[Pkg-voip-commits] r8480 - in /sip-tester/trunk/debian: changelog control rules sip-tester.manpages sipp.1

Sun Jun 13 05:57:52 UTC 2010

Author: msp
Date: Sun Jun 13 05:57:50 2010
New Revision: 8480

URL: http://svn.debian.org/wsvn/pkg-voip/?sc=1&rev=8480
Log:
Add resonable man page with help2man - sipp.1 (Closes: #581067)

Added:
    sip-tester/trunk/debian/sip-tester.manpages
    sip-tester/trunk/debian/sipp.1
Modified:
    sip-tester/trunk/debian/changelog
    sip-tester/trunk/debian/control
    sip-tester/trunk/debian/rules

Modified: sip-tester/trunk/debian/changelog
URL: http://svn.debian.org/wsvn/pkg-voip/sip-tester/trunk/debian/changelog?rev=8480&op=diff
==============================================================================

--- sip-tester/trunk/debian/changelog (original)
+++ sip-tester/trunk/debian/changelog Sun Jun 13 05:57:50 2010
@@ -15,8 +15,9 @@
   * Drop OpenSSL support - Build-Conflicts: libssl-dev
     - possible-gpl-code-linked-with-openssl
     - Fixes "GPL code linked with OpenSSL without permission" (Closes: #581069)
+  * Add resonable man page with help2man - sipp.1 (Closes: #581067)
 
- -- Mark Purcell <msp at debian.org>  Sun, 13 Jun 2010 15:37:50 +1000
+ -- Mark Purcell <msp at debian.org>  Sun, 13 Jun 2010 15:55:15 +1000
 
 sip-tester (3.1.r590-1) unstable; urgency=low
 

Modified: sip-tester/trunk/debian/control
URL: http://svn.debian.org/wsvn/pkg-voip/sip-tester/trunk/debian/control?rev=8480&op=diff
==============================================================================
--- sip-tester/trunk/debian/control (original)
+++ sip-tester/trunk/debian/control Sun Jun 13 05:57:50 2010
@@ -3,9 +3,9 @@
 Priority: optional
 Maintainer: Debian VoIP Team <pkg-voip-maintainers at lists.alioth.debian.org>
 Uploaders: Mark Purcell <msp at debian.org>, ARAKI Yasuhiro <ar at debian.org>, Paul Wise <pabs at debian.org>
-Build-Depends: debhelper (>= 4.0.0), libncurses5-dev, libnet1-dev, libpcap-dev
+Build-Depends: debhelper (>= 7), cdbs, libncurses5-dev, libnet1-dev, libpcap-dev
 Build-Conflicts: libssl-dev
-Standards-Version: 3.8.0
+Standards-Version: 3.8.4
 Homepage: http://sourceforge.net/projects/sipp/
 Vcs-Svn: svn://svn.debian.org/pkg-voip/sip-tester/trunk/
 Vcs-Browser: http://svn.debian.org/wsvn/pkg-voip/sip-tester/?op=log

Modified: sip-tester/trunk/debian/rules
URL: http://svn.debian.org/wsvn/pkg-voip/sip-tester/trunk/debian/rules?rev=8480&op=diff
==============================================================================
--- sip-tester/trunk/debian/rules (original)
+++ sip-tester/trunk/debian/rules Sun Jun 13 05:57:50 2010
@@ -3,3 +3,7 @@
 include /usr/share/cdbs/1/rules/debhelper.mk
 include /usr/share/cdbs/1/class/makefile.mk
 include /usr/share/cdbs/1/rules/simple-patchsys.mk
+
+help2man:
+	/usr/bin/help2man -v '-v' -N -S 'Debian GNU/Linux' -o debian/sipp.1 sipp \
+		-n 'Session Initiation Protol (SIP) performance testing tool'

Added: sip-tester/trunk/debian/sip-tester.manpages
URL: http://svn.debian.org/wsvn/pkg-voip/sip-tester/trunk/debian/sip-tester.manpages?rev=8480&op=file
==============================================================================
--- sip-tester/trunk/debian/sip-tester.manpages (added)
+++ sip-tester/trunk/debian/sip-tester.manpages Sun Jun 13 05:57:50 2010
@@ -1,0 +1,1 @@
+debian/sipp.1

