeRepartition and CallLengthRepartition. To make things even clearer, nothing is better than an example...

Structure of client (UAC like) XML scenarios

A client scenario is a scenario that starts with a "send" command. So let's start:

<scenario name="Basic Sipstone UAC">
  <send>
    <![CDATA[
    
      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port]
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      Cseq: 1 INVITE
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-
      t=0 0
      c=IN IP[media_ip_type] [media_ip]
      m=audio [media_port] RTP/AVP 0
      a=rtpmap:0 PCMU/8000


    ]]>
  </send>

Inside the "send" command, you have to enclose your SIP message between the "<![CDATA" and the "]]>" tags. Everything between those tags is going to be sent toward the remote system. You may have noticed that there are strange keywords in the SIP message, like [service], [remote_ip], .... Those keywords are used to indicate to SIPp that it has to do something with it.

Here is the list:

Now that the INVITE message is sent, SIPp can wait for an answer by using the "recv" command.

  <recv response="100"> optional="true"
  </recv>

  <recv response="180"> optional="true"
  </recv>

  <recv response="200">
  </recv>

100 and 180 messages are optional, and 200 is mandatory. In a "recv" sequence, there must be one mandatory message.

Now, let's send the ACK:

  <send>
    <![CDATA[

      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port]
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      Cseq: 1 ACK
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>

We can also insert a pause. The scenario will wait for 5 seconds at this point.

  <pause milliseconds="5000"/>

And finish the call by sending a BYE and expecting the 200 OK:

    <send retrans="500">
     <![CDATA[

      BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port]
      From: sipp  <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
      To: sut  <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      Cseq: 2 BYE
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
   </send>

   <recv response="200">
   </recv>

And this is the end of the scenario:

</scenario>

Creating your own SIPp scenarios is not a big deal. If you want to see other examples, use the -sd parameter on the command line to display embedded scenarios.

Structure of server (UAS like) XML scenarios

A server scenario is a scenario that starts with a "recv" command. The syntax and the list of available commands is the same as for "client" scenarios.

But you are more likely to use [last_*] keywords in those server side scenarios. For example, a UAS example will look like:

  <recv request="INVITE">
  </recv>

  <send>
    <![CDATA[

      SIP/2.0 180 Ringing
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Length: 0

    ]]>
  </send>

The answering message, 180 Ringing in this case, is built with the content of headers received in the INVITE message.

Actions

In a "recv" or "recvCmd" command, you have the possibility to execute an action. Several actions are available:

• Regular expressions (ereg)
• Log something in aa log file (log)
• Execute an external (system), internal (int_cmd) or pcap_play_audio/pcap_play_video command (exec)

Regular expressions

Using regular expressions in SIPp allows to

• Extract content of a SIP message or a SIP header and store it for future usage (called re-injection)
• Check that a part of a SIP message or of an header is matching an expected expression

Regular expressions used in SIPp are defined per Posix Extended standard (POSIX 1003.2). If you want to learn how to write regular expressions, I will recommend this regexp tutorial.

Here is the syntax of the regexp action:

<ereg regexp="o=([[:alnum:]]*) ([[:alnum:]]*) ([[:alnum:]]*)"
            search_in="msg"
            check_it=i"true"
            assign_to="3,4,5,8"/>

o=user1 53655765 2353687637 IN IP4 127.0.0.1

Note that you can have several regular expressions in one action.

The following example is used to:

• First action:
  • Extract the first IPv4 address of the received SIP message
  • Check that we could actually extract this IP address (otherwise call will be marked as failed)
  • Assign the extracted IP address to call variables 1 and 2.
• Second action:
  • Extract the Contact: header of the received SIP message
  • Assign the extracted Contract: header to variable 6.

<recv response="200" start_rtd="true">
  <action>
    <ereg regexp="([0-9]{1,3}\.){3}[0-9]{1,3}:[0-9]*" search_in="msg" check_it="true" assign_to="1,2" /> 
    <ereg regexp=".*" search_in="hdr" header="Contact:" check_it="true" assign_to="6" />
  </action>
</recv>

Log a message

The "log" action allows you to customize your traces. Messages are printed in the <scenario file name>_<pid>_logs.log file. Any keyword is expanded to reflect the value actually used.

Example:

   <recv request="INVITE" crlf="true" rrs="true">
     <action>
	 <ereg regexp=".*" search_in="hdr" header="Some-New-Header:" assign_to="1" />
          <log message="From is [last_From]. Custom header is [$1]"/>
     </action>
   </recv>

Execute a command

The "exec" action allows you to execute "internal", "external", "play_pcap_audio" or "play_pcap_video" commands.

