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https://wiki.kolmisoft.com/index.php?title=Performance_Testing_with_Sipp&diff=1505
Performance Testing with Sipp
2008-06-16T08:43:43Z
<p>Digilab: </p>
<hr />
<div>With this test enviroment we can simulate a single or multple users calling trought mor/asterisk.<br />
<br />
In the scenario uac_auth.xml, sipp start sending a SIP INVITE to MOR and if AUTH is requested, sipp resend SIP INVITE with PROXY AUTHENTICATION,<br />
Playing with sipp parameters you can increase call data rate and other variable like call duration, etc.<br />
<br />
<br />
Download and compile the last sipp:<br />
<br />
wget http://surfnet.dl.sourceforge.net/sourceforge/sipp/sipp.3.1.src.tar.gz<br />
<br />
tar xvfz sipp.3.1.src.tar.gz<br />
<br />
cd sipp.svn<br />
<br />
apt-get install libssl-dev<br />
<br />
make ossl<br />
<br />
edit a new file "users.csv":<br />
<br />
SEQUENTIAL<br />
<br />
user001;[authentication username=user001 password=password001]<br />
<br />
user002;[authentication username=user001 password=password001]<br />
<br />
user003;[authentication username=user001 password=password001]<br />
<br />
edit a new file "uac_auth.xml":<br />
<br />
<?xml version="1.0" encoding="ISO-8859-1" ?><br />
<!DOCTYPE scenario SYSTEM "sipp.dtd"><br />
<!-- This program is free software; you can redistribute it and/or --><br />
<!-- modify it under the terms of the GNU General Public License as --><br />
<!-- published by the Free Software Foundation; either version 2 of the --><br />
<!-- License, or (at your option) any later version. --><br />
<!-- --><br />
<!-- This program is distributed in the hope that it will be useful, --><br />
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of --><br />
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the --><br />
<!-- GNU General Public License for more details. --><br />
<!-- --><br />
<!-- You should have received a copy of the GNU General Public License --><br />
<!-- along with this program; if not, write to the --><br />
<!-- Free Software Foundation, Inc., --><br />
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA --><br />
<!-- --><br />
<!-- Sipp 'uac_auth' scenario by Alessandro Marzini info@digilabit.com --><br />
<!-- --><br />
<scenario name="UAC with AUTH scenario"><br />
<!-- In client mode (sipp placing calls), the Call-ID MUST be --><br />
<!-- generated by sipp. To do so, use [call_id] keyword. --><br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 1 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]<br />
s=-<br />
c=IN IP[media_ip_type] [media_ip]<br />
t=0 0<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
]]><br />
</send><br />
<recv response="407" auth="true"><br />
</recv><br />
<send><br />
<![CDATA[<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 2 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
[field1]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP4 127.0.0.1<br />
s=-<br />
t=0 0<br />
c=IN IP4 [media_ip]<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
]]><br />
</send><br />
<recv response="100" optional="true"><br />
</recv><br />
<recv response="180" optional="true"><br />
</recv><br />
<recv response="183" optional="true"><br />
</recv><br />
<!-- By adding rrs="true" (Record Route Sets), the route sets --><br />
<!-- are saved and used for following messages sent. Useful to test --><br />
<!-- against stateful SIP proxies/B2BUAs. --><br />
<recv response="200" rtd="true"><br />
</recv><br />
<!-- Packet lost can be simulated in any send/recv message by --><br />
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. --><br />
<send><br />
<![CDATA[<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<!-- This delay can be customized by the -d command-line option --><br />
<!-- or by adding a 'milliseconds = "value"' option here. --><br />
<pause/><br />
<!-- The 'crlf' option inserts a blank line in the statistics report. --><br />
<send retrans="500"><br />
<![CDATA[<br />
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 2 BYE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<recv response="200" crlf="true"><br />
</recv><br />
<!-- definition of the response time repartition table (unit is ms) --><br />
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/><br />
<!-- definition of the call length repartition table (unit is ms) --><br />
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/><br />
</scenario><br />
<br />
Then run this:<br />
<br />
./sipp -inf users.csv -sf uac_auth.xml -s <extension_to_call> <mor_sip_server_sip><br />
<br />
For a more complete sipp syntax look here:<br />
<br />
http://sipp.sourceforge.net/doc1.1/reference.html#xmlsyntax</div>
Digilab
https://wiki.kolmisoft.com/index.php?title=Performance_Testing_with_Sipp&diff=1504
Performance Testing with Sipp
2008-06-15T21:25:25Z
<p>Digilab: </p>
<hr />
<div>Download and compile the last sipp:<br />
<br />
wget http://surfnet.dl.sourceforge.net/sourceforge/sipp/sipp.3.1.src.tar.gz<br />
<br />
tar xvfz sipp.3.1.src.tar.gz<br />
<br />
cd sipp.svn<br />
<br />
apt-get install libssl-dev<br />
<br />
make ossl<br />
<br />
edit a new file "users.csv":<br />
<br />
SEQUENTIAL<br />
<br />
user001;[authentication username=user001 password=password001]<br />
<br />
user002;[authentication username=user001 password=password001]<br />
<br />
