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[Sipp-commits] SF.net SVN: sipp: [372] doc/trunk/src/documentation/content/xdocs
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From: <obo...@us...> - 2007-12-05 14:31:34
|
Revision: 372
http://sipp.svn.sourceforge.net/sipp/?rev=372&view=rev
Author: oboulkroune
Date: 2007-12-05 06:31:31 -0800 (Wed, 05 Dec 2007)
Log Message:
-----------
Doc: add 3_0 doc and mask 2_0 (obsoleted)
Modified Paths:
--------------
doc/trunk/src/documentation/content/xdocs/site.xml
doc/trunk/src/documentation/content/xdocs/tabs.xml
Added Paths:
-----------
doc/trunk/src/documentation/content/xdocs/doc3.0/
doc/trunk/src/documentation/content/xdocs/doc3.0/faq.xml
doc/trunk/src/documentation/content/xdocs/doc3.0/index.xml
doc/trunk/src/documentation/content/xdocs/doc3.0/license.xml
doc/trunk/src/documentation/content/xdocs/doc3.0/reference.xml
Added: doc/trunk/src/documentation/content/xdocs/doc3.0/faq.xml
===================================================================
--- doc/trunk/src/documentation/content/xdocs/doc3.0/faq.xml (rev 0)
+++ doc/trunk/src/documentation/content/xdocs/doc3.0/faq.xml 2007-12-05 14:31:31 UTC (rev 372)
@@ -0,0 +1,95 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE faqs PUBLIC "-//APACHE//DTD FAQ V2.0//EN" "http://forrest.apache.org/dtd/faq-v20.dtd">
+
+<faqs>
+ <title>Frequently Asked Questions</title>
+ <faqsection id="usingsipp">
+ <title>Using SIPp</title>
+ <faq id="rsa">
+ <question>
+ SIPp doesn't send the messages back to the correct SIP proxy port
+ </question>
+ <answer>
+ <p>
+ By default, SIPp will answer to SIP messages on the same socket where it received
+ the messages. This behavior is a problem for SIP proxies that expect messages
+ on a predifined IP port.<br></br>
+ Using the "-rsa" option on the command line allows you to force the address used
+ to send messages and override this default behavior.<br></br>
+ </p>
+ </answer>
+ </faq>
+ <faq id="automation">
+ <question>
+ How to automate scenario execution and collect pass / failed results
+ </question>
+ <answer>
+ <p>
+You don't need to modify SIPp sources to do that. All you need to do is run one
+call, exit (using -m command line option) and test the return code,
+as specified in SIPp help:<br/>
+Exit code:<br/>
+<br/>
+ Upon exit (on fatal error or when the number of asked calls (-m
+ option) is reached, sipp exits with one of the following exit
+ code:<br/>
+ 0: All calls were successful<br/>
+ 1: At least one call failed<br/>
+ 99: Normal exit without calls processed<br/>
+ -1: Fatal error<br/>
+<br/>
+A shell script could look like:<br/>
+<br/>
+ocadmin@idefix:~/OJA/sipp.2004-09-21$ cat run.sh<br/>
+#!/bin/sh<br/>
+>results.txt<br/>
+for i in $*<br/>
+do<br/>
+ echo Launching test $i >> results.txt<br/>
+ ./sipp -sf $i -m 1 127.0.0.1<br/>
+ if test $? -ne 0<br/>
+ then<br/>
+ echo Test $i failed >> results.txt<br/>
+ else<br/>
+ echo Test $i succeeded >> results.txt<br/>
+ fi<br/>
+done<br/>
+exit 0<br/>
+ocadmin@idefix:~/OJA/sipp.2004-09-21$<br/>
+<br/>
+Launch the tests: ./run.sh *.xml<br/>
+Then you will get a result file like this:<br/>
+<br/>
+ocadmin@idefix:~/OJA/sipp.2004-09-21$ cat results.txt<br/>
+Launching test regexp.xml<br/>
+Test regexp.xml failed<br/>
+Launching test uac.xml<br/>
+Test uac.xml succeeded<br/>
+ocadmin@idefix:~/OJA/sipp.2004-09-21$<br/>
+<br/>
+</p>
+ </answer>
+ </faq>
+ <!-- More faqs or parts here -->
+ </faqsection>
+ <faqsection id="docs">
+ <title>Documentation</title>
+ <faq id="forrest">
+ <question>
+ How can I help write documentation?