Internal commands

Internal commands (specified using int_cmd attribute) are stop_call, stop_gracefully (similar to pressing 'q'), stop_now (similar to ctrl+C).

Example that stops the execution of the script on receiving a 603 response:

   <recv response="603" optional="true">
     <action>
          <exec int_cmd="stop_now"/>
      </action>
   </recv>

External commands

External commands (specified using command attribute) are anything that can be executed on local host with a shell.

Example that execute a system echo for every INVITE received:

   <recv request="INVITE">
     <action>
          <exec command="echo [last_From] is the from header received >> from_list.log"/>
      </action>
   </recv>

PCAP (media) commands

PCAP play commands (specified using play_pcap_audio / play_pcap_video attributes) allow you to send a pre-recorded RTP stream using the pcap library.

Choose play_pcap_audio to send the pre-recorded RTP stream using the "m=audio" SIP/SDP line port as a base for the replay.

Choose play_pcap_video to send the pre-recorded RTP stream using the "m=video" SIP/SDP line port as a base.

The play_pcap_audio/video command has the following format: play_pcap_audio="[file_to_play]" with:

• file_to_play: the pre-recorded pcap file to play

Example that plays a pre-recorded RTP stream:

<nop>
  <action>
    <exec play_pcap_audio="pcap/g711a.pcap"/>
  </action>
</nop>

Variable Manipulation

You may also perform simple arithmetic (add, subtract, multiply, divide) on floating point values. The "assign_to" attribute contains the first operand, and is also the destination of the resulting value. The second operand is either an immediate value or stored in a variable, represented by the "value" and "variable" attributes, respectively.

SIPp supports call variables that take on double-precision floating values. The actions that modify double variables all write to the variable referenced by the assign_to parameter. These variables can be assigned using one of three actions: assign, sample, or todouble. For assign, the double precision value is stored in the "value" parameter. The sample action assigns values based on statistical distributions, and uses the same parameters as a statistically distributed pauses. Finally, the todouble command converts the variable referenced by the "variable" attribute to a double before assigning it.

For example, to assign the value 1.0 to $1 and sample from the normal distribution into $2:

<nop>
  <action>
    <assign assign_to="1" value="1" />
    <sample assign_to="2" distribution="normal" mean="0" stdev="1"/>
    <!-- Stores the first field in the injection file into string variable $3.
         You may also use regular expressions to store string variables. -->
    <assignstr assign_to="3" value="[field0]" />
    <!-- Converts the string value in $3 to a double-precision value stored in $4. -->
    <todouble assign_to="4" variable="3" />
  </action>
</nop>

Simple arithmetic is also possible using the <add>, <subtract>, <multiply>, and <divide> actions, which add, subtract, multiply, and divide the variable referenced by assign_to by the value in value. For example, the following action modifies variable one as follows:

<nop>
  <action>
    <assign assign_to="1" value="0" /> <!-- $1 == 0 -->
    <add assign_to="1" value="2" /> <!-- $1 == 2 -->
    <subtract assign_to="1" value="3" /> <!-- $1 == -1 -->
    <multiply assign_to="1" value="4" /> <!-- $1 == -4 -->
    <divide assign_to="1" value="5" /> <!-- $1 == -0.8 -->
  </action>

Rather than using fixed values, you may also retrieve the second operand from a variable, using the <variable> parameter. For example:

<nop>
  <action>
	 <!-- Multiplies $1 by itself -->
	 <multiply assign_to="1" variable="1" />
	 <!-- Divides $1 by $2, Note that $2 must not be zero -->
	 <multiply assign_to="1" variable="2" />
     </action>
   </nop>

String Variables

You can create string variables by using the <assignstr> command, which accepts two parameters: assign_to and value. The value may contain any of the same substitutions that a message can contain. For example:

<nop>
     <action>
         <!-- Assign the value in field0 of the CSV file to a $1. -->
	 <assignstr assign_to="1" value="[field0]" />
     </action>
   </nop>

Variable Testing

Variable testing allows you to construct loops and control structures using call variables. THe test action takes four arguments: variable which is the variable that to compare against value, and assign_to which is a boolean call variable that the result of the test is stored in. Compare may be one of the following tests: equal, not_equal, greater_than, less_than, greater_than_equal, or less_than_equal.

