Provided by: sip-tester_3.2-1build1_amd64 
NAME
sipp - Session Initiation Protol (SIP) performance testing tool
DESCRIPTION
Usage:
sipp remote_host[:remote_port] [options]
Available options:
-v : Display version and copyright information.
-aa : Enable automatic 200 OK answer for INFO, UPDATE and NOTIFY messages.
-base_cseq
: Start value of [cseq] for each call.
-bg : Launch SIPp in background mode.
-bind_local
: Bind socket to local IP address, i.e. the local IP address is used as the source
IP address. If SIPp runs in server mode it will only listen on the local IP
address instead of all IP addresses.
-buff_size
: Set the send and receive buffer size.
-cid_str
: Call ID string (default %u-%p@%s). %u=call_number, %s=ip_address,
%p=process_number, %%=% (in any order).
-ci : Set the local control IP address
-cp : Set the local control port number. Default is 8888.
-d : Controls the length of calls. More precisely, this controls the duration of
'pause' instructions in the scenario, if they do not have a 'milliseconds' section.
Default value is 0 and default unit is milliseconds.
-deadcall_wait
: How long the Call-ID and final status of calls should be kept to improve message
and error logs (default unit is ms).
-default_behaviors: Set the default behaviors that SIPp will use.
Possbile values are: - all Use all default behaviors - none Use no default
behaviors - bye Send byes for aborted calls - abortunexp Abort calls on
unexpected messages - pingreply Reply to ping requests If a behavior is
prefaced with a -, then it is turned off. Example: all,-bye
-f : Set the statistics report frequency on screen. Default is 1 and default unit is
seconds.
-fd : Set the statistics dump log report frequency. Default is 60 and default unit is
seconds.
-i : Set the local IP address for 'Contact:','Via:', and 'From:' headers. Default is
primary host IP address.
-inf : Inject values from an external CSV file during calls into the scenarios. First
line of this file say whether the data is to be read in sequence (SEQUENTIAL),
random (RANDOM), or user (USER) order. Each line corresponds to one call and has
one or more ';' delimited data fields. Those fields can be referred as [field0],
[field1], ... in the xml scenario file. Several CSV files can be used
simultaneously (syntax: -inf f1.csv -inf f2.csv ...)
-infindex
: file field Create an index of file using field. For example -inf users.csv
-infindex users.csv 0 creates an index on the first key.
-ip_field
: Set which field from the injection file contains the IP address from which the
client will send its messages. If this option is omitted and the '-t ui' option is
present, then field 0 is assumed. Use this option together with '-t ui'
-l : Set the maximum number of simultaneous calls. Once this limit is reached, traffic
is decreased until the number of open calls goes down. Default:
(3 * call_duration (s) * rate).
-lost : Set the number of packets to lose by default (scenario specifications override
this value).
-m : Stop the test and exit when 'calls' calls are processed
-mi : Set the local media IP address
-master
: 3pcc extended mode: indicates the master number
-max_recv_loops
: Set the maximum number of messages received read per cycle. Increase this value
for high traffic level. The default value is 1000.
-max_sched_loops : Set the maximum number of calsl run per event loop.
Increase this value for high traffic level. The default value is 1000.
-max_reconnect
: Set the the maximum number of reconnection.
-max_retrans
: Maximum number of UDP retransmissions before call ends on timeout. Default is 5
for INVITE transactions and 7 for others.
-max_invite_retrans: Maximum number of UDP retransmissions for invite
transactions before call ends on timeout.
-max_non_invite_retrans: Maximum number of UDP retransmissions for non-invite
transactions before call ends on timeout.
-max_log_size
: What is the limit for error and message log file sizes.
-max_socket
: Set the max number of sockets to open simultaneously. This option is significant
if you use one socket per call. Once this limit is reached, traffic is distributed
over the sockets already opened. Default value is 50000
-mb : Set the RTP echo buffer size (default: 2048).
-mp : Set the local RTP echo port number. Default is 6000.
