Provided by: sip-tester_3.2-1build1_amd64 bug

NAME

       sipp - Session Initiation Protol (SIP) performance testing tool

DESCRIPTION

       Usage:

              sipp remote_host[:remote_port] [options]

              Available options:

       -v     : Display version and copyright information.

       -aa    : Enable automatic 200 OK answer for INFO, UPDATE and NOTIFY messages.

       -base_cseq
              : Start value of [cseq] for each call.

       -bg    : Launch SIPp in background mode.

       -bind_local
              :  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.

       -buff_size
              : Set the send and receive buffer size.

       -cid_str
              :    Call   ID   string   (default   %u-%p@%s).    %u=call_number,   %s=ip_address,
              %p=process_number, %%=% (in any order).

       -ci    : Set the local control IP address

       -cp    : Set the local control port number. Default is 8888.

       -d     : 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.

       -deadcall_wait
              : How long the Call-ID and final status of calls should be kept to improve  message
              and error logs (default unit is ms).

       -default_behaviors: 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

       -f     : Set the statistics report frequency on screen. Default is 1 and default  unit  is
              seconds.

       -fd    :  Set  the statistics dump log report frequency. Default is 60 and default unit is
              seconds.

       -i     : Set the local IP address for 'Contact:','Via:', and 'From:' headers.  Default  is
              primary host IP address.

       -inf   :  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: -inf f1.csv -inf f2.csv ...)

       -infindex
              : file field Create an index of file  using  field.   For  example  -inf  users.csv
              -infindex users.csv 0 creates an index on the first key.

       -ip_field
              :  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'

       -l     : Set the maximum number of simultaneous calls. Once this limit is reached, traffic
              is decreased until the number of open calls goes down. Default:

              (3 * call_duration (s) * rate).

       -lost  : Set the number of packets to lose by default  (scenario  specifications  override
              this value).

       -m     : Stop the test and exit when 'calls' calls are processed

       -mi    : Set the local media IP address

       -master
              : 3pcc extended mode: indicates the master number

       -max_recv_loops
              :  Set  the maximum number of messages received read per cycle. Increase this value
              for high traffic level.  The default value is 1000.

       -max_sched_loops : Set the maximum number of calsl run per event loop.
              Increase this value for high traffic level.  The default value is 1000.

       -max_reconnect
              : Set the the maximum number of reconnection.

       -max_retrans
              : Maximum number of UDP retransmissions before call ends on timeout.  Default is  5
              for INVITE transactions and 7 for others.

       -max_invite_retrans: Maximum number of UDP retransmissions for invite
              transactions before call ends on timeout.

       -max_non_invite_retrans: Maximum number of UDP retransmissions for non-invite
              transactions before call ends on timeout.

       -max_log_size
              : What is the limit for error and message log file sizes.

       -max_socket
              : 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

       -mb    : Set the RTP echo buffer size (default: 2048).

       -mp    : Set the local RTP echo port number. Default is 6000.

       -nd    :  No  Default.  Disable all default behavior of SIPp which are the following: - On
              UDP retransmission timeout, abort the call by

              sending a BYE or a CANCEL

              - On receive timeout with no ontimeout attribute, abort

              the call by sending a BYE or a CANCEL

              - 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

              sending a BYE or a CANCEL

       -nr    : Disable retransmission in UDP mode.

       -nostdin
              : Disable stdin.

       -p     : Set the local port number.  Default is a random free port chosen by the system.

       -pause_msg_ign
              : Ignore the messages received during a pause defined in the scenario

       -periodic_rtd
              : Reset response time partition counters each logging interval.

       -r     : 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 -rp option is used, the call rate is
              calculated with the period in ms given by the user.

       -rp    :  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 -r
              n -rp m).  Example: -r 7 -rp 2000 ==> 7 calls every 2 seconds.

              -r 10 -rp 5s => 10 calls every 5 seconds.

       -rate_scale
              : Control the units for the '+', '-', '*', and '/' keys.

       -rate_increase
              :  Specify the rate increase every -fd units (default is seconds).  This allows you
              to increase the load for each independent logging period.  Example:  -rate_increase
              10 -fd 10s

              ==> increase calls by 10 every 10 seconds.

       -rate_max
              :  If -rate_increase is set, then quit after the rate reaches this value.  Example:
              -rate_increase 10 -rate_max 100

              ==> increase calls by 10 until 100 cps is hit.

       -no_rate_quit
              : If -rate_increase is set, do not quit after the rate reaches -rate_max.

       -recv_timeout
              : Global receive timeout. Default unit is milliseconds. If the expected message  is
              not received, the call times out and is aborted.

       -send_timeout
              :  Global send timeout. Default unit is milliseconds. If a message is not sent (due
              to congestion), the call times out and is aborted.

       -reconnect_close : Should calls be closed on reconnect?

       -reconnect_sleep : How long (in milliseconds) to sleep between the close and
              reconnect?

