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sipp_uac_test_inv.xml
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sipp_uac_test_inv.xml
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<?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:[email protected] SIP/2.0
From: <sip:[email protected]>;tag=e27b3
To: <sip:[email protected]>
Call-Id: scb2f9c3f297d0197b29f4591169fcc23
Cseq: 25077 INVITE
Session-Expires: 1800
Min-Expires: 90
Content-Type: application/sdp
Content-Length: 375
Expires: 180
Date: Tue, 01 Mar 2011 04:19:12 GMT
Max-Forwards: 69
User-Agent: wxCommunicator
Accept-Language: en
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,INFO,PRACK,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REGISTER,REFER
Supported: replaces,timer,100rel,from-change,norefersub
Via: SIP/2.0/UDP 173.170.11.84:5060;branch=z9hG4bK-83c6ce099e7d;rport
Contact: <sip:[email protected]:5060>
v=0
o=sipX 5 10 IN IP4 192.168.1.138
s=call
c=IN IP4 192.168.1.138
t=0 0
m=audio 9000 RTP/AVP 96 97
a=candidate:0 t UDP 1.0 192.168.1.138 9000
a=candidate:0 t UDP 1.0 192.168.1.138 9001
a=candidate:1 t UDP 0.5 5.210.195.106 9000
a=candidate:1 t UDP 0.5 5.210.195.106 9001
a=rtpmap:96 telephone-event/8000/1
a=rtpmap:97 speex/32000/1
a=fmtp:97 mode=4
a=ptime:20
]]>
</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-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>