+ </question>
+ <answer>
+ <p>
+ This project uses <link href="ext:forrest">Apache Forrest</link> to
+ generate documentation from XML. Please download a copy of Forrest,
+ which can be used to <link
+ href="ext:forrest/validation">validate</link>, <link
+ href="ext:forrest/webapp">develop</link> and render a project site.
+ </p>
+ <p>Documentation sources (XML files) are available on request (oja...@us...).</p>
+ </answer>
+ </faq>
+ <!-- More faqs or parts here -->
+ </faqsection>
+ <!-- More faqs or parts here -->
+</faqs>
Added: doc/trunk/src/documentation/content/xdocs/doc3.0/index.xml
===================================================================
--- doc/trunk/src/documentation/content/xdocs/doc3.0/index.xml (rev 0)
+++ doc/trunk/src/documentation/content/xdocs/doc3.0/index.xml 2007-12-05 14:31:31 UTC (rev 372)
@@ -0,0 +1,15 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE document PUBLIC "-//APACHE//DTD Documentation V1.2//EN" "http://apache.org/forrest/dtd/document-v12.dtd">
+<document>
+ <header>
+ <title>SIPp documentation</title>
+ </header>
+ <body>
+ <p>This is the documentation section. You can find the following documents here:</p>
+ <ul>
+ <li><link href="reference.html">Reference documentation</link></li>
+ <li><link href="faq.html">FAQ</link></li>
+ <li><link href="license.html">License</link></li>
+ </ul>
+ </body>
+</document>
\ No newline at end of file
Added: doc/trunk/src/documentation/content/xdocs/doc3.0/license.xml
===================================================================
--- doc/trunk/src/documentation/content/xdocs/doc3.0/license.xml (rev 0)
+++ doc/trunk/src/documentation/content/xdocs/doc3.0/license.xml 2007-12-05 14:31:31 UTC (rev 372)
@@ -0,0 +1,334 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE document PUBLIC "-//APACHE//DTD Documentation V2.0//EN" "http://forrest.apache.org/dtd/document-v20.dtd">
+<document>
+ <header>
+ <title>License</title>
+ <subtitle>License for SIPp</subtitle>
+ </header>
+ <body>
+ <section>
+ <title>License</title>
+ <p>SIPp is under <a href="#gpl">GNU General Public License</a>.</p>
+ <p>send_packets.c and send_packets.h share source from tcpreplay
+tools by Aaron Turner
+<a href="http://tcpreplay.sourceforge.net/">http://tcpreplay.sourceforge.net/</a> and are under <a href="#bsd">BSD license</a></p>
+ </section>
+ <anchor id="gpl" /><section>
+ <title>GNU GENERAL PUBLIC LICENSE</title>
+ <source>
+ GNU GENERAL PUBLIC LICENSE
+ Version 2, June 1991
+
+ Copyright (C) 1989, 1991 Free Software Foundation, Inc.
+ 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ Everyone is permitted to copy and distribute verbatim copies
+ of this license document, but changing it is not allowed.
+
+ Preamble
+
+ The licenses for most software are designed to take away your
+freedom to share and change it. By contrast, the GNU General Public
+License is intended to guarantee your freedom to share and change free
+software--to make sure the software is free for all its users. This
+General Public License applies to most of the Free Software
+Foundation's software and to any other program whose authors commit to
+using it. (Some other Free Software Foundation software is covered by
+the GNU Library General Public License instead.) You can apply it to
+your programs, too.
+
+ When we speak of free software, we are referring to freedom, not
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+ Finally, any free program is threatened constantly by software
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+ NO WARRANTY
+
+ 11. BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE, THERE IS NO WARRANTY
+FOR THE PROGRAM, TO THE EXTENT PERMITTED BY APPLICABLE LAW. EXCEPT WHEN
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+
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+POSSIBILITY OF SUCH DAMAGES.
+
+ END OF TERMS AND CONDITIONS </source>
+ </section>
+ <anchor id="bsd" /><section>
+ <title>Copyright notice for send_packets.c and send_packets.h</title>
+ <source>
+ Copyright (c) 2001-2004 Aaron Turner.
