Usage
sipp remote_host[:remote_port] [options]
下列是命令选项，其中比较常用的如下:
-v 显示版本号
-sn 使用系统内置脚本场景
-sf 使用自己制定脚本场景
-i 设置sip报文'Contact:','Via:'和'From:'的ip地址
等等
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Available options
-v : Display version and copyright information.
-aa : Enable automatic 200 OK answer for INFO 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.
-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 (in milliseconds) 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.
-f : Set the statistics report frequency on screen (in
seconds). Default is 1.
-fd : Set the statistics dump log report frequency (in
seconds). Default is 60.
-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) or random (RANDOM) 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.
-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).
-m : Stop the test and exit when 'calls' calls are processed
-mi : Set the local media IP address
-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_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_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.
-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
-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,
pressing '-' key to decrease call rate by 1,
pressing '*' key to increase call rate by 10,
pressing '/' key to decrease call rate by 10.
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 in milliseconds for the call
rate. Default is 1 second. 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.
-rate_increase : Specify the rate increase every -fd seconds. This allows
you to increase the load for each independent logging
period.
Example: -rate_increase 10 -fd 10
==> 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 -max_rate 100
==> increase calls by 10 until 100 cps is hit.
-recv_timeout : Global receive timeout in milliseconds. If the expected
message is not received, the call times out and is
aborted.
-reconnect_close : Should calls be closed on reconnect?
-reconnect_sleep : How long to sleep between the close and reconnect?
-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.
-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).
-timeout : Global timeout in seconds. If this option is set, SIPp
quits after nb seconds.
-timer_resol : Set the timer resolution in 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.
-trace_msg : Displays sent and received SIP messages in
name>__messages.log
-trace_screen : Dump statistic screens in the
__ s.log file when quitting
SIPp. Useful to get a final status report in background
mode (-bg option).
-trace_err : Trace all unexpected messages in name>__errors.log.
-trace_timeout : Displays call ids for calls with timeouts in file name>__timeout.log
-trace_stat : Dumps all statistics in _.csv file.
Use the '-h stat' option for a detailed description of
the statistics file content.
-trace_rtt : Allow tracing of all response times in name>__rtt.csv.
-trace_logs : Allow tracing of actions in name>__logs.log.
-up_nb : Set the number of updates of the internal clock during
the reading of received messages. Default value is 1.
-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 : key value
Set the generic parameter named "key" to "value".
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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
__screens.log file. Especially useful
in background mode to know what the current status is.
Example: kill -SIGUSR2 732
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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
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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
