Checked-in! http://sipp.sourceforge.net/doc1.1/reference.html
Thanks Martin, this is the first patch to documentation. Olivier. Martin Mathieson wrote: > Hi, > > Please find attached a patch to xdocs/doc1.1/reference.xml. Many of > the changes are just pedantic English, but I also mentioned: > - SVN repo now used rather than CVS (please check path!) > - note that sipp.dtd exists (at least for jEdit) whereas it said there > is no help available > - "ethereal" -> "wireshark" > > Regards, > Martin > ------------------------------------------------------------------------ > > Index: reference.xml > =================================================================== > --- reference.xml (revision 71) > +++ reference.xml (working copy) > @@ -49,7 +49,7 @@ > <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 mentionned features will work :) </p> > + 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 > @@ -57,9 +57,9 @@ > stable download page</a></p> > </section> > <section><title>Unstable release</title> > - <p>Unstable release: every new features and bug fixes are > checked in > - <a href="http://cvs.sourceforge.net/viewcvs.py/sipp/sipp/";>SIPp's > - CVS</a> repository as soon as they are available. Every night, an > + <p>Unstable release: all new features and bug fixes are checked > in > + <a > href="https://svn.sourceforge.net/svnroot/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 > @@ -136,7 +136,7 @@ > <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 to increase the maximum number of file descriptors:</p> > + 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> > @@ -459,7 +459,7 @@ > <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 serie of action, like increasing the call rate > smoothly, > + <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> > @@ -492,7 +492,7 @@ > <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 background. The PID of the SIPp process is provided. If you > didn't specified a number of calls to execute > + 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 > @@ -501,8 +501,9 @@ > <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 > - (although there are currently no DTD to help you write SIPp > - scenarios). A scenario will always start with:</p> > + (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> > @@ -1214,15 +1215,15 @@ > <li><strong>Internal</strong> commands (specified using > int_cmd attribute) are stop_call, stop_gracefully (similar to pressing 'q'), > stop_now (similar to ctrl+C).</li> > <li><strong>External</strong> commands (specified using > command attribute) are anything that can be executed on local host with a > shell.</li> > <li><strong>PCAP play</strong> commands (specified using > play_pcap_audio / play_pcap_video attributes) > - allow to send a pre-recorded RTP stream using the <a > href="http://www.tcpdump.org/pcap3_man.html";>pcap library</a>. > + allow you to send a pre-recorded RTP stream using the <a > href="http://www.tcpdump.org/pcap3_man.html";>pcap library</a>. > <p>Choose <strong>play_pcap_audio</strong> to send the > pre-recorded RTP stream using the "m=audio" SIP/SDP line port as a base for > the replay.</p> > <p>Choose <strong>play_pcap_video</strong> to send the > pre-recorded RTP stream using the "m=video" SIP/SDP line port as a base.</p> > <p>The play_pcap_audio/video command has the following > format: play_pcap_audio="[file_to_play]" with:</p> > <ul> > <li>file_to_play: the pre-recorded pcap file to > play</li> > </ul> > - <note>The action is non-blocking. SIPp will start > a light-weighted thread to play the file > - and the scenario with continue immediatly. If > needed, you will need to add a pause > + <note>The action is non-blocking. SIPp will start > a light-weight thread to play the file > + and the scenario with continue immediately. If > needed, you will need to add a pause > to wait for the end of the pcap play.</note> > </li> > </ul> > @@ -1399,7 +1400,7 @@ > </section> > <section><title>Screens</title> > <p>Several screens are available to monitor SIP traffic. You can > - change of screen by pressing 1, 2, 3 or 4 keys on the > keyboard.</p> > + change the screen view by pressing 1, 2, 3 or 4 keys on the > keyboard.