./sipp -sd uac >uac.xml
./sipp -sd uas >uas.xml

./sipp -sf file.xml .....

SIPp cmd lines:
./sipp -sn uas -d 0 -p 5060 -l laptop_ip -rsa ser_ip:5060 laptop_ip:5080
./sipp -r 200 -rp 100 -d 0 -sn uac -p 5080 -l laptop_ip -rsa ser_ip:5060 laptop_ip:5060
(sipp tries to send ~ 2000 cps: 200 calls each 100ms)

-sn uas default SipStone scenario
-d 0 call duration is default (0ms)
-p 5060 local port number to listen at
-l laptop_ip
-rsa ser_ip:5060 set remote sending address for sending the message
laptop_ip:5080 remote host and port number

-r 200 call rate in calls per second (200 cps)
-rp 100 rate period in ms for the call rate
-d 0 call duration is default (0ms)
-sn uac default SipStone scenario
-p 5080 local port number to transmit at
-l laptop_ip
-rsa ser_ip:5060 set remote sending address for sending the message
laptop_ip:5060 remote host and port number

Compiling:
gunzip sipp.1.1rc2.tar.gz
tar -xf sipp.1.1rc2.tar
cd sipp
make

make ossl (compile with the authentication option enabled, requires OpenSSL)

nmap -sT -O host
netstat -l

INVITE sip:service@remote_ip:remote_port SIP/2.0
Via: SIP/2.0/transport local_ip:local_port
From: sipp <sip:sipp@local_ip:local_port>;tag=call_number
To: sut <sip:service@remote_ip:remote_port>
Call-ID: call_id
Cseq: 1 INVITE
Contact: sip:sipp@local_ip:local_port
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: len

o=user1 53655765 2353687637 IN IP4 127.0.0.1
s=-
t=0 0
c=IN IP4 media_ip
m=audio media_port RTP/AVP 0
a=rtpmap:0 PCMU/8000

sipp -sf register_client.xml -inf register_client.csv -r 10 -trace_err -trace_stat -nd -fd 1 -i sender_ip sipx_ip

sip log files in sipX:
mediaserver.log
sds.log
sipauthproxy.log
sipproxy.log
sipregistrar.log
sipstatus.log

grep <call-id> /var/log/sipxpbx/sipproxy.log | syslogviewer | less

grep <call-id> /var/log/sipxpbx/sipproxy.log | syslog2siptrace > log.xml
sipviewer log.xml

USAGE:
syslogviewer < sip log file

USAGE:
syslog2siptrace < sip.log > /tmp/sip.log.xml

USAGE:
syslog2siptrace < sip.log > /tmp/sip.log.xml
sipviewer /tmp/sip.log.xml

sipp remote_host.remote_port options

-v : Display version and copyright information.

-p local_port : Set the local port number. Default is a
random free port chosen by the system.

'Via:', and 'From:' headers. Default is
primary host IP address.

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.

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.

This value can be changed 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.

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 h1. > 7 calls every 2 seconds.

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.

-sf filename : 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.

the sipp executable). If this option is omitted,
the Standard SipStone UAC scenario is loaded.
Available values in this version:

'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.

'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.

the sipp executable)

u1: UDP with one socket (default),
un: UDP with one socket per call,
t1: TCP with one socket,
tn: TCP with one socket per call,

sippcomp.so plugin. 2 additionnal
transport modes are available:

cn: un + compression.

<scenario file name>_<ppid>_messages.log

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

<scenario file name>_<ppid>_timeout.log

file. Use the '-h stat' option for a detailed
description of the statistics file content.

-trace_err : Trace all unexpected messages in
<scenario file name>_<ppid>_errors.log.

<scenario file name>_<ppid>_logs.log.

<scenario file name>_<ppid>_rtt.csv.

every freq calls in the log file defined
by -trace_rtt. Default value is 200.

Default is 'service'.

Default is 'password'

Default is 'cacert.pem'

Default is 'cakey.pem'

(in seconds). Default is 1.

(in seconds). Default is 60.

calls. Once this limit is reached, traffic
is decreased until the number of open calls
goes down. Default: (3 * call_duration (s) * rate).

processed.

is none. RTP/UDP packets received on that
port 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).

-3pcc ip:port : 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.
  Example: 3PCC-C-A scenario.
• When the first twin command is 'recvCmd' then
  this is the address of the local twin socket.
  Example: 3PCC-C-B scenario.

-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 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

for sending the messages.

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>_<ppid>_screens.log file. Especially useful
in background mode to know what the current status is.
Example: kill -SIGUSR2 732

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

export DISPLAY=:0.0