user003;[authentication username=user001 password=password001]<br />
<br />
edit a new file "uac_auth.xml":<br />
<br />
<?xml version="1.0" encoding="ISO-8859-1" ?><br />
<!DOCTYPE scenario SYSTEM "sipp.dtd"><br />
<!-- This program is free software; you can redistribute it and/or --><br />
<!-- modify it under the terms of the GNU General Public License as --><br />
<!-- published by the Free Software Foundation; either version 2 of the --><br />
<!-- License, or (at your option) any later version. --><br />
<!-- --><br />
<!-- This program is distributed in the hope that it will be useful, --><br />
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of --><br />
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the --><br />
<!-- GNU General Public License for more details. --><br />
<!-- --><br />
<!-- You should have received a copy of the GNU General Public License --><br />
<!-- along with this program; if not, write to the --><br />
<!-- Free Software Foundation, Inc., --><br />
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA --><br />
<!-- --><br />
<!-- Sipp 'uac_auth' scenario by Alessandro Marzini info@digilabit.com --><br />
<!-- --><br />
<scenario name="UAC with AUTH scenario"><br />
<!-- In client mode (sipp placing calls), the Call-ID MUST be --><br />
<!-- generated by sipp. To do so, use [call_id] keyword. --><br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 1 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]<br />
s=-<br />
c=IN IP[media_ip_type] [media_ip]<br />
t=0 0<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
]]><br />
</send><br />
<recv response="407" auth="true"><br />
</recv><br />
<send><br />
<![CDATA[<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 2 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
[field1]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP4 127.0.0.1<br />
s=-<br />
t=0 0<br />
c=IN IP4 [media_ip]<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
]]><br />
</send><br />
<recv response="100" optional="true"><br />
</recv><br />
<recv response="180" optional="true"><br />
</recv><br />
<recv response="183" optional="true"><br />
</recv><br />
<!-- By adding rrs="true" (Record Route Sets), the route sets --><br />
<!-- are saved and used for following messages sent. Useful to test --><br />
<!-- against stateful SIP proxies/B2BUAs. --><br />
<recv response="200" rtd="true"><br />
</recv><br />
<!-- Packet lost can be simulated in any send/recv message by --><br />
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. --><br />
<send><br />
<![CDATA[<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<!-- This delay can be customized by the -d command-line option --><br />
<!-- or by adding a 'milliseconds = "value"' option here. --><br />
<pause/><br />
<!-- The 'crlf' option inserts a blank line in the statistics report. --><br />
<send retrans="500"><br />
<![CDATA[<br />
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 2 BYE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<recv response="200" crlf="true"><br />
</recv><br />
<!-- definition of the response time repartition table (unit is ms) --><br />
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/><br />
<!-- definition of the call length repartition table (unit is ms) --><br />
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/><br />
</scenario><br />
<br />
Then run this:<br />
<br />
./sipp -inf users.csv -sf uac_auth.xml -s <extension_to_call> <mor_sip_server_sip><br />
<br />
For a more complete sipp syntax look here:<br />
<br />
http://sipp.sourceforge.net/doc1.1/reference.html#xmlsyntax</div>
Digilab
https://wiki.kolmisoft.com/index.php?title=Performance_Testing_with_Sipp&diff=1503
Performance Testing with Sipp
2008-06-15T21:24:27Z
<p>Digilab: </p>
<hr />
<div>Download and compile the last sipp:<br />
<br />
wget http://surfnet.dl.sourceforge.net/sourceforge/sipp/sipp.3.1.src.tar.gz<br />
<br />
tar xvfz sipp.3.1.src.tar.gz<br />
<br />
cd sipp.svn<br />
<br />
apt-get install libssl-dev<br />
<br />
make ossl<br />
<br />
edit a new file "users.csv":<br />
<br />
SEQUENTIAL<br />
user001;[authentication username=user001 password=password001]<br />
user002;[authentication username=user001 password=password001]<br />
user003;[authentication username=user001 password=password001]<br />
<br />
edit a new file "uac_auth.xml":<br />
<br />
<?xml version="1.0" encoding="ISO-8859-1" ?><br />
<!DOCTYPE scenario SYSTEM "sipp.dtd"><br />
<!-- This program is free software; you can redistribute it and/or --><br />
<!-- modify it under the terms of the GNU General Public License as --><br />
<!-- published by the Free Software Foundation; either version 2 of the --><br />
<!-- License, or (at your option) any later version. --><br />
<!-- --><br />
<!-- This program is distributed in the hope that it will be useful, --><br />
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of --><br />
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the --><br />
<!-- GNU General Public License for more details. --><br />
<!-- --><br />
<!-- You should have received a copy of the GNU General Public License --><br />
<!-- along with this program; if not, write to the --><br />
<!-- Free Software Foundation, Inc., --><br />
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA --><br />
<!-- --><br />
<!-- Sipp 'uac_auth' scenario by Alessandro Marzini info@digilabit.com --><br />