</p> > <ul> > <li>Key '1': Scenario screen. It displays a call flow of > the scenario as well as some important informations. > @@ -1474,7 +1475,7 @@ > </source> > </section> > <anchor id="maxsocket" /><section><title>Multi-socket > limit</title> > - <p>When using one of the "multi-socket" transport, the > maximum number of sockets that can be opened > + <p>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 <a href="#filedesc">how to increase file > descriptors section</a> to > modify those limits). You can also limit the number of > socket used by using the <code>-max_socket</code> > @@ -1498,15 +1499,15 @@ > <anchor id="pcapplay" /><section><title>PCAP Play</title> > <p>The PCAP play feature makes use of the <a > href="http://www.tcpdump.org/pcap3_man.html";>PCAP library</a> > to replay pre-recorded RTP streams towards a destination. RTP > streams can be > - recorded by tools like <a > href="http://www.ethereal.com/";>Ethereal</a> > - or <a href="http://www.tcpdump.org/";>tcpdump</a>. This allows > to:</p> > + recorded by tools like <a > href="http://www.wireshark.org/";>Wireshark</a> > + (formerly known as Ethereal) or <a > href="http://www.tcpdump.org/";>tcpdump</a>. This allows you to:</p> > <ul> > <li>Play any RTP stream (voice, video, voice+video, out of > band DTMFs/RFC 2833, T38 fax, ...)</li> > <li>Use any codec as the codec is not handled by SIPp</li> > <li>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).</li> > - <li>Reproduce exactly what has been captured using an IP > sniffer like <a href="http://www.ethereal.com/";>Ethereal</a>.</li> > + <li>Reproduce exactly what has been captured using an IP > sniffer like <a href="http://www.wireshark.org/";>Wireshark</a>.</li> > </ul> > <p>A good example is the <a href="#uac_with_media">UAC with > media</a> (uac_pcap) embedded scenario.</p> > <p>SIPp comes with a G711 alaw pre-recorded pcap file and > @@ -1595,7 +1596,7 @@ > option but no matching header found.</li> > <li>OutOfCallMsgs: > Number of SIP messages received that cannot be associated > - to an existing call.</li> > + with an existing call.</li> > <li>AutoAnswered: > Number of unexpected specific messages received for new Call-ID. > The message has been automatically answered by a 200 OK > @@ -1992,7 +1993,7 @@ > (as described in "<a href="#scheduling">SIPp's internal > scheduling</a>" > section). You might want to use another "objective" method if > you want to measure those response times with a high precision (a > tool > - like <a href="http://www.ethereal.com/";>Ethereal</a> will allow you > to do so).</p> > + like <a href="http://www.wireshark.org/";>Wireshark</a> will allow > you to do so).</p> > </section> > <anchor id="scheduling" /><section><title>SIPp's internal > scheduling</title> > <p>Three parameters can be set to allow SIPp to benefit of the > @@ -2045,14 +2046,14 @@ > (<a href="http://sipp.sourceforge.net/doc/sipp.dtd";>sipp.dtd</a>) > in the same directory as your XML scenario.</p> > </section> > - <section><title>Ethereal/tethereal</title> > - <p>Ethereal (<a > href="http://www.ethereal.com/";>http://www.ethereal.com/</a>) is a > - GNU GPL protocol analyzer. It supports SIP.</p> > + <section><title>Wireshark/tshark</title> > + <p>Wireshark (<a > href="http://www.wireshark.org/";>http://www.wireshark.org/</a>) is a > + GNU GPL protocol analyzer. It was formerly known as Ethereal. It > supports SIP/SDP/RTP.</p> > <p></p> > </section> > <section><title>SIP callflow</title> > <p>When tracing SIP calls, it is very useful to be able > - to get a call flow from an ethereal trace. The "callflow" tool > allows you to do > + to get a call flow from an wireshark trace. The "callflow" tool > allows you to do > that in a graphical way: > <a > href="http://callflow.sourceforge.net/";>http://callflow.sourceforge.net/</a></p> > <p>An equivalent exist if you want to generate HTML only call > flows > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Using Tomcat but need to do more? Need to support web services, security? > Get stuff done quickly with pre-integrated technology to make your job easier > Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo > http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 > ------------------------------------------------------------------------ > > _______________________________________________ > Sipp-users mailing list > [email protected] > https://lists.sourceforge.net/lists/listinfo/sipp-users > -- Olivier HP OpenCall Software http://www.hp.com/go/opencall/ ------------------------------------------------------------------------- Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 _______________________________________________ Sipp-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/sipp-users