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ 1. Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ 2. Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in the
+ documentation and/or other materials provided with the distribution.
+ 3. Neither the names of the copyright owners nor the names of its
+ contributors may be used to endorse or promote products derived from
+ this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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+ OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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+</source>
+ </section>
+ </body>
+</document>
+
Added: doc/trunk/src/documentation/content/xdocs/doc3.0/reference.xml
===================================================================
--- doc/trunk/src/documentation/content/xdocs/doc3.0/reference.xml (rev 0)
+++ doc/trunk/src/documentation/content/xdocs/doc3.0/reference.xml 2007-12-05 14:31:31 UTC (rev 372)
@@ -0,0 +1,2573 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE document PUBLIC "-//APACHE//DTD Documentation V2.0//EN" "http://forrest.apache.org/dtd/document-v20.dtd">
+<document>
+ <header>
+ <title>SIPp</title>
+ <subtitle>SIPp reference documentation</subtitle>
+ <authors>
+ <person name="Richard GAYRAUD [initial code]" email="ric...@us..."/>
+ <person name="Olivier JACQUES [code/documentation]" email="oja...@us..."/>
+ <person name="Many contributors [code]" email="no...@em..."/>
+ </authors>
+ </header>
+ <body>
+ <section>
+ <title>Foreword</title>
+ <p>SIPp is a performance testing tool 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
+ XML scenario files describing any performance testing configuration. 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, regular expressions and variables in scenario files, and
+ dynamically adjustable call rates.</p>
+ <p>SIPp can be used to test many real SIP equipements like SIP proxies,
+ B2BUAs, SIP media servers, SIP/x gateways, SIP PBX, ... It is also very
+ useful to emulate thousands of user agents calling your SIP system. </p>
+ <p><strong>Want to see it?</strong></p>
+ <p>Here is a screenshot</p>
+ <p><img src="images/sipp-01.jpg" alt="SIPp screenshot" /></p>
+ <p>And here is a video (Windows Media Player 9 codec or above required) of
+ SIPp in action:</p>
+ <p><icon src="images/wmv.gif" alt="wmv" /><a href="images/sipp-01.wmv">sipp-01.wmv</a></p>
+ </section>
+ <section><title>Installation</title>
+ <section>
+ <title>Getting SIPp</title>
+ <p>SIPp is released under the
+ <a href="http://www.gnu.org/copyleft/gpl.html">GNU GPL license</a>.
+ All the terms of the license apply. It is provided to the SIP
+ community by <a href="http://www.hp.com">Hewlett-Packard</a>
+ engineers in hope it can be useful.</p> <p>We receive some support
+ from our company to work on this tool freely, but <strong>HP does not
+ provide any support nor warranty concerning SIPp.</strong></p>
+ </section>
+ <section><title>Stable release</title>
+ <p>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 :) </p>
+ <note>Use the stable release for your everyday use and if you are
+ not blocked by a specific feature present in the "unstable release"
+ (see below).</note> <p><a
+ href="http://sourceforge.net/project/showfiles.php?group_id=104305">SIPp
+ stable download page</a></p>
+ </section>
+ <section><title>Unstable release</title>
+ <p>Unstable release: all new features and bug fixes are checked in
+ <a href="http://sipp.svn.sourceforge.net/viewvc/sipp/sipp/trunk/">SIPp's
+ SVN</a> repository as soon as they are available. Every night, an
+ automatic extraction is done and the source code of this release is
+ made available. </p>
+ <note> Use the unstable release if you absolutely need a bug fix or
+ a feature that is not in the stable release. </note>
+ <p><a href="http://sipp.sourceforge.net/snapshots/">SIPp "unstable" download page</a></p>
+ </section>
+ <section><title>Available platforms</title>
+ <p>SIPp is available on almost all UNIX platforms: HPUX, Tru64,
+ Linux (RedHat, Debian, FreeBSD), Solaris/SunOS.</p>
+ <p>A Windows port has been contributed. You can now compile SIPp under
+ Cygwin. A binary package with a Windows installer is also available.
