SIPp Tool:



SIPp is a  anxiety or execution test device/ movement generator for the SIP convention. It can work in both Scenarios (UAC/UAS) and makes and discharges various calls with the INVITE and BYE methods.It can likewise peruses Custom XML situation records depicting from extremely easy to complex call streams. It highlights the element showcase of measurements about running tests (call rate, round trek defer, and message insights), occasional CSV measurements dumps, TCP and UDP over various attachments or multiplexed with retransmission administration and alterably flexible call rates.


SIPp can likewise send media (RTP) movement through RTP reverberation and RTP/pcap replay. Media can be sound or feature.

SIPp can be utilized to test different genuine SIP gear like SIP intermediaries, B2BUAs, SIP media servers, SIP/x passages, SIP PBX, ... It is additionally exceptionally valuable to imitate a large number of client specialists calling your SIP framework.

Introducing SIPP on Ubuntu 12.4 linux

1. Run Terminal on Ubuntu

2. Goto root  utilizing su - and Pwd

3. wget  http://sourceforge.net/ventures/sipp/documents/sipp/3.3/sipp-3.3.tar.gz/

4. adept get or Yum  intsall sipp-XXX-tar.gz

5. compact disc Sipp-XX

6. ls -l

7. adept get introduce  openssl

8. adept get introduce  libssl-dev

9. adept get introduce  libssl1.0.0

10. adept get introduce  libssl0.9.8

11. adept get introduce  pcaputils

12. adept get introduce  libssl-dev

13. adept get introduce  libncursesw5-dev

14. adept get introduce  libncurses5-dev

15. adept get introduce  libcap-dev

16. ./sipp XXX.XX.XX.XXX -sf REGISTER_client.xml -inf Register_client.csv -m 1

In the event that you get any lapse as beneath


tring to install sipp on ubuntu I get this error
qwerty at qwerty-desktop:~/sipp.2007-02-08$ sudo make
make OSNAME=`uname|sed -e "s/CYGWIN.*/CYGWIN/"` MODELNAME=`uname -m|sed "s/Power Macintosh/ppc/"` sipp
make[1]: Entering directory `/home/qwerty/sipp.2007-02-08'
gcc   -D__LINUX -pthread -D__3PCC__       -I. -I/opt/openssl/include  -c -o scenario.o scenario.cpp
gcc: error trying to exec 'cc1plus': execvp: No such file or directory
make[1]: *** [message.o] Error 127
make[1]: Leaving directory `/home/qwerty/sipp.2007-02-08'
make: *** [all] Error 2




use these commands

sudo apt-get install -f
sudo apt-get remove libncurses5-dev libncurses5 libtinfo5 ncurses-bin
sudo apt-get update
sudo apt-get upgrade
sudo apt-get install build-essential linux-image linux-image-generic libncurses5-dev libncurses5




SIPP COMMANDS: 



sipp remote_host[:remote_port] [options] 

In place of Options use the bellow commands required

-v   : Display version and copyright information.
-aa  : Enable automatic 200 OK answer for INFO and NOTIFY
                     messages
-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.
-i     : Set the local IP address for 'Contact:','Via:', and
                     'From:' headers. Default is primary host IP address.

  -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 <scenario file
                     name>_<pid>_messages.log

  -trace_screen    : Dump statistic screens in the
                     <scenario_name>_<pid>_      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 <scenario file
                     name>_<pid>_errors.log.

  -trace_timeout   : Displays call ids for calls with timeouts in <scenario
                     file name>_<pid>_timeout.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_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.

  -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".







SIPp Scripts:


Register UAC:

Register.csv :

Register.csv file should contain the User data in below format

SEQUENTIAL
1000;10.XXX.XXX.XX;[authentication username=1000 password=123456];



Register.xml:

Register.xml file should contain  below messages

<?xml version="1.0" encoding="ISO-8859-1" ?>

<scenario name="register_client">
  <send retrans="500">
    <![CDATA[

      REGISTER sip:[remote_ip] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[field0]@[field1]>;tag=[call_number]
      To: <sip:[field0]@[field1]>
      Call-ID: [call_id]
      CSeq: 1 REGISTER
      Contact: sip:[field0]@[local_ip]:[local_port]
      Max-Forwards: 70
      Expires: 1800
      User-Agent: SIPp/Linux
      Content-Length: 0

    ]]>
  </send>

  <recv response="401" auth="true">
  </recv>

  <send retrans="500">
    <![CDATA[

      REGISTER sip:[remote_ip] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[field0]@[field1]>;tag=[call_number]
      To: <sip:[field0]@[field1]>
      Call-ID: [call_id]
      CSeq: 2 REGISTER
      Contact: sip:[field0]@[local_ip]:[local_port]
      [authentication username=21613 password=123456]
      Max-Forwards: 70
      Expires: 1800
      User-Agent: SIPp/Linux
      Content-Length: 0

    ]]>
  </send>

  <recv response="200" rtd="true">
  </recv>

</scenario>


INVITE UAC :

Invite.CSV:

SEQUENTIAL
1000;10.XXX.XXX.XXX;[authentication username=1000 password=123456];1001;
  
INVITE.XML 

Invite.xml should contain below message format  


<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<!-- You will need to compile SIPp with OpenSSL support 'make ossl' to use this call scenario -->

<!-- Execute this script with SIPp using the following command assuming your UAS is 10.0.0.10 -->
<!-- Replace 10.0.0.10 with your SIP proxy’s address. The command will generate 10 calls (-r) per 10000 -->
<!-- milliseconds (-rp), max 100 concurrent calls (-l) and make a max of 100000 calls (-m) -->

<!-- ./sipp 10.0.0.10 -sf invite-auth-sdp-nomedia.xml -inf user-accounts.csv -m 100000 -l 100 -r 10 -rp 10000 -->

<scenario name="UAC INvite    ">
  <send retrans="500" start_txn="invite">
    <![CDATA[

INVITE sip:[field3]@[field1]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[field3]@[field1]:[remote_port]>
Call-ID: [call_id]
CSeq: [cseq] INVITE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
User-Agent: SIPp Tester UAC
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Content-Type: application/sdp
Content-Length: [len]

v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000


]]>
  </send>

  <recv response="100" optional="true" response_txn="invite">
  </recv>

  <recv response="180" optional="true" response_txn="invite">
  </recv>


  <recv response="200" rtd="true" rrs="true" response_txn="invite">
    
  </recv>

   <send ack_txn="invite">
    <![CDATA[

ACK sip:[field3]@[field1]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
[last_To]
[routes]
Call-ID: [call_id]
CSeq: [cseq] ACK
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
User-Agent: SIPp Tester UAC
Content-Length: 0

]]>
  </send>
 <pause milliseconds="10000"/>
 <send retrans="500">
    <![CDATA[
BYE sip:[field3]@[field1]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
[last_To]
Call-ID: [call_id]
CSeq: [cseq] BYE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
User-Agent: SIPp Tester UAC
Content-Length: 0
]]>
  </send>

  <recv response="200" crlf="true">
  </recv>

  <!-- definition of the response time repartition table (unit is ms) -->
  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>

  <!-- definition of the call length repartition table (unit is ms) -->
  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>


</scenario>