<!-- --><br />
<scenario name="UAC with AUTH scenario"><br />
<!-- In client mode (sipp placing calls), the Call-ID MUST be --><br />
<!-- generated by sipp. To do so, use [call_id] keyword. --><br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 1 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]<br />
s=-<br />
c=IN IP[media_ip_type] [media_ip]<br />
t=0 0<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
]]><br />
</send><br />
<recv response="407" auth="true"><br />
</recv><br />
<send><br />
<![CDATA[<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 2 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
[field1]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP4 127.0.0.1<br />
s=-<br />
t=0 0<br />
c=IN IP4 [media_ip]<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
]]><br />
</send><br />
<recv response="100" optional="true"><br />
</recv><br />
<recv response="180" optional="true"><br />
</recv><br />
<recv response="183" optional="true"><br />
</recv><br />
<!-- By adding rrs="true" (Record Route Sets), the route sets --><br />
<!-- are saved and used for following messages sent. Useful to test --><br />
<!-- against stateful SIP proxies/B2BUAs. --><br />
<recv response="200" rtd="true"><br />
</recv><br />
<!-- Packet lost can be simulated in any send/recv message by --><br />
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. --><br />
<send><br />
<![CDATA[<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<!-- This delay can be customized by the -d command-line option --><br />
<!-- or by adding a 'milliseconds = "value"' option here. --><br />
<pause/><br />
<!-- The 'crlf' option inserts a blank line in the statistics report. --><br />
<send retrans="500"><br />
<![CDATA[<br />
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 2 BYE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<recv response="200" crlf="true"><br />
</recv><br />
<!-- definition of the response time repartition table (unit is ms) --><br />
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/><br />
<!-- definition of the call length repartition table (unit is ms) --><br />
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/><br />
</scenario><br />
<br />
Then run this:<br />
<br />
./sipp -inf users.csv -sf uac_auth.xml -s <extension_to_call> <mor_sip_server_sip><br />
<br />
For a more complete sipp syntax look here:<br />
<br />
http://sipp.sourceforge.net/doc1.1/reference.html#xmlsyntax</div>
Digilab
https://wiki.kolmisoft.com/index.php?title=Performance_Testing_with_Sipp&diff=1502
Performance Testing with Sipp
2008-06-15T21:10:31Z
<p>Digilab: </p>
<hr />
<div>Download and compile the last sipp:<br />
<br />
wget http://surfnet.dl.sourceforge.net/sourceforge/sipp/sipp.3.1.src.tar.gz<br />
<br />
tar xvfz sipp.3.1.src.tar.gz<br />
<br />
cd sipp.svn<br />
<br />
apt-get install libssl-dev<br />
<br />
make ossl<br />
<br />
<br />
<br />
<?xml version="1.0" encoding="ISO-8859-1" ?><br />
<!DOCTYPE scenario SYSTEM "sipp.dtd"><br />
<!-- This program is free software; you can redistribute it and/or --><br />
<!-- modify it under the terms of the GNU General Public License as --><br />
<!-- published by the Free Software Foundation; either version 2 of the --><br />
<!-- License, or (at your option) any later version. --><br />
<!-- --><br />
<!-- This program is distributed in the hope that it will be useful, --><br />
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of --><br />
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the --><br />
<!-- GNU General Public License for more details. --><br />
<!-- --><br />
<!-- You should have received a copy of the GNU General Public License --><br />
<!-- along with this program; if not, write to the --><br />
<!-- Free Software Foundation, Inc., --><br />
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA --><br />
<!-- --><br />
<!-- Sipp 'uac_auth' scenario by Alessandro Marzini info@digilabit.com --><br />
<!-- --><br />
<scenario name="UAC with AUTH scenario"><br />
<!-- In client mode (sipp placing calls), the Call-ID MUST be --><br />
<!-- generated by sipp. To do so, use [call_id] keyword. --><br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 1 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]<br />
s=-<br />
c=IN IP[media_ip_type] [media_ip]<br />
t=0 0<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
]]><br />
</send><br />
<recv response="407" auth="true"><br />
</recv><br />
<send><br />
<![CDATA[<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 2 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
[field1]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP4 127.0.0.1<br />
s=-<br />
t=0 0<br />
c=IN IP4 [media_ip]<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
]]><br />
</send><br />
<recv response="100" optional="true"><br />
</recv><br />
<recv response="180" optional="true"><br />
</recv><br />
<recv response="183" optional="true"><br />
</recv><br />
<!-- By adding rrs="true" (Record Route Sets), the route sets --><br />
<!-- are saved and used for following messages sent. Useful to test --><br />
<!-- against stateful SIP proxies/B2BUAs. --><br />
<recv response="200" rtd="true"><br />
</recv><br />
<!-- Packet lost can be simulated in any send/recv message by --><br />
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. --><br />