+ Check <a href="http://sourceforge.net/project/showfiles.php?group_id=104305">the
+ download page</a> to download it and run SIPp under Windows.</p>
+ <note>SIPp works only over Windows XP and will not
+ work on Win2000. This is because of IPv6 support. The Windows installer
+ should prevent someone to install SIPp on Win2000.</note>
+ </section>
+ <anchor id="installing" /><section><title>Installing SIPp</title>
+ <ul>
+ <li>On Linux, SIPp is provided in the form of source code. You will need to
+ compile SIPp to actually use it.</li>
+ <li>Pre-requisites to compile SIPp are (see <a href="http://sipp.sourceforge.net/wiki/index.php/Compilation" >Compilation tips</a>):
+ <ul>
+ <li>C++ Compiler</li>
+ <li>curses or ncurses library</li>
+ <li>For authentication and TLS support: OpenSSL >= 0.9.8</li>
+ <li>For pcap play support: libpcap and libnet</li>
+ <li>For distributed pauses: <a href="http://www.gnu.org/software/gsl/">Gnu Scientific Libraries</a></li>
+ </ul>
+ </li>
+ <li>You have four options to compile SIPp:
+ <ul>
+ <li><strong>Without TLS (Transport Layer Security) and authentication support</strong>:
+ This is the recommended setup if you don't need to handle SIP authentication
+ and/or TLS. In this case, there are <strong>no depencies to install</strong> before
+ building SIPp. It is straight forward:
+<source># gunzip sipp-xxx.tar.gz
+# tar -xvf sipp-xxx.tar
+# cd sipp
+# make
+</source></li>
+ <li><strong>With TLS and <a href="#authentication">authentication</a> support</strong>, you must have
+ installed <a href="http://www.openssl.org/">OpenSSL library</a> (>=0.9.8)
+ (which may come with your system). Building SIPp consist only in adding
+ the "ossl" option to the make command:
+<source># gunzip sipp-xxx.tar.gz
+# tar -xvf sipp-xxx.tar
+# cd sipp
+# make ossl
+</source></li>
+ <li><strong>With <a href="#pcapplay">PCAP play</a> and without <a href="#authentication">authentication</a> support</strong>:
+<source># gunzip sipp-xxx.tar.gz
+# tar -xvf sipp-xxx.tar
+# cd sipp
+# make pcapplay
+</source></li>
+ <li><strong>With <a href="#pcapplay">PCAP play</a> and <a href="#authentication">authentication</a> support</strong>:
+<source># gunzip sipp-xxx.tar.gz
+# tar -xvf sipp-xxx.tar
+# cd sipp
+# make pcapplay_ossl
+</source></li>
+
+ </ul>
+ <anchor id="gsl"/><note>To enable <a href="http://www.gnu.org/software/gsl/">GSL</a> 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.</note>
+ </li>
+ <li>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 (<a href="http://win6.jp/Cygwin/">http://win6.jp/Cygwin/</a>),
+ as well as OpenSSL and libncurses.</warning></li>
+ <li>To compile SIPp on Windows with pcap (media support), you must:
+ <ul>
+ <li>Copy the <a href="http://www.winpcap.org/devel.htm">WinPcap developer package</a> to "C:\cygwin\lib\WpdPack"</li>
+ <li>Remove or rename "pthread.h" in "C:\cygwin\lib\WpdPack\Include", as it interfers with pthread.h from cygwin</li>
+ <li>Compile using either "make pcapplay_cygwin" or "pcapplay_ossl_cygwin"</li>
+ </ul>
+ </li>
+ </ul>
+ </section>
+ <anchor id="filedesc" /><section><title>Increasing File Descriptors Limit</title>
+ <p>If your system does not supports enough file descriptors,
+ you may experience problems when using the TCP/TLS mode with many simultaneous calls.</p>
+ <p>You have two ways to overcome this limit: either use the <a href="#maxsocket"><code>-max_socket</code></a>
+ command line option or change the limits of your system.</p>
+ <p>Depending on the operating system you use, different procedures
+ allow you to increase the maximum number of file descriptors:</p>
+ <ul>
+ <li><p>On Linux 2.4 kernels the default number of file descriptors can
+ be increased by modifying the <code>/etc/security/limits.conf</code>
+ and the <code>/etc/pam.d/login</code> file. </p>
+ <p>Open the <code>/etc/security/limits.conf</code> file and add the following lines:</p>
+<source>soft nofile 1024
+hard nofile 65535</source>
+ <p>Open the <code>/etc/pam.d/login</code> and add the following line</p>
+<source>session required /lib/security/pam_limits.so</source>
+ <p>The system file descriptor limit is set in the <code>/proc/sys/fs/file-max</code> file.
