How to install and stress test SIP server with SIPp in Centos/Alma

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Install dependencies.

cd /opt
yum install make gcc gcc-c++ ncurses ncurses.x86_64 ncurses-devel ncurses-devel.x86_64 openssl libnet libpcap libpcap-devel libpcap.x86_64 libpcap-devel.x86_64 gsl gsl-devel

Download SIPp
you cand download SIPp 3.3 through the below link. it's the latest version.
https://sourceforge.net/projects/sipp/files/sipp/3.4/sipp-3.3.990.tar.gz/download

*if it was not working, refer to http://sipp.sourceforge.net/

# Download from either of the site
wget https://sourceforge.net/projects/sipp/files/sipp/3.4/sipp-3.3.990.tar.gz
# or
wget https://excellmedia.dl.sourceforge.net/project/sipp/sipp/3.4/sipp-3.3.990.tar.gz

Unpacking tar.gz

tar -zxvf sipp-3.3.990.tar.gz

Configuring SIPp

cd sipp-3.3.990
./configure --with-pcap

Installing SIPp

make all

Execute SIPp

./sipp

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
./sipp -sn uac -d 60000 -s 1009202000 sip_server_ip:port -l 500 -r 10 -trace_err -error_file sipperror
./sipp -sn uac -rtp_echo -d 60000 -s 1009202000 sip_server_ip:port -l 500 -r 10 -trace_err -error_file sipperror

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#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.

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*** 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,
                      - 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.

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*** 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

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*** Call behavior options:
 
   -aa              : Enable automatic 200 OK answer for INFO, UPDATE and NOTIFY messages.
   -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.  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
                     
   -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

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*** 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 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'

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*** 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.

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*** 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 -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.
   -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.

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*** 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

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*** 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

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*** 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.

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*** 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).
   -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.

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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

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./sipp -sn uac -rtp_echo -d 60000 -s 1009202000 [[sip_server_ip:port]] -l 500 -r 10 -trace_err -error_file sipperror
load average: 25.34, 17.80, 10.88
load average: 44.44, 28.14, 15.33
load average: 56.15, 27.42, 14.49


------------------------------ Scenario Screen -------- [1-9]: Change Screen --
  Call-rate(length)   Port   Total-time  Total-calls  Remote-host
10.0(60000 ms)/1.000s   5060     192.40 s         1378  [[sip_server_ip:port]](UDP)


  4 new calls during 1.001 s period      1 ms scheduler resolution
  500 calls (limit 500)                  Peak was 500 calls, after 50 s
  0 Running, 762 Paused, 73 Woken up
  2066 dead call msg (discarded)         113 out-of-call msg (discarded)
  3 open sockets
  3250544 Total echo RTP pckts 1st stream 2776.005 last period RTP rate (kB/s)
  0 Total echo RTP pckts 2nd stream      0.000 last period RTP rate (kB/s)


                                 Messages  Retrans   Timeout   Unexpected-Msg
      INVITE ---------->         1378      3314      41
         100 <----------         1235      2660      0         0
         180 <----------         0         0         0         0
         183 <----------         0         0         0         0
         200 <----------  E-RTD1 1173      2329      0         0
         ACK ---------->         1173      2329
       Pause [     1:00]         1173                          18
         BYE ---------->         879       2423      35
         200 <----------         784       0         0         0


------ [+|-|*|/]: Adjust rate ---- [q]: Soft exit ---- [p]: Pause traffic -----


Last Error: Discarding message which can't be mapped to a known SIPp cal...
------------------------------ Scenario Screen -------- [1-9]: Change Screen --
  Call-rate(length)   Port   Total-time  Total-calls  Remote-host
10.0(60000 ms)/1.000s   5060     306.17 s         1381  [[sip_server_ip:port]](UDP)


  0 new calls during 0.000 s period      0 ms scheduler resolution
  0 calls (limit 500)                    Peak was 500 calls, after 50 s
  0 Running, 115 Paused, 0 Woken up
  5201 dead call msg (discarded)         373 out-of-call msg (discarded)
  1 open sockets


                                 Messages  Retrans   Timeout   Unexpected-Msg
      INVITE ---------->         1381      3433      83
         100 <----------         1298      3018      0         0
         180 <----------         0         0         0         0
         183 <----------         0         0         0         0
         200 <----------  E-RTD1 1298      2833      0         0
         ACK ---------->         1298      2833
       Pause [     1:00]         1298                          43
         BYE ---------->         1255      2843      59
         200 <----------         1196      0         0         0


------------------------------ Test Terminated --------------------------------




----------------------------- Statistics Screen ------- [1-9]: Change Screen --
  Start Time             | 2023-02-14   07:39:44.663589 1676360384.663589
  Last Reset Time        | 2023-02-14   07:44:50.842439 1676360690.842439
  Current Time           | 2023-02-14   07:44:50.843429 1676360690.843429
-------------------------+---------------------------+--------------------------
  Counter Name           | Periodic value            | Cumulative value
-------------------------+---------------------------+--------------------------
  Elapsed Time           | 00:00:00:000000           | 00:05:06:179000
  Call Rate              |    0.000 cps              |    4.510 cps
-------------------------+---------------------------+--------------------------
  Incoming call created  |        0                  |        0
  OutGoing call created  |        0                  |     1381
  Total Call created     |                           |     1381
  Current Call           |        0                  |
-------------------------+---------------------------+--------------------------
  Successful call        |        0                  |     1196
  Failed call            |        0                  |      185
-------------------------+---------------------------+--------------------------
  Response Time 1        | 00:00:00:000000           | 00:00:12:324000
  Call Length            | 00:00:00:000000           | 00:01:14:375000
------------------------------ Test Terminated --------------------------------




2023-02-14      07:44:50.653313 1676360690.653313: Dead call 1100-9409@[[sipp_ip]] (aborted at index 1), received 'BYE sip:sipp@[[sipp_ip]]:5060 SIP/2.0
Via: SIP/2.0/UDP [[sip_server_ip:port]];rport;branch=z9hG4bKPj31230a90-40f0-4133-a112-49db091fa35b
From: "1009202000" <sip:1009202000@[[sip_server_ip]]>;tag=68540b17-e8a0-4c78-b84f-81c1c650bfe0
To: "sipp" <sip:sipp@[[sipp_ip]]>;tag=9409SIPpTag001100
Call-ID: 1100-9409@[[sipp_ip]]
CSeq: 15763 BYE
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.0
Content-Length:  0