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SIPP(1)				 User Commands			       SIPP(1)

NAME
       sipp - manual page for sipp

DESCRIPTION
       Usage:

	      sipp remote_host[:remote_port] [options]

       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

	      Available options:

       *** Scenario file options:

       -sd    : Dumps a default scenario (embedded 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.

       -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.

       *** IP, port and protocol options:

       -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, -
	      c1: u1 + compression (only if compression plugin loaded), -  cn:
	      un  +  compression  (only	 if  compression plugin loaded).  This
	      plugin is

	      not provided with SIPp.

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

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

       -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.

       -ci    : Set the local control IP address

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

       -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

       -max_reconnect
	      : Set the the maximum number of reconnection.

       -reconnect_close : Should calls be closed on reconnect?

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

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

       *** SIPp overall behavior options:

       -v     : Display version and copyright information.

       -bg    : Launch SIPp in background mode.

       -nostdin
	      : Disable stdin.

       -plugin
	      : Load a plugin.

       -sleep : How long to sleep for at startup. Default unit is seconds.

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

       -buff_size
	      : Set the send and receive buffer size.

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

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

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

       -set   : variable value Set the global variable parameter named	"vari‐
	      able" to "value".

       -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}

       -dynamicStart
	      : variable value Set the start offset of dynamic_id variable

       -dynamicMax
	      : variable value Set the maximum of dynamic_id variable

       -dynamicStep
	      : variable value Set the increment of dynamic_id variable

       *** Call behavior options:

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

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

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

       -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).

       -auth_uri
	      :	 Force	the  value of the URI for authentication.  By default,
	      the URI is composed of remote_ip:remote_port.

       -au    : Set  authorization  username  for  authentication  challenges.
	      Default is taken from -s argument

       -ap    :	 Set  the  password  for authentication challenges. Default is
	      'password'

       -s     : Set the username part of the request  URI.  Default  is	 'ser‐
	      vice'.

       -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

       -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	 unex‐
	      pected  CANCEL  send a 200 OK and close the call - On unexpected
	      PING send a 200 OK and continue the call - On  any  other	 unex‐
	      pected message, abort the call by sending a BYE or a

	      CANCEL

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

       *** Injection file options:

       -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'

       *** RTP behaviour options:

       -mi    : Set the local media IP address (default: local primary host IP
	      address)

       -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).

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

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

       *** Call rate options:

       -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, press‐
	      ing '*' key to increase call rate by 10 *	 rate_scale,  pressing
	      '/' key to decrease call rate by 10 * rate_scale.

       -rp    :	 Specify the rate period for the call rate.  Default is 1 sec‐
	      ond 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  sec‐
	      onds).   This  allows you to increase the load for each indepen‐
	      dent 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.

       -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).

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

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

       *** Retransmission and timeout options:

       -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 mes‐
	      sage is not sent (due to congestion), the call times out and  is
	      aborted.

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

       -timeout_error
	      :	 SIPp  fails if the global timeout is reached is set (-timeout
	      option required).

       -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.

       -nr    : Disable retransmission in UDP mode.

       -rtcheck
	      :	 Select the retransmission detection method: full (default) or
	      loose.

       -T2    : Global T2-timer in milli seconds

       *** Third-party call control options:

       -3pcc  : Launch the tool in 3pcc mode ("Third Party call control"). The
	      passed  IP  address  depends 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 scenarios).

	      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.

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

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

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

       *** Performance and watchdog options:

       -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.

       -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 calls run per event loop.
       Increase this value for
	      high traffic level.  The default value is 1000.

       -watchdog_interval: Set gap between watchdog timer firings.
	      Default is 400.

       -watchdog_reset
	      : If the watchdog timer has not fired in	more  than  this  time
	      period,  then  reset  the	 max triggers counters.	 Default is 10
	      minutes.

       -watchdog_minor_threshold: If it	 has  been  longer  than  this	period
       between watchdog executions count a
	      minor trip.  Default is 500.

       -watchdog_major_threshold:  If  it  has	been  longer  than this period
       between watchdog executions count a
	      major trip.  Default is 3000.