Added: sip-tester/trunk/debian/sipp.1
URL: http://svn.debian.org/wsvn/pkg-voip/sip-tester/trunk/debian/sipp.1?rev=8480&op=file
==============================================================================
--- sip-tester/trunk/debian/sipp.1 (added)
+++ sip-tester/trunk/debian/sipp.1 Sun Jun 13 05:57:50 2010
@@ -1,0 +1,510 @@
+.\" DO NOT MODIFY THIS FILE!  It was generated by help2man 1.38.2.
+.TH SIPP "1" "June 2010" "Debian GNU/Linux" "User Commands"
+.SH NAME
+sipp \- Session Initiation Protol (SIP) performance testing tool
+.SH DESCRIPTION
+Usage:
+.IP
+sipp remote_host[:remote_port] [options]
+.IP
+Available options:
+.TP
+\fB\-v\fR
+: Display version and copyright information.
+.TP
+\fB\-aa\fR
+: Enable automatic 200 OK answer for INFO, UPDATE and
+NOTIFY messages.
+.TP
+\fB\-base_cseq\fR
+: Start value of [cseq] for each call.
+.TP
+\fB\-bg\fR
+: Launch SIPp in background mode.
+.TP
+\fB\-bind_local\fR
+: Bind socket to local IP address, i.e. the local IP
+address is used as the source IP address.  If SIPp runs
+in server mode it will only listen on the local IP
+address instead of all IP addresses.
+.TP
+\fB\-buff_size\fR
+: Set the send and receive buffer size.
+.TP
+\fB\-cid_str\fR
+: Call ID string (default %u\-%p@%s).  %u=call_number,
+%s=ip_address, %p=process_number, %%=% (in any order).
+.TP
+\fB\-ci\fR
+: Set the local control IP address
+.TP
+\fB\-cp\fR
+: Set the local control port number. Default is 8888.
+.TP
+\fB\-d\fR
+: Controls the length of calls. More precisely, this
+controls the duration of 'pause' instructions in the
+scenario, if they do not have a 'milliseconds' section.
+Default value is 0 and default unit is milliseconds.
+.TP
+\fB\-deadcall_wait\fR
+: How long the Call\-ID and final status of calls should be
+kept to improve message and error logs (default unit is
+ms).
+.TP
+\fB\-default_behaviors\fR: Set the default behaviors that SIPp will use.
+Possbile
+values are:
+\- all     Use all default behaviors
+\- none    Use no default behaviors
+\- bye     Send byes for aborted calls
+\- abortunexp      Abort calls on unexpected messages
+\- pingreply       Reply to ping requests
+If a behavior is prefaced with a \-, then it is turned
+off.  Example: all,\-bye
+.TP
+\fB\-f\fR
+: Set the statistics report frequency on screen. Default is
+1 and default unit is seconds.
+.TP
+\fB\-fd\fR
+: Set the statistics dump log report frequency. Default is
+60 and default unit is seconds.
+.TP
+\fB\-i\fR
+: Set the local IP address for 'Contact:','Via:', and
+\&'From:' headers. Default is primary host IP address.
+.TP
+\fB\-inf\fR
+: Inject values from an external CSV file during calls into
+the scenarios.
+First line of this file say whether the data is to be
+read in sequence (SEQUENTIAL), random (RANDOM), or user
+(USER) order.
+Each line corresponds to one call and has one or more
+\&';' delimited data fields. Those fields can be referred
+as [field0], [field1], ... in the xml scenario file.
+Several CSV files can be used simultaneously (syntax:
+\fB\-inf\fR f1.csv \fB\-inf\fR f2.csv ...)
+.TP
+\fB\-infindex\fR
+: file field
+Create an index of file using field.  For example \fB\-inf\fR
+users.csv \fB\-infindex\fR users.csv 0 creates an index on the
+first key.
+.TP
+\fB\-ip_field\fR
+: Set which field from the injection file contains the IP
+address from which the client will send its messages.
+If this option is omitted and the '\-t ui' option is
+present, then field 0 is assumed.
+Use this option together with '\-t ui'
+.TP
+\fB\-l\fR
+: Set the maximum number of simultaneous calls. Once this
+limit is reached, traffic is decreased until the number
+of open calls goes down. Default:
+.IP
+(3 * call_duration (s) * rate).
+.TP
+\fB\-lost\fR
+: Set the number of packets to lose by default (scenario
+specifications override this value).
+.TP
+\fB\-m\fR