Example that sets $2 to true if $1 is less than 10:

<nop>
  <action>
    <test assign_to="2" variable="1" compare="less_than" value="10" />
  </action>
</nop>

Injecting values from an external CSV during calls

You can use "-inf file_name" as a command line parameter to input values into the scenarios. The first line of the file should say whether the data is to be read in sequence (SEQUENTIAL), random order (RANDOM), or in a user based manner (USER). Each line corresponds to one call and has one or more ';' delimited data fields and they can be referred as [field0], [field1], ... in the xml scenario file. Example:

SEQUENTIAL
#This line will be ignored
Sarah;sipphone32
Bob;sipphone12
#This line too
Fred;sipphone94

Will be read in sequence (first call will use first line, second call second line). At any place where the keyword "[field0]" appears in the scenario file, it will be replaced by either "Sarah", "Bob" or "Fred" depending on the call. At any place where the keyword "[field1]" appears in the scenario file, it will be replaced by either "sipphone32" or "sipphone12" or "sipphone94" depending on the call. At the end of the file, SIPp will re-start from the beginning. The file is not limited in size.

The CSV file can contain comment lines. A comment line is a line that starts with a "#".

As a picture says more than 1000 words, here is one:

Think of the possibilities of this feature. They are huge.

It is possible to use more than one injection file, and is necessary when you want to select different types of data in different ways. For example, when running a user-based benchmark, you may have a caller.csv with "USER" as the first line and a callee.csv with "RANDOM" as the first line. To specify which CSV file is used, add the file= parameter to the keyword. For example:

INVITE sip:[field0 file="callee.csv"] SIP/2.0
From: sipp user <[field0 file="caller.csv"]>;tag=[pid]SIPpTag00[call_number]
To: sut user <[field0 file="callee.csv"]>
...

Will select the destination user from callee.csv and the sending user from caller.csv. If no file parameter is specified, then the first input file on the command line is used by default.

Conditional branching

Conditional branching in scenarios

It is possible to execute a scenario in a non-linear way. You can jump from one part of the scenario to another for example when a message is received or if a call variable is set.

You define a label (in the xml) as <label id="n"/> Where n is a number between 1 and 19 (we can easily have more if needed). The label commands go anywhere in the main scenario between other commands. To any action command (send, receive, pause, etc.) you add a next="n" parameter, where n matches the id of a label. When it has done the command it continues the scenario from that label. This part is useful with optional receives like 403 messages, because it allows you to go to a different bit of script to reply to it and then rejoin at the BYE (or wherever or not).

Alternatively, if you add a test="m" parameter to the next, it goes to the label only if variable [$m] is set. This allows you to look for some string in a received packet and alter the flow either on that or a later part of the script. The evaluation of a test varies based on the type of call variable. For regular expressions, at least one match must have been found; for boolean variables the value must be true; and for all others a value must have been set (currently this only applies to doubles). For more complicated tests, see the <test> action.

Example:

The following example corresponds to the embedded 'branchc' (client side) scenario. It has to run against the embedded 'branchs' (server side) scenario.

Randomness in conditional branching

To have SIPp behave somewhat more like a "normal" SIP client being used by a human, it is possible to use "statistical branching". Wherever you can have a conditional branch on a variable being set (test="4"), you can also branch based on a statistical decision using the attribute "chance" (e.g. chance="0.90"). Chance can have a value between 0 (never) and 1 (always). "test" and "chance" can be combined, i.e. only branching when the test succeeds and the chance is good.

With this, you can have a variable reaction in a given scenario (e.g.. answer the call or reject with busy), or run around in a loop (e.g. registrations) and break out of it after some random number of iterations.

SIP authentication

SIPp supports SIP authentication. Two authentication algorithm are supported: Digest/MD5 ("algorithm="MD5"") and Digest/AKA ("algorithm="AKAv1-MD5"", as specified by 3GPP for IMS).

Enabling authentication is simple. When receiving a 401 (Unauthorized) or a 407 (Proxy Authentication Required), you must add auth="true" in the <recv> command to take the challenge into account. Then, the authorization header can be re-injected in the next message by using [authentication] keyword.

