VoIP: Perintah SIPp

From OnnoWiki
Revision as of 10:07, 14 February 2010 by Onnowpurbo (talk | contribs) (New page: ==Usage:== sipp remote_host[:remote_port] [options] ==Available options:== -v : Display version and copyright information. -aa : Enable automatic 20...)
(diff) ← Older revision | Latest revision (diff) | Newer revision → (diff)
Jump to navigation Jump to search

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.
  -auth_uri        : Force the value of the URI for authentication.
                     By default, the URI is composed of
                     remote_ip:remote_port.
  -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.
  -calldebug_file  : Set the name of the call debug file.
  -calldebug_overwrite: Overwrite the call debug file (default true).
  -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
                     
  -error_file      : Set the name of the error log file.
  -error_overwrite : Overwrite the error log file (default true).
  -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).
  -log_file        : Set the name of the log actions log file.
  -log_overwrite   : Overwrite the log actions log file (default true).
  -lost            : Set the number of packets to lose by default (scenario
                     specifications override this value).
  -rtcheck         : Select the retransmisison detection method: full
                     (default) or loose.
  -m               : Stop the test and exit when 'calls' calls are processed
  -mi              : Set the local media IP address (default: local primary
                     host 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).
  -message_file    : Set the name of the message log file.
  -message_overwrite: Overwrite the message log file (default true).
  -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.
  -plugin          : Load a plugin.
  -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.
  -sleep           : How long to sleep for at startup. Default unit is
                     seconds.
  -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.
  -shortmessage_file: Set the name of the short message log file.
  -shortmessage_overwrite: Overwrite the short message log file (default true).
  -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).
  -timeout_error   : SIPp fails if the global timeout is reached is set
                     (-timeout option required).
  -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>_cenaris.log file when quitting
                     SIPp. Useful to get a final status report in background
                     mode (-bg option).
  -trace_err       : Trace all unexpected messages in <scenario file
                     name>_<pid>_errors.log.
  -trace_calldebug : Dumps debugging information about aborted calls to
                     <scenario_name>_<pid>_calldebug.log file.
  -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.
  -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.
  -ap              : Set the password for authentication challenges. Default
                     is 'password
  -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_crl         : Set the name for Certificate Revocation List file. If not
                     specified, X509 CRL is not activated.
  -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".
  -set             : variable value
                     Set the global variable parameter named "variable" to
                     "value".
  -dynamicStart    : variable value
                     Set the start offset of dynamic_id varaiable
  -dynamicMax      : variable value
                     Set the maximum of dynamic_id variable     
  -dynamicStep     : variable value
                     Set the increment of dynamic_id variable

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


Pranala Menarik