<send><br />
<![CDATA[<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<!-- This delay can be customized by the -d command-line option --><br />
<!-- or by adding a 'milliseconds = "value"' option here. --><br />
<pause/><br />
<!-- The 'crlf' option inserts a blank line in the statistics report. --><br />
<send retrans="500"><br />
<![CDATA[<br />
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 2 BYE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<recv response="200" crlf="true"><br />
</recv><br />
<!-- definition of the response time repartition table (unit is ms) --><br />
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/><br />
<!-- definition of the call length repartition table (unit is ms) --><br />
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/><br />
</scenario></div>
Digilab
https://wiki.kolmisoft.com/index.php?title=Performance_Testing_with_Sipp&diff=1501
Performance Testing with Sipp
2008-06-15T21:10:08Z
<p>Digilab: </p>
<hr />
<div>Download and compile the last sipp:<br />
<br />
wget http://surfnet.dl.sourceforge.net/sourceforge/sipp/sipp.3.1.src.tar.gz<br />
tar xvfz sipp.3.1.src.tar.gz<br />
cd sipp.svn<br />
apt-get install libssl-dev<br />
make ossl<br />
<br />
<br />
<br />
<?xml version="1.0" encoding="ISO-8859-1" ?><br />
<!DOCTYPE scenario SYSTEM "sipp.dtd"><br />
<!-- This program is free software; you can redistribute it and/or --><br />
<!-- modify it under the terms of the GNU General Public License as --><br />
<!-- published by the Free Software Foundation; either version 2 of the --><br />
<!-- License, or (at your option) any later version. --><br />
<!-- --><br />
<!-- This program is distributed in the hope that it will be useful, --><br />
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of --><br />
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the --><br />
<!-- GNU General Public License for more details. --><br />
<!-- --><br />
<!-- You should have received a copy of the GNU General Public License --><br />
<!-- along with this program; if not, write to the --><br />
<!-- Free Software Foundation, Inc., --><br />
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA --><br />
<!-- --><br />
<!-- Sipp 'uac_auth' scenario by Alessandro Marzini info@digilabit.com --><br />
<!-- --><br />
<scenario name="UAC with AUTH scenario"><br />
<!-- In client mode (sipp placing calls), the Call-ID MUST be --><br />
<!-- generated by sipp. To do so, use [call_id] keyword. --><br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 1 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]<br />
s=-<br />
c=IN IP[media_ip_type] [media_ip]<br />
t=0 0<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
]]><br />
</send><br />
<recv response="407" auth="true"><br />
</recv><br />
<send><br />
<![CDATA[<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 2 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
[field1]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP4 127.0.0.1<br />
s=-<br />
t=0 0<br />
c=IN IP4 [media_ip]<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
]]><br />
</send><br />
<recv response="100" optional="true"><br />
</recv><br />
<recv response="180" optional="true"><br />
</recv><br />
<recv response="183" optional="true"><br />
</recv><br />
<!-- By adding rrs="true" (Record Route Sets), the route sets --><br />
<!-- are saved and used for following messages sent. Useful to test --><br />
<!-- against stateful SIP proxies/B2BUAs. --><br />
<recv response="200" rtd="true"><br />
</recv><br />
<!-- Packet lost can be simulated in any send/recv message by --><br />
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. --><br />
<send><br />
<![CDATA[<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<!-- This delay can be customized by the -d command-line option --><br />
<!-- or by adding a 'milliseconds = "value"' option here. --><br />
<pause/><br />
<!-- The 'crlf' option inserts a blank line in the statistics report. --><br />
<send retrans="500"><br />
<![CDATA[<br />
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 2 BYE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<recv response="200" crlf="true"><br />
</recv><br />
<!-- definition of the response time repartition table (unit is ms) --><br />
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/><br />
<!-- definition of the call length repartition table (unit is ms) --><br />
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/><br />
</scenario></div>
Digilab
https://wiki.kolmisoft.com/index.php?title=Performance_Testing_with_Sipp&diff=1500
Performance Testing with Sipp
2008-06-15T21:09:31Z
<p>Digilab: </p>
<hr />
<div>Download and compile the last sipp:<br />
<br />
''wget http://surfnet.dl.sourceforge.net/sourceforge/sipp/sipp.3.1.src.tar.gz<br />
tar xvfz sipp.3.1.src.tar.gz<br />
cd sipp.svn<br />
apt-get install libssl-dev<br />
make ossl''<br />
<br />
<br />
<br />
<?xml version="1.0" encoding="ISO-8859-1" ?><br />
<!DOCTYPE scenario SYSTEM "sipp.dtd"><br />
<!-- This program is free software; you can redistribute it and/or --><br />
<!-- modify it under the terms of the GNU General Public License as --><br />
<!-- published by the Free Software Foundation; either version 2 of the --><br />
<!-- License, or (at your option) any later version. --><br />
<!-- --><br />