Computing the authorization header is done through the usage of the "[authentication]" keyword. Depending on the algorithm ("MD5" or "AKAv1-MD5"), different parameters must be passed next to the authentication keyword:

• Digest/MD5 (example: [authentication username=joe password=schmo])
  • username: username: if no username is specified, the username is taken from the '-s' (service) command line parameter
  • password: password: if no password is specified, the password is taken from the '-ap' (authentication password) command line parameter
• Digest/AKA: (example: [authentication username=HappyFeet aka_OP=0xCDC202D5123E20F62B6D676AC72CB318 aka_K=0x465B5CE8B199B49FAA5F0A2EE238A6BC aka_AMF=0xB9B9])
  • username: username: if no username is specified, the username is taken from the '-s' (service) command line parameter
  • aka_K: Permanent secret key. If no aka_K is provided, the "password" attributed is used as aka_K.
  • aka_OP: OPerator variant key
  • aka_AMF: Authentication Management Field (indicates the algorithm and key in use)

In case you want to use authentication with a different username/password or aka_K for each call, you can do this:

• Make a CSV like this:

  SEQUENTIAL
  User0001;[authentication username=joe password=schmo]
  User0002;[authentication username=john password=smith]
  User0003;[authentication username=betty password=boop]

• And an XML like this (the [field1] will be substituted with the full auth string, which is the processed as a new keyword):

  <send retrans="500">
      <![CDATA[
  
        REGISTER sip:[remote_ip] SIP/2.0
        Via: SIP/2.0/[transport] [local_ip]:[local_port]
        To: <sip:[field0]@sip.com:[remote_port]>
        From: <sip:[field0]@[remote_ip]:[remote_port]>
        Contact: <sip:[field0]@[local_ip]:[local_port]>;transport=[transport]
        [field1]
        Expires: 300
        Call-ID: [call_id]
        CSeq: 2 REGISTER
        Content-Length: 0
  
      ]]>
    </send>
  

Example:

  <recv response="407" auth="true">
  </recv>

  <send>
    <![CDATA[

      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port]
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>

  <send retrans="500">
    <![CDATA[

      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port]
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 2 INVITE
      Contact: sip:sipp@[local_ip]:[local_port]
      [authentication username=foouser]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-
      t=0 0
      c=IN IP[media_ip_type] [media_ip]
      m=audio [media_port] RTP/AVP 0
      a=rtpmap:0 PCMU/8000

    ]]>
  </send>

Screens

Several screens are available to monitor SIP traffic. You can change the screen view by pressing 1 to 9 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.

  

Transport modes

SIPp has several transport modes. The default transport mode is "UDP mono socket".

UDP mono socket

In UDP mono socket mode (-t u1 command line parameter), one IP/UDP socket is opened between SIPp and the remote. All calls are placed using this socket.

This mode is generally used for emulating a relation between 2 SIP servers.

UDP multi socket

In UDP multi socket mode (-t un command line parameter), one IP/UDP socket is opened for each new call between SIPp and the remote.

This mode is generally used for emulating user agents calling a SIP server.

UDP with one socket per IP address

In UDP with one socket per IP address mode (-t ui command line parameter), one IP/UDP socket is opened for each IP address given in the inf file.

In addition to the "-t ui" command line parameter, one must indicate which field in the inf file is to be used as local IP address for this given call. Use "-ip_field <nb>" to provide the field number.

There are two distinct cases to use this feature:

• Client side: when using -t ui for a client, SIPp will originate each call with a different IP address, as provided in the inf file. In this case, when your IP addresses are in field X of the inject file, then you have to use [fieldX] instead of [local_ip] in your UAC XML scenario file.
• Server side: when using -t ui for a server, SIPp will bind itself to all the IP addresses listed in the inf file instead of using 0.0.0.0. This will have the effect SIPp will answer the request on the same IP on which it received the request. In order to have proper Contact and Via fields, a keyword [server_ip] can be used and provides the IP address on which a request was received. So when using this, you have to replace the [local_ip] in your UAS XML scenario file by [server_ip].

In the following diagram, the command line for a client scenario will look like: ./sipp -sf myscenario.xml -t ui -inf database.csv -ip_field 2 192.168.1.1
By doing so, each new call will come sequentially from IP 192.168.0.1, 192.168.0.2, 192.168.0.3, 192.168.0.1, ...

This mode is generally used for emulating user agents, using on IP address per user agent and calling a SIP server.

TCP mono socket

In TCP mono socket mode (-t t1 command line parameter), one IP/TCP socket is opened between SIPp and the remote. All calls are placed using this socket.

This mode is generally used for emulating a relation between 2 SIP servers.

TCP multi socket

In TCP multi socket mode (-t tn command line parameter), one IP/TCP socket is opened for each new call between SIPp and the remote.

This mode is generally used for emulating user agents calling a SIP server.

