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
