Provided by: sip-tester_3.6.1-1build1_amd64 bug

NAME

       sipp - SIP testing tool and traffic generator

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, - l1: TLS  with  one
              socket,  - ln: TLS with one socket per call, - s1: SCTP with one socket, - sn: SCTP
              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 chosen 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.

       -tls_cert
              : Set the name for TLS Certificate file. Default is 'cacert.pem

       -tls_key
              : Set the name for TLS Private Key file. Default is 'cakey.pem'

       -tls_ca
              :  Set  the  name  for  TLS  CA  file.  If  not specified, X509 verification is not
              activated.

       -tls_crl
              : Set the name for Certificate Revocation List file. If not specified, X509 CRL  is
              not activated.

       -tls_version
              : Set the TLS protocol version to use (1.0, 1.1, 1.2) -- default is autonegotiate

       -multihome
              : Set multihome address for SCTP

       -heartbeat
              : Set heartbeat interval in ms for SCTP

       -assocmaxret
              : Set association max retransmit counter for SCTP

       -pathmaxret
              : Set path max retransmit counter for SCTP

       -pmtu  : Set path MTU for SCTP

       -gracefulclose
              : If true, SCTP association will be closed with SHUTDOWN (default).  If false, SCTP
              association will be closed by ABORT.

       *** 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  specifications  override
              this value).

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

       -set   : variable value Set the global variable parameter named "variable" 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, NOTIFY, OPTIONS and UPDATE.

       -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 'service'.

       -default_behaviors: Set the default behaviors that SIPp will use.
              Possible 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 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

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

       -callid_slash_ign:  Don't  treat  a  triple-slash  in Call-IDs as indicating an extra SIPp
              prefix.

       *** 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
              ../path/to/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.

       -rtp_payload
              : RTP default payload type.

       -rtp_threadtasks : RTP number of playback tasks per thread.

       -rtp_buffsize
              : Set the rtp socket send/receive buffer size.

       *** 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,
              pressing  '*'  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 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 -rate_interval units (default is seconds).   This
              allows  you  to  increase  the  load for each independent logging period.  Example:
              -rate_increase 10 -rate_interval 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.

       -rate_interval
              :  Set  the  interval by which the call rate is increased. Defaults to the value of
              -fd.

       -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 message 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 seconds).

       -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 statistics 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  <scenario_name>_<pid>_screens.log  file  when
              quitting SIPp. Useful to get a final status report in background mode (-bg option).

       -screen_file
              : Set the name of the screen file.

       -screen_overwrite: Overwrite the screen file (default true).

       -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 calldebug file sizes.

       Signal handling:

              SIPp can be controlled using POSIX signals.  The  following  signals  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 v3.6.1-TLS-SCTP-PCAP-RTPSTREAM.

              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.