<!-- This program is distributed in the hope that it will be useful, --><br />
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of --><br />
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the --><br />
<!-- GNU General Public License for more details. --><br />
<!-- --><br />
<!-- You should have received a copy of the GNU General Public License --><br />
<!-- along with this program; if not, write to the --><br />
<!-- Free Software Foundation, Inc., --><br />
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA --><br />
<!-- --><br />
<!-- Sipp 'uac_auth' scenario by Alessandro Marzini info@digilabit.com --><br />
<!-- --><br />
<scenario name="UAC with AUTH scenario"><br />
<!-- In client mode (sipp placing calls), the Call-ID MUST be --><br />
<!-- generated by sipp. To do so, use [call_id] keyword. --><br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 1 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]<br />
s=-<br />
c=IN IP[media_ip_type] [media_ip]<br />
t=0 0<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
]]><br />
</send><br />
<recv response="407" auth="true"><br />
</recv><br />
<send><br />
<![CDATA[<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 2 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
[field1]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP4 127.0.0.1<br />
s=-<br />
t=0 0<br />
c=IN IP4 [media_ip]<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
]]><br />
</send><br />
<recv response="100" optional="true"><br />
</recv><br />
<recv response="180" optional="true"><br />
</recv><br />
<recv response="183" optional="true"><br />
</recv><br />
<!-- By adding rrs="true" (Record Route Sets), the route sets --><br />
<!-- are saved and used for following messages sent. Useful to test --><br />
<!-- against stateful SIP proxies/B2BUAs. --><br />
<recv response="200" rtd="true"><br />
</recv><br />
<!-- Packet lost can be simulated in any send/recv message by --><br />
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. --><br />
<send><br />
<![CDATA[<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<!-- This delay can be customized by the -d command-line option --><br />
<!-- or by adding a 'milliseconds = "value"' option here. --><br />
<pause/><br />
<!-- The 'crlf' option inserts a blank line in the statistics report. --><br />
<send retrans="500"><br />
<![CDATA[<br />
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 2 BYE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<recv response="200" crlf="true"><br />
</recv><br />
<!-- definition of the response time repartition table (unit is ms) --><br />
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/><br />
<!-- definition of the call length repartition table (unit is ms) --><br />
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/><br />
</scenario></div>
Digilab
https://wiki.kolmisoft.com/index.php?title=Performance_Testing_with_Sipp&diff=1499
Performance Testing with Sipp
2008-06-15T20:58:35Z
<p>Digilab: </p>
<hr />
<div><?xml version="1.0" encoding="ISO-8859-1" ?><br />
<!DOCTYPE scenario SYSTEM "sipp.dtd"><br />
<!-- This program is free software; you can redistribute it and/or --><br />
<!-- modify it under the terms of the GNU General Public License as --><br />
<!-- published by the Free Software Foundation; either version 2 of the --><br />
<!-- License, or (at your option) any later version. --><br />
<!-- --><br />
<!-- This program is distributed in the hope that it will be useful, --><br />
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of --><br />
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the --><br />
<!-- GNU General Public License for more details. --><br />
<!-- --><br />
<!-- You should have received a copy of the GNU General Public License --><br />
<!-- along with this program; if not, write to the --><br />
<!-- Free Software Foundation, Inc., --><br />
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA --><br />
<!-- --><br />
<!-- Sipp 'uac_auth' scenario by Alessandro Marzini info@digilabit.com --><br />
<!-- --><br />
<scenario name="UAC with AUTH scenario"><br />
<!-- In client mode (sipp placing calls), the Call-ID MUST be --><br />
<!-- generated by sipp. To do so, use [call_id] keyword. --><br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 1 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]<br />
s=-<br />
c=IN IP[media_ip_type] [media_ip]<br />
t=0 0<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
]]><br />
</send><br />
<recv response="407" auth="true"><br />
</recv><br />
<send><br />
<![CDATA[<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 2 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
[field1]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP4 127.0.0.1<br />
s=-<br />
t=0 0<br />
c=IN IP4 [media_ip]<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
]]><br />
</send><br />
<recv response="100" optional="true"><br />
</recv><br />
<recv response="180" optional="true"><br />
</recv><br />
<recv response="183" optional="true"><br />
</recv><br />
<!-- By adding rrs="true" (Record Route Sets), the route sets --><br />
<!-- are saved and used for following messages sent. Useful to test --><br />
<!-- against stateful SIP proxies/B2BUAs. --><br />
<recv response="200" rtd="true"><br />
</recv><br />
<!-- Packet lost can be simulated in any send/recv message by --><br />