+ The following command will increase the file descriptor limit:</p>
+<source>echo 65535> /proc/sys/fs/file-max</source>
+ <p>To increase the number of file descriptors to its maximum limit
+ (65535) set in the <code>/etc/security/limits.conf</code> file, type:</p>
+<source>ulimit -n unlimited</source>
+ <p>Logout then login again to make the changes effective.</p>
+ </li>
+ <li><p>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:</p>
+<source>maxfiles : 4096
+maxfiles_lim : 4096
+nfiles : 4096
+ninode : 4096
+max_thread_proc : 4096
+nkthread : 4096</source>
+ </li>
+ </ul>
+ </section>
+ </section>
+ <section><title>Using SIPp</title>
+ <section>
+ <title>Main features</title>
+ <p>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.</p>
+ <p>Run sipp with embedded server (uas) scenario:</p>
+ <source># ./sipp -sn uas</source>
+ <p>On the same host, run sipp with embedded client (uac) scenario</p>
+ <source># ./sipp -sn uac 127.0.0.1</source>
+ </section>
+ <section>
+ <title>Integrated scenarios</title>
+ <p>Integrated scenarios? Yes, there are scenarios that are embedded
+ in SIPp executable. While you can create your own custom SIP
+ scenarios (see <a href="#xmlsyntax">how to create your own XML
+ scenarios</a>), a few basic (yet useful) scenarios are available
+ in SIPp executable.</p>
+ <section>
+ <title>UAC</title>
+ <p>Scenario file: <a href="uac.xml.html">uac.xml</a> (<a href="uac.xml">original XML file</a>)</p>
+ <source>SIPp UAC Remote
+ |(1) INVITE |
+ |------------------>|
+ |(2) 100 (optional) |
+ |<------------------|
+ |(3) 180 (optional) |
+ |<------------------|
+ |(4) 200 |
+ |<------------------|
+ |(5) ACK |
+ |------------------>|
+ | |
+ |(6) PAUSE |
+ | |
+ |(7) BYE |
+ |------------------>|
+ |(8) 200 |
+ |<------------------|
+</source>
+ </section>
+ <anchor id="uac_with_media"/><section>
+ <title>UAC with media</title>
+ <p>Scenario file: <a href="uac_pcap.xml.html">uac_pcap.xml</a> (<a href="uac_pcap.xml">original XML file</a>)</p>
+ <source>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 |
+ |<------------------|
+</source>
+ </section>
+ <section>
+ <title>UAS</title>
+ <p>Scenario file: <a href="uas.xml.html">uas.xml</a> (<a href="uas.xml">original XML file</a>)</p>
+ <source>Remote SIPp UAS
+ |(1) INVITE |
+ |------------------>|
+ |(2) 180 |
+ |<------------------|
+ |(3) 200 |
+ |<------------------|
+ |(4) ACK |
+ |------------------>|
+ | |
+ |(5) PAUSE |
+ | |
+ |(6) BYE |
+ |------------------>|
+ |(7) 200 |
+ |<------------------|
+</source>
+ </section>
+ <section>
+ <title>regexp</title>
+ <p>Scenario file: <a href="regexp.xml.html">regexp.xml</a> (<a href="regexp.xml">original XML file</a>)</p>
+ <p>This scenario, which behaves as an UAC is explained in greater details in <a href="#action_regexp">this section</a>.</p>
+ <source>SIPp regexp Remote
+ |(1) INVITE |
+ |------------------>|
+ |(2) 100 (optional) |
+ |<------------------|
+ |(3) 180 (optional) |
+ |<------------------|
+ |(4) 200 |
+ |<------------------|
+ |(5) ACK |
+ |------------------>|
+ | |
+ |(6) PAUSE |
+ | |
+ |(7) BYE |
+ |------------------>|
+ |(8) 200 |
+ |<------------------|
+</source>
+ </section>
+ <anchor id="scenario_branch" /><section>
+ <title>branch</title>
+ <p>Scenario files: <a href="branchc.xml.html">branchc.xml</a> (<a href="branchc.xml">original XML file</a>) and
+ <a href="branchs.xml.html">branchs.xml</a> (<a href="branchs.xml">original XML file</a>)</p>
+ <p>Those scenarios, which work against each other (branchc for client side and
+ branchs for server side) are explained in greater details in <a href="#branching">this section</a>.</p>
+ <source><![CDATA[ REGISTER ---------->
+ 200 <----------
+ 200 <----------
+ INVITE ---------->
+ 100 <----------
+ 180 <----------
+ 403 <----------
+ 200 <----------
+ ACK ---------->
+ [ 5000 ms]
+ BYE ---------->
+ 200 <----------]]></source>
+ </section>
+ <anchor id="ThreePCC" /><section>
+ <title>3PCC</title>
+ <p>3PCC stands for 3rd Party Call Control. 3PCC is described in
+ <a href="http://www.ietf.org/rfc/rfc3725.txt">RFC 3725</a>.