       -ringbuffer_files: How many error/message files should be kept after
              rotation?

       -ringbuffer_size : How large should error/message files be before they get
              rotated?

       -rsa   : Set the remote sending address to host:port for sending the messages.

       -rtp_echo
              : Enable RTP echo. RTP/UDP packets received on port defined by -mp  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).

       -rtt_freq
              : freq is mandatory. Dump response times every freq calls in the log  file  defined
              by -trace_rtt. Default value is 200.

       -s     : Set the username part of the resquest URI. Default is 'service'.

       -sd    : Dumps a default scenario (embeded in the sipp executable)

       -sf    :  Loads  an alternate xml scenario file.  To learn more about XML scenario syntax,
              use the -sd option to dump embedded scenarios. They contain all the necessary help.

       -oocsf : Load out-of-call scenario.

       -oocsn : Load out-of-call scenario.

       -skip_rlimit
              : Do not perform rlimit tuning of file descriptor limits.  Default: false.

       -slave : 3pcc extended mode: indicates the slave number

       -slave_cfg
              : 3pcc extended mode: indicates the file where the master and slave  addresses  are
              stored

       -sn    :  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:

       - 'uac'
              : Standard SipStone UAC (default).

       - 'uas'
              : Simple UAS responder.

       - 'regexp'
              : Standard SipStone UAC - with regexp and

              variables.

       - 'branchc'
              : Branching and conditional branching in

              scenarios - client.

       - 'branchs'
              : Branching and conditional branching in

              scenarios - server.

              Default 3pcc scenarios (see -3pcc option):

              - '3pcc-C-A' : Controller A side (must be started after

              all other 3pcc scenarios)

              - '3pcc-C-B' : Controller B side.  - '3pcc-A'   : A side.  - '3pcc-B'   : B side.

       -stat_delimiter
              : Set the delimiter for the statistics file

       -stf   : Set the file name to use to dump statistics

       -t     :  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

              addresses must be defined in the injection file.

              - 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

              loaded),

              - cn: un + compression (only if compression plugin

       loaded).
              This plugin is not provided with sipp.

       -timeout
              : Global timeout. Default unit is seconds.  If this option is set, SIPp quits after
              nb units (-timeout 20s quits after 20 seconds).

       -timer_resol
              :  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.

       -sendbuffer_warn : Produce warnings instead of errors on SendBuffer
              failures.

       -trace_msg
              :   Displays   sent   and    received    SIP    messages    in    <scenario    file
              name>_<pid>_messages.log

       -trace_shortmsg
              :   Displays   sent   and   received   SIP   messages  as  CSV  in  <scenario  file
              name>_<pid>_shortmessages.log

       -trace_screen
              : Dump statistic screens in the <scenario_name>_<pid>_0ms.

       -trace_err
              : Trace all unexpected messages in <scenario file name>_<pid>_errors.log.

       -trace_stat
              : Dumps all statistics in <scenario_name>_<pid>.csv file.  Use the '-h stat' option
              for a detailed description of the statistics file content.

       -trace_counts
              : Dumps individual message counts in a CSV file.

       -trace_rtt
              : Allow tracing of all response times in <scenario file name>_<pid>_rtt.csv.

       -trace_logs
              : Allow tracing of <log> actions in <scenario file name>_<pid>_logs.log.

       -users :  Instead  of  starting calls at a fixed rate, begin 'users' calls at startup, and
              keep the number of calls constant.

       -3pcc  : 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

       the address of the remote twin socket.
              SIPp will try to

              connect to this address:port to send  the  twin  command  (This  instance  must  be
              started after all other 3PCC scenarii).

              Example: 3PCC-C-A scenario.

              - When the first twin command is 'recvCmd' then this is

              the  address  of  the local twin socket. SIPp will open this address:port to listen
              for twin command.

              Example: 3PCC-C-B scenario.

       -tdmmap
              : Generate and handle a table of TDM circuits.  A circuit must be available for the
              call to be placed.  Format: -tdmmap {0-3}{99}{5-8}{1-31}

       -key   : keyword value Set the generic parameter named "keyword" to "value".

       Signal handling:

              SIPp  can  be  controlled  using  posix signals. The following signals are handled:
              USR1: Similar to press 'q' keyboard key. It triggers a soft exit

              of SIPp. No more new calls are placed and all ongoing  calls  are  finished  before
              SIPp exits.  Example: kill -SIGUSR1 732

              USR2: Triggers a dump of all statistics screens in

              <scenario_name>_<pid>_screens.log  file.  Especially  useful  in background mode to
              know what the current status is.  Example: kill -SIGUSR2 732

       Exit code:

              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:

              0: All calls were successful 1: At least one call failed

              97: exit on internal command. Calls may have been processed 99: Normal exit without
              calls processed -1: Fatal error

       Example:

              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

              SIPp v3.1, version unknown, built Jun 13 2010, 15:34:03.

              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.

              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.

              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

              Author: see source files.