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. --><br />
<send><br />
<![CDATA[<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<!-- This delay can be customized by the -d command-line option --><br />
<!-- or by adding a 'milliseconds = "value"' option here. --><br />
<pause/><br />
<!-- The 'crlf' option inserts a blank line in the statistics report. --><br />
<send retrans="500"><br />
<![CDATA[<br />
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 2 BYE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<recv response="200" crlf="true"><br />
</recv><br />
<!-- definition of the response time repartition table (unit is ms) --><br />
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/><br />
<!-- definition of the call length repartition table (unit is ms) --><br />
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/><br />
</scenario></div>
Digilab
https://wiki.kolmisoft.com/index.php?title=Performance_Testing_with_Sipp&diff=1498
Performance Testing with Sipp
2008-06-15T20:55:37Z
<p>Digilab: </p>
<hr />
<div><nowiki><br />
<?xml version="1.0" encoding="ISO-8859-1" ?><br />
<!DOCTYPE scenario SYSTEM "sipp.dtd"><br />
<!-- This program is free software; you can redistribute it and/or --><br />
<!-- modify it under the terms of the GNU General Public License as --><br />
<!-- published by the Free Software Foundation; either version 2 of the --><br />
<!-- License, or (at your option) any later version. --><br />
<!-- --><br />
<!-- This program is distributed in the hope that it will be useful, --><br />
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of --><br />
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the --><br />
<!-- GNU General Public License for more details. --><br />
<!-- --><br />
<!-- You should have received a copy of the GNU General Public License --><br />
<!-- along with this program; if not, write to the --><br />
<!-- Free Software Foundation, Inc., --><br />
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA --><br />
<!-- --><br />
<!-- Sipp 'uac_auth' scenario by Alessandro Marzini info@digilabit.com --><br />
<!-- --><br />
<scenario name="UAC with AUTH scenario"><br />
<!-- In client mode (sipp placing calls), the Call-ID MUST be --><br />
<!-- generated by sipp. To do so, use [call_id] keyword. --><br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 1 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]<br />
s=-<br />
c=IN IP[media_ip_type] [media_ip]<br />
t=0 0<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
]]><br />
</send><br />
<recv response="407" auth="true"><br />
</recv><br />
<send><br />
<![CDATA[<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 2 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
[field1]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP4 127.0.0.1<br />
s=-<br />
t=0 0<br />
c=IN IP4 [media_ip]<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
]]><br />
</send><br />
<recv response="100" optional="true"><br />
</recv><br />
<recv response="180" optional="true"><br />
</recv><br />
<recv response="183" optional="true"><br />
</recv><br />
<!-- By adding rrs="true" (Record Route Sets), the route sets --><br />
<!-- are saved and used for following messages sent. Useful to test --><br />
<!-- against stateful SIP proxies/B2BUAs. --><br />
<recv response="200" rtd="true"><br />
</recv><br />
<!-- Packet lost can be simulated in any send/recv message by --><br />
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. --><br />
<send><br />
<![CDATA[<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<!-- This delay can be customized by the -d command-line option --><br />
<!-- or by adding a 'milliseconds = "value"' option here. --><br />
<pause/><br />
<!-- The 'crlf' option inserts a blank line in the statistics report. --><br />
<send retrans="500"><br />
<![CDATA[<br />
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 2 BYE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<recv response="200" crlf="true"><br />
</recv><br />
<!-- definition of the response time repartition table (unit is ms) --><br />
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/><br />
<!-- definition of the call length repartition table (unit is ms) --><br />
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/><br />
</scenario><br />
</nowiki></div>
Digilab
https://wiki.kolmisoft.com/index.php?title=Performance_Testing_with_Sipp&diff=1497
Performance Testing with Sipp
2008-06-15T20:54:14Z
<p>Digilab: </p>
<hr />
<div><br />
<?xml version="1.0" encoding="ISO-8859-1" ?><br />
<!DOCTYPE scenario SYSTEM "sipp.dtd"><br />
<!-- This program is free software; you can redistribute it and/or --><br />
<!-- modify it under the terms of the GNU General Public License as --><br />
<!-- published by the Free Software Foundation; either version 2 of the --><br />
<!-- License, or (at your option) any later version. --><br />
<!-- --><br />
<!-- This program is distributed in the hope that it will be useful, --><br />
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of --><br />
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the --><br />
<!-- GNU General Public License for more details. --><br />
<!-- --><br />
<!-- You should have received a copy of the GNU General Public License --><br />
<!-- along with this program; if not, write to the --><br />
<!-- Free Software Foundation, Inc., --><br />
