SIPp with SIP-I (ISUP) support

SIPp with SIP-I support works on Linux.

Download sipp source code (tar.gz) from https://github.com/SIPp/sipp/releases
You can use wget to download
# wget https://github.com/SIPp/sipp/archive/v3.6.0.tar.gz

Unpack the files
# tar -zxvf v3.6.0.tar.gz

Change directory to sipp folder
#  cd sipp-3.6.0

You have four options to compile SIPp:

a) Without TLS (Transport Layer Security), SCTP or PCAP support
./configure

b) With TLS support
./configure --with-openssl

c) With PCAP play support:
./configure --with-pcap

d) With SCTP support:
./configure --with-sctp

e) You can also combine these various options, e.g.:
./configure --with-sctp --with-pcap --with-openssl

After that give make command
# make

See more details on https://sipp.readthedocs.io/en/v3.6.0/installation.html#installing-sipp

Errors and Solutions normally encountered

Error:  /bin/sh: gcc: command not found
Solution: # sudo yum install gcc

 Error: /bin/sh: g++: command not found
Solution: # sudo yum groupinstall 'Development Tools'

 Error: sipp.hpp:52:20: fatal error: curses.h: No such file or directory
Solution: # sudo yum install ncurses-devel ncurses

 Error: auth.c:32:10: fatal error: openssl/md5.h: No such file or directory
Solution: # sudo yum install openssl
                # sudo yum install openssl-devel

 Error: send_packets.c:44:10: fatal error: pcap.h: No such file or directory
Solution: # sudo yum install libpcap
                # sudo yum install libpcap-devel

configure: error: SCTP library missing
Solution: # sudo yum install lksctp*

finaly you will witness make completion.
make[1]: Leaving directory `/root/sipp-3.6.0'

SIPp is ready. Lets Test it.

 # ./sipp -v
SIPp v3.6.0-TLS-SCTP-PCAP-RTPSTREAM.

Sample UAS Script

<?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <!-- 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 --> <!-- --> <!-- Sipp default 'uas' scenario. --> <!-- --> <scenario name="Basic UAS responder"> <!-- By adding rrs="true" (Record Route Sets), the route sets --> <!-- are saved and used for following messages sent. Useful to test --> <!-- against stateful SIP proxies/B2BUAs. --> <recv request="INVITE" crlf="true"> </recv> <!-- The '[last_*]' keyword is replaced automatically by the --> <!-- specified header if it was present in the last message received --> <!-- (except if it was a retransmission). If the header was not --> <!-- present or if no message has been received, the '[last_*]' --> <!-- keyword is discarded, and all bytes until the end of the line --> <!-- are also discarded. --> <!-- --> <!-- If the specified header was present several times in the --> <!-- message, all occurences are concatenated (CRLF seperated) --> <!-- to be used in place of the '[last_*]' keyword. --> <send> <![CDATA[ SIP/2.0 180 Ringing [last_Via:] [last_From:] [last_To:];tag=[pid]SIPpTag01[call_number] [last_Call-ID:] [last_CSeq:] Contact: <sip:[local_ip]:[local_port];transport=[transport]> Content-Type: multipart/mixed;boundary=testing Content-Length: [len] MIME-Version: 1.0 --testing Content-Type: application/sdp 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 --testing Content-Type: application/isup;base=itu-t92+;version=itu Content-Disposition: session;handling=optional \x06\x14\x01\x00 --testing-- ]]> </send> <send retrans="500"> <![CDATA[ SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:];tag=[pid]SIPpTag01[call_number] [last_Call-ID:] [last_CSeq:] Contact: <sip:[local_ip]:[local_port];transport=[transport]> Content-Type: multipart/mixed;boundary=testing Content-Length: [len] MIME-Version: 1.0 --testing Content-Type: application/sdp 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 --testing Content-Type: application/isup;base=itu-t92+;version=itu Content-Disposition: session;handling=optional \x09\x00 --testing-- ]]> </send> <recv request="ACK" optional="true" rtd="true" crlf="true"> </recv> <recv request="BYE"> </recv> <send> <![CDATA[ SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:] [last_Call-ID:] [last_CSeq:] Contact: <sip:[local_ip]:[local_port];transport=[transport]> Content-Type: application/isup;base=itu-t92+;version=itu Content-Disposition: session;handling=optional Content-Length: [len] \x10\x00 ]]> </send> <!-- Keep the call open for a while in case the 200 is lost to be --> <!-- able to retransmit it if we receive the BYE again. --> <timewait milliseconds="4000"/> <!-- 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>

