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.                    -->
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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 'uac' scenario with pcap (rtp) play           -->
20. <!--                                                                    -->
21.  
22. <scenario name="UAC with media">
23.   <!-- In client mode (sipp placing calls), the Call-ID MUST be         -->
24.   <!-- generated by sipp. To do so, use [call_id] keyword.                -->
25.   <send retrans="500">
26.     <![CDATA[
27.  
28.       INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
29.       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
30.       From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
31.       To: sut <sip:[service]@[remote_ip]:[remote_port]>
32.       Call-ID: [call_id]
33.       CSeq: 1 INVITE
34.       Contact: sip:sipp@[local_ip]:[local_port]
35.       Max-Forwards: 70
36.       Subject: Performance Test
37.       Content-Type: application/sdp
38.       Content-Length: [len]
39.  
40.       v=0
41.       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
42.       s=-
43.       c=IN IP[local_ip_type] [local_ip]
44.       t=0 0
45.       m=audio [auto_media_port] RTP/AVP 8
46.       a=rtpmap:8 G722/8000
47.       a=rtpmap:101 telephone-event/8000
48.       a=fmtp:101 0-11,16
49.  
50.     ]]>
51.   </send>
52.  
53.   <recv response="100" optional="true">
54.   </recv>
55.  
56.   <recv response="180" optional="true">
57.   </recv>
58.  
59.   <recv response="183" optional="true">
60.   </recv>
61.  
62.   <!-- By adding rrs="true" (Record Route Sets), the route sets         -->
63.   <!-- are saved and used for following messages sent. Useful to test   -->
64.   <!-- against stateful SIP proxies/B2BUAs.                             -->
65.   <recv response="200" rtd="true" crlf="true">
66.   </recv>
67.  
68.   <!-- Packet lost can be simulated in any send/recv message by         -->
69.   <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
70.   <send>
71.     <![CDATA[
72.  
73.       ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
74.       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
75.       From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
76.       To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
77.       Call-ID: [call_id]
78.       CSeq: 1 ACK
79.       Contact: sip:sipp@[local_ip]:[local_port]
80.       Max-Forwards: 70
81.       Subject: Performance Test
82.       Content-Length: 0
83.  
84.     ]]>
85.   </send>
86.  
87.   <!-- Play a pre-recorded PCAP file (RTP stream)                       -->
88.   <nop>
89.     <action>
90.       <exec play_pcap_audio="pcap_afterfix_rtp.pcap"/>
91.     </action>
92.   </nop>
93.  
94.   <!-- Pause 8 seconds, which is approximately the duration of the      -->
95.   <!-- PCAP file                                                        -->
96.   <pause milliseconds="2000000"/>
97.  
98.   <!-- Play an out of band DTMF '1'                                    
99.   <nop>
100.     <action>
101.       <exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/>
102.     </action>
103.   </nop>
104.   -->
105.  
106.   <pause milliseconds="1000"/>
107.  
108.   <!-- The 'crlf' option inserts a blank line in the statistics report. -->
109.   <send retrans="500">
110.     <![CDATA[
111.  
112.       BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
113.       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
114.       From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
115.       To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
116.       Call-ID: [call_id]
117.       CSeq: 2 BYE
118.       Contact: sip:sipp@[local_ip]:[local_port]
119.       Max-Forwards: 70
120.       Subject: Performance Test
121.       Content-Length: 0
122.  
123.     ]]>
124.   </send>
125.  
126.   <recv response="200" crlf="true">
127.   </recv>
128.  
129.   <!-- definition of the response time repartition table (unit is ms)   -->
130.   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
131.  
132.   <!-- definition of the call length repartition table (unit is ms)     -->
133.   <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
134.  
135. </scenario>
136.  
137.