-nd : No Default. Disable all default behavior of SIPp which are the following: - On
UDP retransmission timeout, abort the call by
sending a BYE or a CANCEL
- On receive timeout with no ontimeout attribute, abort
the call by sending a BYE or a CANCEL
- On unexpected BYE send a 200 OK and close the call - On unexpected CANCEL send a
200 OK and close the call - On unexpected PING send a 200 OK and continue the call
- On any other unexpected message, abort the call by
sending a BYE or a CANCEL
-nr : Disable retransmission in UDP mode.
-nostdin
: Disable stdin.
-p : Set the local port number. Default is a random free port chosen by the system.
-pause_msg_ign
: Ignore the messages received during a pause defined in the scenario
-periodic_rtd
: Reset response time partition counters each logging interval.
-r : Set the call rate (in calls per seconds). This value can bechanged during test
by pressing '+','_','*' or '/'. Default is 10. pressing '+' key to increase call
rate by 1 * rate_scale, pressing '-' key to decrease call rate by 1 * rate_scale,
pressing '*' key to increase call rate by 10 * rate_scale, pressing '/' key to
decrease call rate by 10 * rate_scale. If the -rp option is used, the call rate is
calculated with the period in ms given by the user.
-rp : Specify the rate period for the call rate. Default is 1 second and default unit
is milliseconds. This allows you to have n calls every m milliseconds (by using -r
n -rp m). Example: -r 7 -rp 2000 ==> 7 calls every 2 seconds.
-r 10 -rp 5s => 10 calls every 5 seconds.
-rate_scale
: Control the units for the '+', '-', '*', and '/' keys.
-rate_increase
: Specify the rate increase every -fd units (default is seconds). This allows you
to increase the load for each independent logging period. Example: -rate_increase
10 -fd 10s
==> increase calls by 10 every 10 seconds.
-rate_max
: If -rate_increase is set, then quit after the rate reaches this value. Example:
-rate_increase 10 -rate_max 100
==> increase calls by 10 until 100 cps is hit.
-no_rate_quit
: If -rate_increase is set, do not quit after the rate reaches -rate_max.
-recv_timeout
: Global receive timeout. Default unit is milliseconds. If the expected message is
not received, the call times out and is aborted.
-send_timeout
: Global send timeout. Default unit is milliseconds. If a message is not sent (due
to congestion), the call times out and is aborted.
-reconnect_close : Should calls be closed on reconnect?
-reconnect_sleep : How long (in milliseconds) to sleep between the close and
reconnect?
-ringbuffer_files: How many error/message files should be kept after
rotation?
-ringbuffer_size : How large should error/message files be before they get
rotated?
-rsa : Set the remote sending address to host:port for sending the messages.
-rtp_echo
: Enable RTP echo. RTP/UDP packets received on port defined by -mp are echoed to
their sender. RTP/UDP packets coming on this port + 2 are also echoed to their
sender (used for sound and video echo).
-rtt_freq
: freq is mandatory. Dump response times every freq calls in the log file defined
by -trace_rtt. Default value is 200.
-s : Set the username part of the resquest URI. Default is 'service'.
-sd : Dumps a default scenario (embeded in the sipp executable)
-sf : Loads an alternate xml scenario file. To learn more about XML scenario syntax,
use the -sd option to dump embedded scenarios. They contain all the necessary help.
-oocsf : Load out-of-call scenario.
-oocsn : Load out-of-call scenario.
-skip_rlimit
: Do not perform rlimit tuning of file descriptor limits. Default: false.
-slave : 3pcc extended mode: indicates the slave number
-slave_cfg
: 3pcc extended mode: indicates the file where the master and slave addresses are
stored
-sn : Use a default scenario (embedded in the sipp executable). If this option is
omitted, the Standard SipStone UAC scenario is loaded. Available values in this
version:
- 'uac'
: Standard SipStone UAC (default).
- 'uas'
: Simple UAS responder.
- 'regexp'
: Standard SipStone UAC - with regexp and
variables.
- 'branchc'
: Branching and conditional branching in
scenarios - client.
- 'branchs'
: Branching and conditional branching in
scenarios - server.
Default 3pcc scenarios (see -3pcc option):
- '3pcc-C-A' : Controller A side (must be started after
all other 3pcc scenarios)
- '3pcc-C-B' : Controller B side. - '3pcc-A' : A side. - '3pcc-B' : B side.