+ 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.</p>
+ <p>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):</p>
+ <source> A Controller B
+ |(1) INVITE no SDP | |
+ |<------------------| |
+ |(2) 200 offer1 | |
+ |------------------>| |
+ | |(3) INVITE offer1 |
+ | |------------------>|
+ | |(4) 200 OK answer1 |
+ | |<------------------|
+ | |(5) ACK |
+ | |------------------>|
+ |(6) ACK answer1 | |
+ |<------------------| |
+ |(7) RTP | |
+ |.......................................|
+</source>
+ <p>Scenario file: <a href="3pcc-A.xml.html">3pcc-A.xml</a> (<a href="3pcc-A.xml">original XML file</a>)</p>
+ <p>Scenario file: <a href="3pcc-B.xml.html">3pcc-B.xml</a> (<a href="3pcc-B.xml">original XML file</a>)</p>
+ <p>Scenario file: <a href="3pcc-C-A.xml.html">3pcc-C-A.xml</a> (<a href="3pcc-C-A.xml">original XML file</a>)</p>
+ <p>Scenario file: <a href="3pcc-C-B.xml.html">3pcc-C-B.xml</a> (<a href="3pcc-C-B.xml">original XML file</a>)</p>
+ <p>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).</p>
+ <p>The 3PCC call flow I will, in reality, look like this
+ (Controller has been divided in two SIPp instances):</p>
+ <source>
+ 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 | | |
+ |..........................................................|
+</source>
+ <p>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 <<a href="#sendCmd">sendCmd</a>> command. This:</p>
+<source><sendCmd>
+ <![CDATA[
+ Call-ID: [call_id]
+ [$1]
+
+ ]]>
+</sendCmd>
+</source>
+ <p>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:</p>
+<source><recvCmd>
+ <action
+ <ereg regexp="Content-Type:.*"
+ search_in="msg"
+ assign_to="2"/>
+ </action>
+</recvCmd>
+</source>
+ <p>Will receive a "command" from the twin SIPp instance.