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA --><br />
<!-- --><br />
<!-- Sipp 'uac_auth' scenario by Alessandro Marzini info@digilabit.com --><br />
<!-- --><br />
<scenario name="UAC with AUTH scenario"><br />
<!-- In client mode (sipp placing calls), the Call-ID MUST be --><br />
<!-- generated by sipp. To do so, use [call_id] keyword. --><br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 1 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]<br />
s=-<br />
c=IN IP[media_ip_type] [media_ip]<br />
t=0 0<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
]]><br />
</send><br />
<recv response="407" auth="true"><br />
</recv><br />
<send><br />
<![CDATA[<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 2 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
[field1]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP4 127.0.0.1<br />
s=-<br />
t=0 0<br />
c=IN IP4 [media_ip]<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
]]><br />
</send><br />
<recv response="100" optional="true"><br />
</recv><br />
<recv response="180" optional="true"><br />
</recv><br />
<recv response="183" optional="true"><br />
</recv><br />
<!-- By adding rrs="true" (Record Route Sets), the route sets --><br />
<!-- are saved and used for following messages sent. Useful to test --><br />
<!-- against stateful SIP proxies/B2BUAs. --><br />
<recv response="200" rtd="true"><br />
</recv><br />
<!-- Packet lost can be simulated in any send/recv message by --><br />
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. --><br />
<send><br />
<![CDATA[<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<!-- This delay can be customized by the -d command-line option --><br />
<!-- or by adding a 'milliseconds = "value"' option here. --><br />
<pause/><br />
<!-- The 'crlf' option inserts a blank line in the statistics report. --><br />
<send retrans="500"><br />
<![CDATA[<br />
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 2 BYE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
]]><br />
</send><br />
<recv response="200" crlf="true"><br />
</recv><br />
<!-- definition of the response time repartition table (unit is ms) --><br />
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/><br />
<!-- definition of the call length repartition table (unit is ms) --><br />
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/><br />
</scenario></div>
Digilab
https://wiki.kolmisoft.com/index.php?title=Performance_Testing_with_Sipp&diff=1496
Performance Testing with Sipp
2008-06-15T20:50:24Z
<p>Digilab: </p>
<hr />
<div><br />
<br />
<?xml version="1.0" encoding="ISO-8859-1" ?><br />
<!DOCTYPE scenario SYSTEM "sipp.dtd"><br />
<br />
<!-- This program is free software; you can redistribute it and/or --><br />
<!-- modify it under the terms of the GNU General Public License as --><br />
<!-- published by the Free Software Foundation; either version 2 of the --><br />
<!-- License, or (at your option) any later version. --><br />
<!-- --><br />
<!-- This program is distributed in the hope that it will be useful, --><br />
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of --><br />
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the --><br />
<!-- GNU General Public License for more details. --><br />
<!-- --><br />
<!-- You should have received a copy of the GNU General Public License --><br />
<!-- along with this program; if not, write to the --><br />
<!-- Free Software Foundation, Inc., --><br />
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA --><br />
<!-- --><br />
<!-- Sipp 'uac_auth' scenario by Alessandro Marzini info@digilabit.com --><br />
<!-- --><br />
<br />
<scenario name="UAC with AUTH scenario"><br />
<!-- In client mode (sipp placing calls), the Call-ID MUST be --><br />
<!-- generated by sipp. To do so, use [call_id] keyword. --><br />
<br />
<send retrans="500"><br />
<![CDATA[<br />
<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 1 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]<br />
s=-<br />
c=IN IP[media_ip_type] [media_ip]<br />
t=0 0<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
<br />
]]><br />
</send><br />
<br />
<recv response="407" auth="true"><br />
</recv><br />
<br />
<send><br />
<![CDATA[<br />
<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
<br />
]]><br />
</send><br />
<br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 2 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
[field1]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
<br />
v=0<br />
o=user1 53655765 2353687637 IN IP4 127.0.0.1<br />
s=-<br />
t=0 0<br />
c=IN IP4 [media_ip]<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
<br />
]]><br />
</send><br />
<br />
<recv response="100" optional="true"><br />
</recv><br />
<br />
<recv response="180" optional="true"><br />
</recv><br />
<br />
<recv response="183" optional="true"><br />
</recv><br />
<br />
<!-- By adding rrs="true" (Record Route Sets), the route sets --><br />
<!-- are saved and used for following messages sent. Useful to test --><br />
<!-- against stateful SIP proxies/B2BUAs. --><br />
<br />
<recv response="200" rtd="true"><br />
</recv><br />
<br />
<!-- Packet lost can be simulated in any send/recv message by --><br />
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. --><br />
<br />
<send><br />
<![CDATA[<br />
<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
