Ticket #641: uas-srtp.xml

File uas-srtp.xml, 5.2 KB (added by bennylp, 16 years ago)

SIPp scenario file

Line 
1<?xml version="1.0" encoding="ISO-8859-1" ?>
2<!DOCTYPE scenario SYSTEM "sipp.dtd">
3
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.                    -->
8<!--                                                                    -->
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.                       -->
13<!--                                                                    -->
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             -->
18<!--                                                                    -->
19<!--                 Sipp default 'uas' scenario.                       -->
20<!--                                                                    -->
21
22<scenario name="Basic UAS responder">
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">
27        <action>
28                <ereg regexp=".*" search_in="hdr" header="Via:" check_it="true", assign_to="10" />
29                <ereg regexp=".*" search_in="hdr" header="CSeq:" check_it="true", assign_to="11" />
30        </action>
31  </recv>
32
33  <!-- The '[last_*]' keyword is replaced automatically by the          -->
34  <!-- specified header if it was present in the last message received  -->
35  <!-- (except if it was a retransmission). If the header was not       -->
36  <!-- present or if no message has been received, the '[last_*]'       -->
37  <!-- keyword is discarded, and all bytes until the end of the line    -->
38  <!-- are also discarded.                                              -->
39  <!--                                                                  -->
40  <!-- If the specified header was present several times in the         -->
41  <!-- message, all occurences are concatenated (CRLF seperated)        -->
42  <!-- to be used in place of the '[last_*]' keyword.                   -->
43
44  <send>
45    <![CDATA[
46
47      SIP/2.0 180 Ringing
48      [last_Via:]
49      [last_From:]
50      [last_To:];tag=[call_number]
51      [last_Call-ID:]
52      [last_CSeq:]
53      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
54      Content-Length: 0
55
56    ]]>
57  </send>
58
59  <send retrans="500">
60    <![CDATA[
61
62      SIP/2.0 200 OK
63      [last_Via:]
64      [last_From:]
65      [last_To:];tag=[call_number]
66      [last_Call-ID:]
67      [last_CSeq:]
68      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
69      Content-Type: application/sdp
70      Content-Length: [len]
71
72      v=0
73      o=userX 20000001 20000001 IN IP4 192.168.27.101
74      s=pjmedia
75      c=IN IP4 192.168.27.101
76      t=0 0
77      m=audio 10600 RTP/SAVP 0 8 101
78      a=rtpmap:0 PCMU/8000/1
79      a=rtpmap:8 PCMA/8000/1
80      a=rtpmap:101 telephone-event/8000/1
81
82    ]]>
83  </send>
84
85  <recv request="BYE" optional="false">
86  </recv>
87
88  <recv request="ACK"
89        optional="false"
90        rtd="true"
91        crlf="true">
92  </recv>
93
94  <recv request="BYE" optional="false">
95  </recv>
96
97  <send>
98    <![CDATA[
99
100      SIP/2.0 404 Not Found
101      [last_Via:]
102      [last_From:]
103      [last_To:]
104      [last_Call-ID:]
105      [last_CSeq:]
106      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
107      Content-Length: 0
108
109    ]]>
110  </send>
111
112  <send>
113    <![CDATA[
114
115      SIP/2.0 200 OK
116      Via: [$10]
117      [last_From:]
118      [last_To:];tag=[call_number]
119      [last_Call-ID:]
120      CSeq: [$11]
121      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
122      Content-Type: application/sdp
123      Content-Length: [len]
124
125      v=0
126      o=userX 20000001 20000001 IN IP4 192.168.27.101
127      s=pjmedia
128      c=IN IP4 192.168.27.101
129      t=0 0
130      m=audio 10600 RTP/SAVP 0 8 101
131      a=rtpmap:0 PCMU/8000/1
132      a=rtpmap:8 PCMA/8000/1
133      a=rtpmap:101 telephone-event/8000/1
134
135    ]]>
136  </send>
137
138  <recv request="ACK"
139        optional="false"
140        rtd="true"
141        crlf="true">
142  </recv>
143
144  <!-- Keep the call open for a while in case the 200 is lost to be     -->
145  <!-- able to retransmit it if we receive the BYE again.               -->
146  <pause milliseconds="4000"/>
147
148
149  <!-- definition of the response time repartition table (unit is ms)   -->
150  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
151
152  <!-- definition of the call length repartition table (unit is ms)     -->
153  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
154
155</scenario>
156