+ Using the <a href="#action_regexp">regular expression</a> mechanism,
+ the content is retrieved
+ and stored in a call variable ($2 in this case), ready to be
+ reinjected</p>
+<source> <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>
+</source>
+ <p>In other words, <a href="#sendCmd">sendCmd</a> and <a href="#recvCmd">recvCmd</a> can be seen as synchronization points
+ between two SIPp instances, with the ability to pass parameters
+ between each other.</p>
+ <p>Another scenario that has been reported to be do-able with the
+ 3PCC feature is the following:</p>
+ <ul>
+ <li>A calls B. B answers. B and A converse</li>
+ <li>B calls C. C answers. C and B converse</li>
+ <li>B "REFER"s A to C and asks to replace A-B call with B-C call.</li>
+ <li>A accepts. A and C talk. B drops out of the calls.</li>
+ </ul>
+ </section>
+ </section>
+ <anchor id="ThreePCCExtended" /><section>
+ <title>3PCC Extended</title>
+ <p>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.</p>
+ <p>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:</p>
+<source>
+ s1;127.0.0.1:8080
+ s2;127.0.0.1:7080
+ m;127.0.0.1:6080
+</source>
+ <p>Each twin sipp instance must access a different copy of this file.</p>
+ <p><a href="#sendCmd">sendCmd</a> and <a href="#recvCmd">recvCmd</a> have additional attributes:</p>
+<source><sendCmd dest="s1">
+ <![CDATA[
+ Call-ID: [call_id]
+ From: m
+ [$1]
+
+ ]]>
+</sendCmd>
+</source>
+ <p>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. </p>
+<source><recvCmd src="m">
+ <action
+ <ereg regexp="Content-Type:.*"
+ search_in="msg"
+ assign_to="2"/>
+ </action>
+</recvCmd>
+</source>
+ <p>Indicates that the twin command is expected to be received from the "m" peer instance.</p>
+ <p>Note that the master must be the launched at last.</p>
+ <p>There is no integrated scenarios for the 3pcc extended mode, but you can easily adapt those from 3pcc.</p>
+ <p><b>Example:</b> 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.<br/><br/>
+ <img alt="Master / slave feature" src="images/master_slave.png"></img>
+ </p>
+ </section>
+ <anchor id="traffic_control" /><section><title>Traffic control</title>
+ <p>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: </p>
+ <ul>
+ <li>Interactively, by pressing keys on the keyboard
+ <ul>
+ <li>'+' key to increase call rate by 1</li>
+ <li>'-' key to decrease call rate by 1</li>
+ <li>'*' key to increase call rate by 10</li>
+ <li>'/' key to decrease call rate by 10</li>
+ </ul>
+ </li>
+ <li>At starting time, by specifying parameters on the command line:
+ <ul>
+ <li>"-r" to specify the call rate in number of calls per seconds</li>
+ <li>"-rp" to specify the "<strong>r</strong>ate <strong>p</strong>eriod"
+ in milliseconds for the call rate (default is 1000ms/1sec).
+ This allows you to have n calls every m milliseconds (by using <code>-r n -rp m</code>).
+ <note>Example: run SIPp at 7 calls every 2 seconds (3.5 calls per second)</note>
+ <source>./sipp -sn uac -r 7 -rp 2000 127.0.0.1</source>
+ </li>
+ </ul>
+ </li>
+ </ul>
+ <p>You can also <strong>pause</strong> 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 <strong>resume</strong> the traffic by pressing 'p' again.</p>
+ <p>To <strong>quit</strong> 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.</p>
+ <p>You can also force SIPp to <strong>quit</strong> 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.</p>
+ <note><strong>TIP:</strong> you can place a defined number of calls and
+ have SIPp exit when this is done. Use the <code>-m</code> option on the
+ command line.</note>
+ </section>
+ <anchor id="remote_control" /><section><title>Remote control</title>
+ <p>SIPp can be "remote-controlled" through a UDP socket. This allows for example</p>
+ <ul>
+ <li>To automate a series of actions, like increasing the call rate smoothly,
+ wait for 10 seconds, increase more, wait for 1 minute and loop</li>
+ <li>Have a feedback loop so that an application under test can
+ remote control SIPp to lower the load, pause the traffic, ...</li>
+ </ul>
+ <p>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, ...).</p>
+ <p>It is then possible to control SIPp like this:</p>
+ <source>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)</source>
+ <note>All keys available through keyboard are also available in
+ the remote control interface</note>
+ <p>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:</p>
+ <source>#!/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
+</source>
+ </section>
+ <section><title>Running SIPp in background</title>
+ <p>SIPp can be launched in background mode (<code>-bg</code> command
+ line option).</p>
+ <p>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 <code>-m</code> option, SIPp will run forever.</p>
+ <p>There is a mechanism implemented to stop SIPp smoothly. The command
+ <code>kill -SIGUSR1 [SIPp_PID]</code> will instruct SIPp to stop placing
+ any new calls and finish all ongoing calls before exiting.</p>
+ <p>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. </p>
+ </section>
+ <anchor id="xmlsyntax" /><section><title>Create your own XML scenarios</title>
+ <p>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:</p>
+ <source><?xml version="1.0" encoding="ISO-8859-1" ?>
+<scenario name="Basic Sipstone UAC">
+</source>
+ <p>And end with:</p>
+<source></scenario></source>
+ <p>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.</p>
+ <table>
+ <caption>List of commands with their attributes</caption>
+ <tr>
+ <th>Command</th>
+ <th>Attribute(s)</th>
+ <th>Description</th>
+ <th>Example</th>
+ </tr>
+ <tr>
+ <td><anchor id="send"/><strong><send></strong></td>
+ <td>retrans</td>
+ <td>Used for UDP transport only: it specifies the T1 timer value,
+ as described in SIP RFC 3261, section 17.1.1.2.</td>
+ <td><code><send retrans="500"></code>: will initiate T1 timer to 500 milliseconds (RFC3261 default).</td>
+ </tr>
+ <tr>
+ <td><anchor id="start_rtd"/></td>
+ <td>start_rtd</td>
+ <td>Starts one of the 5 "<strong>R</strong>esponse <strong>T</strong>ime <strong>D</strong>uration" timer.