+: Stop the test and exit when 'calls' calls are processed
+.TP
+\fB\-mi\fR
+: Set the local media IP address
+.TP
+\fB\-master\fR
+: 3pcc extended mode: indicates the master number
+.TP
+\fB\-max_recv_loops\fR
+: Set the maximum number of messages received read per
+cycle. Increase this value for high traffic level.  The
+default value is 1000.
+.TP
+\fB\-max_sched_loops\fR : Set the maximum number of calsl run per event loop.
+Increase this value for high traffic level.  The default
+value is 1000.
+.TP
+\fB\-max_reconnect\fR
+: Set the the maximum number of reconnection.
+.TP
+\fB\-max_retrans\fR
+: Maximum number of UDP retransmissions before call ends on
+timeout.  Default is 5 for INVITE transactions and 7 for
+others.
+.TP
+\fB\-max_invite_retrans\fR: Maximum number of UDP retransmissions for invite
+transactions before call ends on timeout.
+.TP
+\fB\-max_non_invite_retrans\fR: Maximum number of UDP retransmissions for non\-invite
+transactions before call ends on timeout.
+.TP
+\fB\-max_log_size\fR
+: What is the limit for error and message log file sizes.
+.TP
+\fB\-max_socket\fR
+: Set the max number of sockets to open simultaneously.
+This option is significant if you use one socket per
+call. Once this limit is reached, traffic is distributed
+over the sockets already opened. Default value is 50000
+.TP
+\fB\-mb\fR
+: Set the RTP echo buffer size (default: 2048).
+.TP
+\fB\-mp\fR
+: Set the local RTP echo port number. Default is 6000.
+.TP
+\fB\-nd\fR
+: No Default. Disable all default behavior of SIPp which
+are the following:
+\- On UDP retransmission timeout, abort the call by
+.IP
+sending a BYE or a CANCEL
+.IP
+\- On receive timeout with no ontimeout attribute, abort
+.IP
+the call by sending a BYE or a CANCEL
+.IP
+\- On unexpected BYE send a 200 OK and close the call
+\- On unexpected CANCEL send a 200 OK and close the call
+\- On unexpected PING send a 200 OK and continue the call
+\- On any other unexpected message, abort the call by
+.IP
+sending a BYE or a CANCEL
+.TP
+\fB\-nr\fR
+: Disable retransmission in UDP mode.
+.TP
+\fB\-nostdin\fR
+: Disable stdin.
+.TP
+\fB\-p\fR
+: Set the local port number.  Default is a random free port
+chosen by the system.
+.TP
+\fB\-pause_msg_ign\fR
+: Ignore the messages received during a pause defined in
+the scenario
+.TP
+\fB\-periodic_rtd\fR
+: Reset response time partition counters each logging
+interval.
+.TP
+\fB\-r\fR
+: Set the call rate (in calls per seconds).  This value can
+bechanged during test by pressing '+','_','*' or '/'.
+Default is 10.
+pressing '+' key to increase call rate by 1 *
+rate_scale,
+pressing '\-' key to decrease call rate by 1 *
+rate_scale,
+pressing '*' key to increase call rate by 10 *
+rate_scale,
+pressing '/' key to decrease call rate by 10 *
+rate_scale.
+If the \fB\-rp\fR option is used, the call rate is calculated
+with the period in ms given by the user.
+.TP
+\fB\-rp\fR
+: Specify the rate period for the call rate.  Default is 1
+second and default unit is milliseconds.  This allows
+you to have n calls every m milliseconds (by using \fB\-r\fR n
+\fB\-rp\fR m).
+Example: \fB\-r\fR 7 \fB\-rp\fR 2000 ==> 7 calls every 2 seconds.
+.IP
+\fB\-r\fR 10 \fB\-rp\fR 5s => 10 calls every 5 seconds.
+.TP
+\fB\-rate_scale\fR
+: Control the units for the '+', '\-', '*', and '/' keys.
+.TP
+\fB\-rate_increase\fR
+: Specify the rate increase every \fB\-fd\fR units (default is
+seconds).  This allows you to increase the load for each
+independent logging period.
+Example: \fB\-rate_increase\fR 10 \fB\-fd\fR 10s
+.IP
+==> increase calls by 10 every 10 seconds.
+.TP
+\fB\-rate_max\fR
+: If \fB\-rate_increase\fR is set, then quit after the rate
+reaches this value.
+Example: \fB\-rate_increase\fR 10 \fB\-rate_max\fR 100
+.IP
+==> increase calls by 10 until 100 cps is hit.
+.TP
