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