+ (see <a href="#Response+times">statistics section</a>).</td>
+ <td><code><send start_rtd="2"></code>: the timer number 2 will start when the message is sent.</td>
+ </tr>
+ <tr>
+ <td><anchor id="rtd"/></td>
+ <td>rtd</td>
+ <td>Stops one of the 5 "<strong>R</strong>esponse <strong>T</strong>ime <strong>D</strong>uration" timer.</td>
+ <td><code><send rtd="2"></code>: the timer number 2 will stop when the message is sent.</td>
+ </tr>
+ <tr>
+ <td></td>
+ <td>repeat_rtd</td>
+ <td>Used with a rtd attribute, it allows the corresponding "<strong>R</strong>esponse <strong>T</strong>ime <strong>D</strong>uration" timer
+ to be counted more than once per call (useful for loop call flows). </td>
+ <td><code><send rtd="1" repeat_rtd="true"></code>: the timer number 1 value will be printed but the timer won't stop.</td>
+ </tr>
+ <tr>
+ <td></td>
+ <td>crlf</td>
+ <td>Displays an empty line <strong>after</strong> the arrow for the message in main SIPp screen.</td>
+ <td><code><send crlf="true"></code></td>
+ </tr>
+ <tr>
+ <td></td>
+ <td>lost</td>
+ <td>Emulate packet lost. The value is specified as a percentage.</td>
+ <td><code><send lost="10"></code>: 10% of the message sent are actually not sent :).</td>
+ </tr>
+ <tr>
+ <td></td>
+ <td>next</td>
+ <td>You can put a "next" in a send to go to another part of the script when you are done with sending the message.
+ See <a href="#branching">conditional branching</a> section for more info.</td>
+ <td>Example to jump to label "12" after sending an ACK:<source><![CDATA[ <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>
+</source></td>
+ </tr>
+ <tr>
+ <td></td>
+ <td>test</td>
+ <td>You can put a "test" next to a "next" attribute to indicate
+ that you only want to branch to the label specified with "next"
+ if the variable specified in "test" is set (through <a href="#action_regexp">regexp</a>
+ for example).
+ See <a href="#branching">conditional branching</a> section for more info.</td>
+ <td>Example to jump to label "6" after sending an ACK only if
+ variable 4 is set:<source><![CDATA[ <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>
+</source></td>
+ </tr>
+ <tr>
+ <td></td>
+ <td>counter</td>
+ <td>Increments the counter given as parameter when the message is sent. A total of 5 counter can be used.
+ The counter are saved in the <a href="#Available+counters">statistic file</a>.</td>
+ <td><code><send counter="1"></code>: Increments counter #1 when the message is sent.</td>
+ </tr>
+ <tr>
+ <td><anchor id="recv"/><strong><recv></strong></td>
+ <td>response</td>
+ <td>Indicates what SIP message code is expected.</td>
+ <td><code><recv response="200"></code>: SIPp will expect a SIP message with code "200".</td>
+ </tr>
+ <tr>
+ <td></td>
+ <td>request</td>
+ <td>Indicates what SIP message request is expected.</td>
+ <td><code><recv request="ACK"></code>: SIPp will expect an "ACK" SIP message.</td>
+ ...
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