       -watchdog_major_maxtriggers: How many times the	major  watchdog	 timer
       can be tripped before the test is
	      terminated.  Default is 10.

       -watchdog_minor_maxtriggers:  How  many	times the minor watchdog timer
       can be tripped before the test is
	      terminated.  Default is 120.

       *** Tracing, logging and statistics options:

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

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

       -stat_delimiter
	      : Set the delimiter for the statistics file

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

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

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

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

       -message_file
	      : Set the name of the message log file.

       -message_overwrite: Overwrite the message log file (default true).

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

       -shortmessage_file: Set the name of the short message log file.

       -shortmessage_overwrite: Overwrite the short message log file  (default
	      true).

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

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

       -error_file
	      : Set the name of the error log file.

       -error_overwrite : Overwrite the error log file (default true).

       -trace_error_codes: Dumps the SIP response codes of unexpected messages
       to <scenario file
	      name>_<pid>_error_codes.log.

       -trace_calldebug : Dumps debugging information about aborted calls to
	      <scenario_name>_<pid>_calldebug.log file.

       -calldebug_file
	      : Set the name of the call debug file.

       -calldebug_overwrite: Overwrite the call debug file (default true).

       -trace_screen
	      :	     Dump      statistic      screens	  in	 the	 <sce‐
	      nario_name>_<pid>_screens.log file when quitting SIPp. Useful to
	      get a final status report in background mode (-bg option).

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

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

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

       -log_file
	      : Set the name of the log actions log file.

       -log_overwrite
	      : Overwrite the log actions log file (default true).

       -ringbuffer_files: How many error, message, shortmessage and  calldebug
       files should be kept
	      after rotation?

       -ringbuffer_size	 :  How	 large should error, message, shortmessage and
       calldebug files be before
	      they get rotated?

       -max_log_size
	      : What is the limit for error, message, shortmessage and callde‐
	      bug file sizes.

       Signal handling:

	      SIPp  can	 be controlled using POSIX signals. The following sig‐
	      nals are handled: USR1: Similar to  pressing  the	 'q'  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 codes:

	      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 -2: Fatal
	      error binding a socket

       Usage:

	      sipp remote_host[:remote_port] [options]

       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

	      Available options:

       *** Scenario file options:

       -sd    : Dumps a default scenario (embedded 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.

       -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.

       *** IP, port and protocol options:

       -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, -
	      c1:  u1 + compression (only if compression plugin loaded), - cn:
	      un + compression (only  if  compression  plugin  loaded).	  This
	      plugin is

	      not provided with SIPp.

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

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

       -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.

       -ci    : Set the local control IP address

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

       -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

       -max_reconnect
	      : Set the the maximum number of reconnection.

       -reconnect_close : Should calls be closed on reconnect?

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

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

       *** SIPp overall behavior options:

       -v     : Display version and copyright information.

       -bg    : Launch SIPp in background mode.

       -nostdin
	      : Disable stdin.

       -plugin
	      : Load a plugin.

       -sleep : How long to sleep for at startup. Default unit is seconds.

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

       -buff_size
	      : Set the send and receive buffer size.

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

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

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

       -set   :	 variable value Set the global variable parameter named "vari‐
	      able" to "value".

       -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}

       -dynamicStart
	      : variable value Set the start offset of dynamic_id variable

       -dynamicMax
	      : variable value Set the maximum of dynamic_id variable

       -dynamicStep
	      : variable value Set the increment of dynamic_id variable

       *** Call behavior options:

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

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

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

       -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).

       -auth_uri
	      : Force the value of the URI for	authentication.	  By  default,
	      the URI is composed of remote_ip:remote_port.

       -au    :	 Set  authorization  username  for  authentication challenges.
	      Default is taken from -s argument

       -ap    : Set the password for  authentication  challenges.  Default  is
	      'password'

       -s     :	 Set  the  username  part of the request URI. Default is 'ser‐
	      vice'.

       -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

       -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 unex‐
	      pected CANCEL send a 200 OK and close the call -	On  unexpected
	      PING  send  a  200 OK and continue the call - On any other unex‐
	      pected message, abort the call by sending a BYE or a

	      CANCEL

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

       *** Injection file options:

       -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'

       *** RTP behaviour options:

       -mi    : Set the local media IP address (default: local primary host IP
	      address)

       -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).