+\fB\-no_rate_quit\fR
+: If \fB\-rate_increase\fR is set, do not quit after the rate
+reaches \fB\-rate_max\fR.
+.TP
+\fB\-recv_timeout\fR
+: Global receive timeout. Default unit is milliseconds. If
+the expected message is not received, the call times out
+and is aborted.
+.TP
+\fB\-send_timeout\fR
+: Global send timeout. Default unit is milliseconds. If a
+message is not sent (due to congestion), the call times
+out and is aborted.
+.HP
+\fB\-reconnect_close\fR : Should calls be closed on reconnect?
+.TP
+\fB\-reconnect_sleep\fR : How long (in milliseconds) to sleep between the close and
+reconnect?
+.TP
+\fB\-ringbuffer_files\fR: How many error/message files should be kept after
+rotation?
+.TP
+\fB\-ringbuffer_size\fR : How large should error/message files be before they get
+rotated?
+.TP
+\fB\-rsa\fR
+: Set the remote sending address to host:port for sending
+the messages.
+.TP
+\fB\-rtp_echo\fR
+: Enable RTP echo. RTP/UDP packets received on port defined
+by \fB\-mp\fR are echoed to their sender.
+RTP/UDP packets coming on this port + 2 are also echoed
+to their sender (used for sound and video echo).
+.TP
+\fB\-rtt_freq\fR
+: freq is mandatory. Dump response times every freq calls
+in the log file defined by \fB\-trace_rtt\fR. Default value is
+200.
+.TP
+\fB\-s\fR
+: Set the username part of the resquest URI. Default is
+\&'service'.
+.TP
+\fB\-sd\fR
+: Dumps a default scenario (embeded in the sipp executable)
+.TP
+\fB\-sf\fR
+: Loads an alternate xml scenario file.  To learn more
+about XML scenario syntax, use the \fB\-sd\fR option to dump
+embedded scenarios. They contain all the necessary help.
+.TP
+\fB\-oocsf\fR
+: Load out\-of\-call scenario.
+.TP
+\fB\-oocsn\fR
+: Load out\-of\-call scenario.
+.TP
+\fB\-skip_rlimit\fR
+: Do not perform rlimit tuning of file descriptor limits.
+Default: false.
+.TP
+\fB\-slave\fR
+: 3pcc extended mode: indicates the slave number
+.TP
+\fB\-slave_cfg\fR
+: 3pcc extended mode: indicates the file where the master
+and slave addresses are stored
+.TP
+\fB\-sn\fR
+: Use a default scenario (embedded in the sipp executable).
+If this option is omitted, the Standard SipStone UAC
+scenario is loaded.
+Available values in this version:
+.TP
+\- 'uac'
+: Standard SipStone UAC (default).
+.TP
+\- 'uas'
+: Simple UAS responder.
+.TP
+\- 'regexp'
+: Standard SipStone UAC \- with regexp and
+.IP
+variables.
+.TP
+\- 'branchc'
+: Branching and conditional branching in
+.IP
+scenarios \- client.
+.TP
+\- 'branchs'
+: Branching and conditional branching in
+.IP
+scenarios \- server.
+.IP
+Default 3pcc scenarios (see \fB\-3pcc\fR option):
+.IP
+\- '3pcc\-C\-A' : Controller A side (must be started after
+.IP
+all other 3pcc scenarios)
+.IP
+\- '3pcc\-C\-B' : Controller B side.
+\- '3pcc\-A'   : A side.
+\- '3pcc\-B'   : B side.
+.TP
+\fB\-stat_delimiter\fR
+: Set the delimiter for the statistics file
+.TP
+\fB\-stf\fR
+: Set the file name to use to dump statistics
+.TP
+\fB\-t\fR
+: Set the transport mode:
+\- u1: UDP with one socket (default),
+\- un: UDP with one socket per call,
+\- ui: UDP with one socket per IP address The IP
+.IP
+addresses must be defined in the injection file.
+.IP
+\- t1: TCP with one socket,
+\- tn: TCP with one socket per call,
+\- l1: TLS with one socket,
+\- ln: TLS with one socket per call,
+\- c1: u1 + compression (only if compression plugin
+.IP
+loaded),
+.IP
+\- cn: un + compression (only if compression plugin
+.TP
+loaded).
+This plugin is not provided with sipp.
+.TP
+\fB\-timeout\fR
+: Global timeout. Default unit is seconds.  If this option
+is set, SIPp quits after nb units (\fB\-timeout\fR 20s quits
+after 20 seconds).
+.TP
+\fB\-timer_resol\fR
+: Set the timer resolution. Default unit is milliseconds.
+This option has an impact on timers precision.Small
+values allow more precise scheduling but impacts CPU
