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SEQUENTIAL | ||
# user, domain, authentication | ||
test;pbx.saghul.lan;[authentication username=test password=test] | ||
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<?xml version="1.0" encoding="ISO-8859-1" ?> | ||
<!DOCTYPE scenario SYSTEM "sipp.dtd"> | ||
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<!-- 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. --> | ||
<!-- --> | ||
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<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. --> | ||
<!-- | ||
Content-Type: application/sdp | ||
--> | ||
<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: 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 123 | ||
a=rtpmap:123 SAGHUL/8000 | ||
]]> | ||
</send> | ||
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<recv response="100" | ||
optional="true"> | ||
</recv> | ||
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<recv response="180" optional="true"> | ||
</recv> | ||
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<recv response="183" optional="true"> | ||
</recv> | ||
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<!-- 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> | ||
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<!-- 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> | ||
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<!-- This delay can be customized by the -d command-line option --> | ||
<!-- or by adding a 'milliseconds = "value"' option here. --> | ||
<pause/> | ||
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<!-- 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-Length: 0 | ||
]]> | ||
</send> | ||
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<recv response="200" crlf="true"> | ||
</recv> | ||
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<!-- definition of the response time repartition table (unit is ms) --> | ||
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/> | ||
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<!-- definition of the call length repartition table (unit is ms) --> | ||
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> | ||
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</scenario> | ||
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<?xml version="1.0" encoding="ISO-8859-1" ?> | ||
<!DOCTYPE scenario SYSTEM "sipp.dtd"> | ||
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||
<!-- 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. --> | ||
<!-- --> | ||
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||
<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. --> | ||
<!-- | ||
Content-Type: application/sdp | ||
--> | ||
<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-Length: 0 | ||
]]> | ||
</send> | ||
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<recv response="100" | ||
optional="true"> | ||
</recv> | ||
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<recv response="180" optional="true"> | ||
</recv> | ||
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<recv response="183" optional="true"> | ||
</recv> | ||
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||
<!-- 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> | ||
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<!-- 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-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> | ||
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<!-- This delay can be customized by the -d command-line option --> | ||
<!-- or by adding a 'milliseconds = "value"' option here. --> | ||
<pause/> | ||
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<!-- 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-Length: 0 | ||
]]> | ||
</send> | ||
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<recv response="200" crlf="true"> | ||
</recv> | ||
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<!-- definition of the response time repartition table (unit is ms) --> | ||
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/> | ||
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<!-- definition of the call length repartition table (unit is ms) --> | ||
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> | ||
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</scenario> | ||
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