Sample UAC Script

<?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <!-- 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 --> <!-- --> <!-- Sipp default 'uac' scenario. --> <!-- --> <scenario name="Basic Sipstone UAC"> <!-- In client mode (sipp placing calls), the Call-ID MUST be --> <!-- generated by sipp. To do so, use [call_id] keyword. --> <send retrans="500"> <![CDATA[ INVITE sip:[service]@[remote_ip]:[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:[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: multipart/mixed;boundary=level3-boundary Content-Length: [len] MIME-Version: 1.0 --level3-boundary Content-Type: application/sdp 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 --level3-boundary Content-Type: application/isup;base=itu-t92+;version=itu Content-Disposition: session;handling=optional \x01\x00\x20\x00\x00\x03\x02\x06\x04\x01\x10\x21\x43\x0a\x08\x01\x15\x44\x21\x43\x65\x87\x09\x00 --level3-boundary-- ]]> </send> <recv response="100" optional="true"> </recv> <recv response="180" optional="true"> </recv> <recv response="183" optional="true"> </recv> <!-- By adding rrs="true" (Record Route Sets), the route sets --> <!-- are saved and used for following messages sent. Useful to test --> <!-- against stateful SIP proxies/B2BUAs. --> <recv response="200" rtd="true"> </recv> <!-- Packet lost can be simulated in any send/recv message by --> <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. --> <send> <![CDATA[ ACK sip:[service]@[remote_ip]:[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:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 1 ACK Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> <!-- This delay can be customized by the -d command-line option --> <!-- or by adding a 'milliseconds = "value"' option here. --> <pause/> <!-- The 'crlf' option inserts a blank line in the statistics report. --> <send retrans="500"> <![CDATA[ BYE sip:[service]@[remote_ip]:[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:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 2 BYE Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Type: application/isup;base=itu-t92+;version=itu Content-Disposition: session;handling=optional Content-Length: [len] \x0c\x02\x00\x02\x87\x90 ]]> </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>

How to extract ISUP data from wireshark pcap file and use in a scenario script?

Open SIP-I pcap file in wireshark.

Select sip message in wireshark packet list pane.
Expand message in packet detail pane and look for "ISDN User Part"
Right click on "ISDN User Part", click on Copy --> Bytes --> Hex Stream

You will get something like this
010020000003020604011021430a08011544214365870900

Insert \x at start and after every two digits. Finaly it should look like below:

\x01\x00\x20\x00\x00\x03\x02\x06\x04\x01\x10\x21\x43\x0a\x08\x01\x15\x44\x21\x43\x65\x87\x09\x00

 

Use this string in --level3-boundary-- of sip message in scenario

INVITE sip:[service]@[remote_ip]:[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:[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: multipart/mixed;boundary=level3-boundary Content-Length: [len] MIME-Version: 1.0 --level3-boundary Content-Type: application/sdp 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 --level3-boundary Content-Type: application/isup;base=itu-t92+;version=itu Content-Disposition: session;handling=optional \x01\x00\x20\x00\x00\x03\x02\x06\x04\x01\x10\x21\x43\x0a\x08\x01\x15\x44\x21\x43\x65\x87\x09\x00 --level3-boundary--

Links:

https://raw.githubusercontent.com/shariramani/sipp-scenarios/master/sh_sipi_uac.xml
https://raw.githubusercontent.com/shariramani/sipp-scenarios/master/sh_sipi_uas.xml