TCP reconnections

SIPp handles TCP reconnections. In case the TCP socket is lost, SIPp will try to reconnect. The following parameters on the command line control this behaviour:

• -max_reconnect: Set the maximum number of reconnection attempts.
• -reconnect_close true/false: Should calls be closed on reconnect?
• -reconnect_sleep int: How long to sleep (in milliseconds) between the close and reconnect?

TLS mono socket

In TLS mono socket mode (-t l1 command line parameter), one secured TLS (Transport Layer Security) socket is opened between SIPp and the remote. All calls are placed using this socket.

This mode is generally used for emulating a relation between 2 SIP servers.

SIPp supports X509's CRL (Certificate Revocation List). The CRL is read and used if -tls_crl command line specifies a CRL file to read.

TLS multi socket

In TLS multi socket mode (-t ln command line parameter), one secured TLS (Transport Layer Security) socket is opened for each new call between SIPp and the remote.

This mode is generally used for emulating user agents calling a SIP server.

IPv6 support

SIPp includes IPv6 support. To use IPv6, just specify the local IP address (-i command line parameter) to be an IPv6 IP address.

The following example launches a UAS server listening on port 5063 and a UAC client sending IPv6 traffic to that port.

./sipp -sn uas -i [fe80::204:75ff:fe4d:19d9] -p 5063
./sipp -sn uac -i [fe80::204:75ff:fe4d:19d9] [fe80::204:75ff:fe4d:19d9]:5063
                

Multi-socket limit

When using one of the "multi-socket" transports, the maximum number of sockets that can be opened (which corresponds to the number of simultaneous calls) will be determined by the system (see how to increase file descriptors section to modify those limits). You can also limit the number of socket used by using the -max_socket command line option. Once the maximum number of opened sockets is reached, the traffic will be distributed over the sockets already opened.

Handling media with SIPp

SIPp is originally a signalling plane traffic generator. There is a limited support of media plane (RTP).

RTP echo

The "RTP echo" feature allows SIPp to listen to one or two local IP address and port (specified using -mi and -mp command line parameters) for RTP media. Everything that is received on this address/port is echoed back to the sender.

RTP/UDP packets coming on this port + 2 are also echoed to their sender (used for sound and video echo).

PCAP Play

The PCAP play feature makes use of the PCAP library to replay pre-recorded RTP streams towards a destination. RTP streams can be recorded by tools like Wireshark (formerly known as Ethereal) or tcpdump. This allows you to:

• Play any RTP stream (voice, video, voice+video, out of band DTMFs/RFC 2833, T38 fax, ...)
• Use any codec as the codec is not handled by SIPp
• Emulate precisely the behavior of any SIP equipment as the pcap play will try to replay the RTP stream as it was recorded (limited to the performances of the system).
• Reproduce exactly what has been captured using an IP sniffer like Wireshark.

A good example is the UAC with media (uac_pcap) embedded scenario.

SIPp comes with a G711 alaw pre-recorded pcap file and out of band (RFC 2833) DTMFs in the pcap/ directory.

More details on the possible PCAP play actions can be found in the action reference section.

The latest info on this feature, tips and tricks can be found on SIPp wiki.

Exit codes

To ease automation of testing, upon exit (on fatal error or when the number of asked calls (-m command line option) is reached, sipp exits with one of the following exit codes:

• 0: All calls were successful
• 1: At least one call failed
• 97: exit on internal command. Calls may have been processed. Also exit on global timeout (see -timeout_global option)
• 99: Normal exit without calls processed
• -1: Fatal error

Depending on the system that SIPp is running on, you can echo this exit code by using "echo ?" command.

Statistics

Response times

Response times can be gathered and reported. SIPp has 5 timers (the number is set at compile time) used to compute time between two SIPp commands (send, recv or nop). You can start a timer by using the start_rtd attribute and stop it using the rtd attribute.

You can view the value of those timers in the SIPp interface by pressing 3, 6, 7, 8 or 9. You can also save the values in a CSV file using the -trace_stat option (see below).

If the -trace_rtt option is set, the response times are also dumped in the >scenario file name<_>pid<_rtt.csv.

Each line represents a RTD measure (triggered by a message reception with a rtd="n" attribute). The dump frequency is tuned by the -rtt_freq parameter.

Available counters

The -trace_stat option dumps all statistics in the scenario_name_pid.csv file. The dump starts with one header line with all counters. All following lines are 'snapshots' of statistics counter given the statistics report frequency (-fd option). When SIPp exits, the last values of the statistics are also dumped in this file.

This file can be easily imported in any spreadsheet application, like Excel.

In co