1
<?xml version="1.0" encoding="ISO-8859-1" ?>
2
<!DOCTYPE scenario SYSTEM "sipp.dtd">
4
<!-- This program is free software; you can redistribute it and/or -->
5
<!-- modify it under the terms of the GNU General Public License as -->
6
<!-- published by the Free Software Foundation; either version 2 of the -->
7
<!-- License, or (at your option) any later version. -->
9
<!-- This program is distributed in the hope that it will be useful, -->
10
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
11
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
12
<!-- GNU General Public License for more details. -->
14
<!-- You should have received a copy of the GNU General Public License -->
15
<!-- along with this program; if not, write to the -->
16
<!-- Free Software Foundation, Inc., -->
17
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
19
<!-- Sipp default 'uas' scenario. -->
22
<scenario name="UAS answer multiple formats, UAS supports UPDATE method">
23
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
24
<!-- are saved and used for following messages sent. Useful to test -->
25
<!-- against stateful SIP proxies/B2BUAs. -->
26
<recv request="INVITE" crlf="true">
28
<ereg regexp=".*" search_in="hdr" header="From" assign_to="3"/>
29
<ereg regexp="sip:(.*)>" search_in="hdr" header="Contact" assign_to="4,5"/>
30
<assign assign_to="4" variable="5" />
34
<!-- The '[last_*]' keyword is replaced automatically by the -->
35
<!-- specified header if it was present in the last message received -->
36
<!-- (except if it was a retransmission). If the header was not -->
37
<!-- present or if no message has been received, the '[last_*]' -->
38
<!-- keyword is discarded, and all bytes until the end of the line -->
39
<!-- are also discarded. -->
41
<!-- If the specified header was present several times in the -->
42
<!-- message, all occurences are concatenated (CRLF seperated) -->
43
<!-- to be used in place of the '[last_*]' keyword. -->
51
[last_To:];tag=[call_number]
54
Contact: sip:sipp@[local_ip]:[local_port]
55
Content-Type: application/sdp
57
Allow: INVITE, UPDATE, ACK, BYE
60
o=- 3441953879 3441953879 IN IP4 192.168.0.15
64
m=audio 4004 RTP/AVP 0 8 3 111
68
a=rtpmap:111 telephone-event/8000
74
<recv request="ACK" crlf="true">
79
<recv request="UPDATE" crlf="true">
88
[last_To:];tag=[call_number]
91
Contact: sip:sipp@[local_ip]:[local_port]
92
Content-Type: application/sdp
94
Allow: INVITE, UPDATE, ACK, BYE
97
o=- 3441953879 3441953879 IN IP4 192.168.0.15
101
m=audio 4004 RTP/AVP 0 111
103
a=rtpmap:111 telephone-event/8000
109
<pause milliseconds="2000"/>
115
Via: SIP/2.0/[transport] [local_ip]:[local_port]
116
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
120
Contact: sip:sipp@[local_ip]:[local_port]
127
<recv response="200">
131
<!-- definition of the response time repartition table (unit is ms) -->
132
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
134
<!-- definition of the call length repartition table (unit is ms) -->
135
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>