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

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

       *** Call rate options:

       -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, press‐
	      ing  '*'	key to increase call rate by 10 * rate_scale, pressing
	      '/' key to decrease call rate by 10 * rate_scale.

       -rp    : Specify the rate period for the call rate.  Default is 1  sec‐
	      ond 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 sec‐
	      onds).  This allows you to increase the load for	each  indepen‐
	      dent 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.

       -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).

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

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

       *** Retransmission and timeout options:

       -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  mes‐
	      sage  is not sent (due to congestion), the call times out and is
	      aborted.

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

       -timeout_error
	      : SIPp fails if the global timeout is reached is	set  (-timeout
	      option required).

       -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.

       -nr    : Disable retransmission in UDP mode.

       -rtcheck
	      : Select the retransmission detection method: full (default)  or
	      loose.

       -T2    : Global T2-timer in milli seconds

       *** Third-party call control options:

       -3pcc  : Launch the tool in 3pcc mode ("Third Party call control"). The
	      passed IP address depends 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 scenarios).

	      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.

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

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

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

       *** Performance and watchdog options:

       -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.

       -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 calls run per event	 loop.
       Increase this value for
	      high traffic level.  The default value is 1000.

       -watchdog_interval: Set gap between watchdog timer firings.
	      Default is 400.

       -watchdog_reset
	      :	 If  the  watchdog  timer has not fired in more than this time
	      period, then reset the max triggers  counters.   Default	is  10
	      minutes.

       -watchdog_minor_threshold:  If  it  has	been  longer  than this period
       between watchdog executions count a
	      minor trip.  Default is 500.

       -watchdog_major_threshold: If it	 has  been  longer  than  this	period
       between watchdog executions count a
	      major trip.  Default is 3000.

       -watchdog_major_maxtriggers:  How  many	times the major watchdog timer
       can be tripped before the test is
	      terminated.  Default is 10.

       -watchdog_minor_maxtriggers: How many times the	minor  watchdog	 timer
       can be tripped before the test is
	      terminated.  Default is 120.

       *** Tracing, logging and statistics options:

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

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

       -stat_delimiter
	      : Set the delimiter for the statistics file

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

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

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

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

       -message_file
	      : Set the name of the message log file.

       -message_overwrite: Overwrite the message log file (default true).

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

       -shortmessage_file: Set the name of the short message log file.

       -shortmessage_overwrite:	 Overwrite the short message log file (default
	      true).

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

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

       -error_file
	      : Set the name of the error log file.

       -error_overwrite : Overwrite the error log file (default true).

       -trace_error_codes: Dumps the SIP response codes of unexpected messages
       to <scenario file
	      name>_<pid>_error_codes.log.

       -trace_calldebug : Dumps debugging information about aborted calls to
	      <scenario_name>_<pid>_calldebug.log file.

       -calldebug_file
	      : Set the name of the call debug file.

       -calldebug_overwrite: Overwrite the call debug file (default true).

       -trace_screen
	      :	    Dump     statistic	   screens	in	the	 <sce‐
	      nario_name>_<pid>_screens.log file when quitting SIPp. Useful to
	      get a final status report in background mode (-bg option).

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

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

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

       -log_file
	      : Set the name of the log actions log file.

       -log_overwrite
	      : Overwrite the log actions log file (default true).

       -ringbuffer_files:  How many error, message, shortmessage and calldebug
       files should be kept
	      after rotation?

       -ringbuffer_size : How large should error,  message,  shortmessage  and
       calldebug files be before
	      they get rotated?

       -max_log_size
	      : What is the limit for error, message, shortmessage and callde‐
	      bug file sizes.

       Signal handling:

	      SIPp can be controlled using POSIX signals. The  following  sig‐
	      nals  are	 handled:  USR1:  Similar  to pressing the 'q' 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 codes:

	      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  -2:	 Fatal
	      error binding a socket

sipp				 October 2013			       SIPP(1)
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