-stat_delimiter
: Set the delimiter for the statistics file
-stf : Set the file name to use to dump statistics
-t : Set the transport mode: - u1: UDP with one socket (default), - un: UDP with one
socket per call, - ui: UDP with one socket per IP address The IP
addresses must be defined in the injection file.
- t1: TCP with one socket, - tn: TCP with one socket per call, - l1: TLS with one
socket, - ln: TLS with one socket per call, - c1: u1 + compression (only if
compression plugin
loaded),
- cn: un + compression (only if compression plugin
loaded).
This plugin is not provided with sipp.
-timeout
: Global timeout. Default unit is seconds. If this option is set, SIPp quits after
nb units (-timeout 20s quits after 20 seconds).
-timer_resol
: Set the timer resolution. Default unit is milliseconds. This option has an
impact on timers precision.Small values allow more precise scheduling but impacts
CPU usage.If the compression is on, the value is set to 50ms. The default value is
10ms.
-sendbuffer_warn : Produce warnings instead of errors on SendBuffer
failures.
-trace_msg
: Displays sent and received SIP messages in <scenario file
name>_<pid>_messages.log
-trace_shortmsg
: Displays sent and received SIP messages as CSV in <scenario file
name>_<pid>_shortmessages.log
-trace_screen
: Dump statistic screens in the <scenario_name>_<pid>_0ms.
-trace_err
: Trace all unexpected messages in <scenario file name>_<pid>_errors.log.
-trace_stat
: Dumps all statistics in <scenario_name>_<pid>.csv file. Use the '-h stat' option
for a detailed description of the statistics file content.
-trace_counts
: Dumps individual message counts in a CSV file.
-trace_rtt
: Allow tracing of all response times in <scenario file name>_<pid>_rtt.csv.
-trace_logs
: Allow tracing of <log> actions in <scenario file name>_<pid>_logs.log.
-users : Instead of starting calls at a fixed rate, begin 'users' calls at startup, and
keep the number of calls constant.
-3pcc : Launch the tool in 3pcc mode ("Third Party call control"). The passed ip address
is depending on the 3PCC role. - When the first twin command is 'sendCmd' then
this is
the address of the remote twin socket.
SIPp will try to
connect to this address:port to send the twin command (This instance must be
started after all other 3PCC scenarii).
Example: 3PCC-C-A scenario.
- When the first twin command is 'recvCmd' then this is
the address of the local twin socket. SIPp will open this address:port to listen
for twin command.
Example: 3PCC-C-B scenario.
-tdmmap
: Generate and handle a table of TDM circuits. A circuit must be available for the
call to be placed. Format: -tdmmap {0-3}{99}{5-8}{1-31}
-key : keyword value Set the generic parameter named "keyword" to "value".
Signal handling:
SIPp can be controlled using posix signals. The following signals are handled:
USR1: Similar to press 'q' keyboard key. It triggers a soft exit
of SIPp. No more new calls are placed and all ongoing calls are finished before
SIPp exits. Example: kill -SIGUSR1 732
USR2: Triggers a dump of all statistics screens in
<scenario_name>_<pid>_screens.log file. Especially useful in background mode to
know what the current status is. Example: kill -SIGUSR2 732
Exit code:
Upon exit (on fatal error or when the number of asked calls (-m option) is reached,
sipp exits with one of the following exit code:
0: All calls were successful 1: At least one call failed
97: exit on internal command. Calls may have been processed 99: Normal exit without
calls processed -1: Fatal error
Example:
Run sipp with embedded server (uas) scenario:
./sipp -sn uas
On the same host, run sipp with embedded client (uac) scenario
./sipp -sn uac 127.0.0.1
SIPp v3.1, version unknown, built Jun 13 2010, 15:34:03.
This program is free software; you can redistribute it and/or modify it under the
terms of the GNU General Public License as published by the Free Software
Foundation; either version 2 of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A
PARTICULAR PURPOSE. See the GNU General Public License for more details.
You should have received a copy of the GNU General Public License along with this
program; if not, write to the Free Software Foundation, Inc., 59 Temple Place,
Suite 330, Boston, MA 02111-1307 USA
Author: see source files.