<br />
]]><br />
</send><br />
<br />
<!-- This delay can be customized by the -d command-line option --><br />
<!-- or by adding a 'milliseconds = "value"' option here. --><br />
<br />
<pause/><br />
<br />
<!-- The 'crlf' option inserts a blank line in the statistics report. --><br />
<br />
<send retrans="500"><br />
<![CDATA[<br />
<br />
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 2 BYE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
<br />
]]><br />
</send><br />
<br />
<recv response="200" crlf="true"><br />
</recv><br />
<br />
<!-- definition of the response time repartition table (unit is ms) --><br />
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/><br />
<br />
<!-- definition of the call length repartition table (unit is ms) --><br />
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/><br />
<br />
</scenario></div>
Digilab
https://wiki.kolmisoft.com/index.php?title=Performance_Testing_with_Sipp&diff=1493
Performance Testing with Sipp
2008-06-15T20:44:25Z
<p>Digilab: New page: <?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 ter...</p>
<hr />
<div><?xml version="1.0" encoding="ISO-8859-1" ?><br />
<!DOCTYPE scenario SYSTEM "sipp.dtd"><br />
<br />
<!-- This program is free software; you can redistribute it and/or --><br />
<!-- modify it under the terms of the GNU General Public License as --><br />
<!-- published by the Free Software Foundation; either version 2 of the --><br />
<!-- License, or (at your option) any later version. --><br />
<!-- --><br />
<!-- This program is distributed in the hope that it will be useful, --><br />
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of --><br />
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the --><br />
<!-- GNU General Public License for more details. --><br />
<!-- --><br />
<!-- You should have received a copy of the GNU General Public License --><br />
<!-- along with this program; if not, write to the --><br />
<!-- Free Software Foundation, Inc., --><br />
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA --><br />
<!-- --><br />
<!-- Sipp 'uac_auth' scenario by Alessandro Marzini info@digilabit.com --><br />
<!-- --><br />
<br />
<scenario name="UAC with AUTH scenario"><br />
<!-- In client mode (sipp placing calls), the Call-ID MUST be --><br />
<!-- generated by sipp. To do so, use [call_id] keyword. --><br />
<send retrans="500"><br />
<![CDATA[<br />
<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 1 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
v=0<br />
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]<br />
s=-<br />
c=IN IP[media_ip_type] [media_ip]<br />
t=0 0<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
<br />
]]><br />
</send><br />
<br />
<recv response="407" auth="true"><br />
</recv><br />
<br />
<send><br />
<![CDATA[<br />
<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
<br />
]]><br />
</send><br />
<br />
<send retrans="500"><br />
<![CDATA[<br />
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]><br />
Call-ID: [call_id]<br />
CSeq: 2 INVITE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
[field1]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Type: application/sdp<br />
Content-Length: [len]<br />
<br />
v=0<br />
o=user1 53655765 2353687637 IN IP4 127.0.0.1<br />
s=-<br />
t=0 0<br />
c=IN IP4 [media_ip]<br />
m=audio [media_port] RTP/AVP 0<br />
a=rtpmap:0 PCMU/8000<br />
<br />
]]><br />
</send><br />
<br />
<recv response="100"<br />
optional="true"><br />
</recv><br />
<br />
<recv response="180" optional="true"><br />
</recv><br />
<br />
<recv response="183" optional="true"><br />
</recv><br />
<br />
<!-- By adding rrs="true" (Record Route Sets), the route sets --><br />
<!-- are saved and used for following messages sent. Useful to test --><br />
<!-- against stateful SIP proxies/B2BUAs. --><br />
<recv response="200" rtd="true"><br />
</recv><br />
<br />
<!-- Packet lost can be simulated in any send/recv message by --><br />
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. --><br />
<send><br />
<![CDATA[<br />
<br />
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 1 ACK<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
<br />
]]><br />
</send><br />
<br />
<!-- This delay can be customized by the -d command-line option --><br />
<!-- or by adding a 'milliseconds = "value"' option here. --><br />
<pause/><br />
<br />
<!-- The 'crlf' option inserts a blank line in the statistics report. --><br />
<send retrans="500"><br />
<![CDATA[<br />
<br />
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br />
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br />
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]<br />
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br />
Call-ID: [call_id]<br />
CSeq: 2 BYE<br />
Contact: sip:[field0]@[local_ip]:[local_port]<br />
Max-Forwards: 70<br />
Subject: Performance Test<br />
Content-Length: 0<br />
<br />
]]><br />
</send><br />
<br />
<recv response="200" crlf="true"><br />
</recv><br />
<br />
<!-- definition of the response time repartition table (unit is ms) --><br />
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/><br />
<br />
<!-- definition of the call length repartition table (unit is ms) --><br />
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/><br />
<br />
</scenario></div>
Digilab