+usage.If the compression is on, the value is set to
+50ms. The default value is 10ms.
+.TP
+\fB\-sendbuffer_warn\fR : Produce warnings instead of errors on SendBuffer
+failures.
+.TP
+\fB\-trace_msg\fR
+: Displays sent and received SIP messages in <scenario file
+name>_<pid>_messages.log
+.TP
+\fB\-trace_shortmsg\fR
+: Displays sent and received SIP messages as CSV in
+<scenario file name>_<pid>_shortmessages.log
+.TP
+\fB\-trace_screen\fR
+: Dump statistic screens in the
+<scenario_name>_<pid>_0ms.
+.TP
+\fB\-trace_err\fR
+: Trace all unexpected messages in <scenario file
+name>_<pid>_errors.log.
+.TP
+\fB\-trace_stat\fR
+: Dumps all statistics in <scenario_name>_<pid>.csv file.
+Use the '\-h stat' option for a detailed description of
+the statistics file content.
+.TP
+\fB\-trace_counts\fR
+: Dumps individual message counts in a CSV file.
+.TP
+\fB\-trace_rtt\fR
+: Allow tracing of all response times in <scenario file
+name>_<pid>_rtt.csv.
+.TP
+\fB\-trace_logs\fR
+: Allow tracing of <log> actions in <scenario file
+name>_<pid>_logs.log.
+.TP
+\fB\-users\fR
+: Instead of starting calls at a fixed rate, begin 'users'
+calls at startup, and keep the number of calls constant.
+.TP
+\fB\-3pcc\fR
+: Launch the tool in 3pcc mode ("Third Party call
+control"). The passed ip address is depending on the
+3PCC role.
+\- When the first twin command is 'sendCmd' then this is
+.TP
+the address of the remote twin socket.
+SIPp will try to
+.IP
+connect to this address:port to send the twin command
+(This instance must be started after all other 3PCC
+scenarii).
+.IP
+Example: 3PCC\-C\-A scenario.
+.IP
+\- When the first twin command is 'recvCmd' then this is
+.IP
+the address of the local twin socket. SIPp will open
+this address:port to listen for twin command.
+.IP
+Example: 3PCC\-C\-B scenario.
+.TP
+\fB\-tdmmap\fR
+: Generate and handle a table of TDM circuits.
+A circuit must be available for the call to be placed.
+Format: \fB\-tdmmap\fR {0\-3}{99}{5\-8}{1\-31}
+.TP
+\fB\-key\fR
+: keyword value
+Set the generic parameter named "keyword" to "value".
+.PP
+Signal handling:
+.IP
+SIPp can be controlled using posix signals. The following signals
+are handled:
+USR1: Similar to press 'q' keyboard key. It triggers a soft exit
+.IP
+of SIPp. No more new calls are placed and all ongoing calls
+are finished before SIPp exits.
+Example: kill \fB\-SIGUSR1\fR 732
+.IP
+USR2: Triggers a dump of all statistics screens in
+.IP
+<scenario_name>_<pid>_screens.log file. Especially useful
+in background mode to know what the current status is.
+Example: kill \fB\-SIGUSR2\fR 732
+.PP
+Exit code:
+.IP
+Upon exit (on fatal error or when the number of asked calls (\fB\-m\fR
+option) is reached, sipp exits with one of the following exit
+code:
+.IP
+0: All calls were successful
+1: At least one call failed
+.IP
+97: exit on internal command. Calls may have been processed
+99: Normal exit without calls processed
+\fB\-1\fR: Fatal error
+.PP
+Example:
+.IP
+Run sipp with embedded server (uas) scenario:
+.IP
+\&./sipp \fB\-sn\fR uas
+.IP
+On the same host, run sipp with embedded client (uac) scenario
+.IP
+\&./sipp \fB\-sn\fR uac 127.0.0.1
+.IP
+SIPp v3.1, version unknown, built Jun 13 2010, 15:34:03.
+.IP
+This program is free software; you can redistribute it and/or
+modify it under the terms of the GNU General Public License as
+published by the Free Software Foundation; either version 2 of
+the License, or (at your option) any later version.
+.IP
+This program is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+GNU General Public License for more details.
+.IP
+You should have received a copy of the GNU General Public
+License along with this program; if not, write to the
+Free Software Foundation, Inc.,
+59 Temple Place, Suite 330, Boston, MA  02111\-1307 USA
+.IP
+Author: see source files.


