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 -->
22
For this test to work, PJSUA-LIB needs to add video line, with
25
pjsua_media.c:1253, after call to pjmedia_endpt_create_sdp():
28
pjmedia_sdp_media *m = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_media);
29
m->desc.media = pj_str("video");
31
m->desc.transport = pj_str("RTP/AVP");
32
m->desc.fmt_count = 1;
33
m->desc.fmt[0] = pj_str("0");
34
sdp->media[sdp->media_count++] = m;
40
<scenario name="UAC with bad ACK">
41
<!-- UAC with bad ACK causes assertion with pjsip 1.4 -->
45
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
46
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
47
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
48
To: sut <sip:[service]@[remote_ip]:[remote_port]>
51
Contact: sip:sipp@[local_ip]:[local_port]
53
Subject: Performance Test
54
Content-Type: application/sdp
58
o=Tester 234 123 IN IP4 89.208.145.194
60
c=IN IP4 89.208.145.194
62
m=audio 17424 RTP/AVP 111 0 18 101
63
a=rtpmap:111 SPEEX/16000
66
a=rtpmap:101 telephone-event/8000
69
m=video 11128 RTP/AVP 34 103 104
70
a=rtpmap:34 H263/90000
71
a=rtpmap:103 H263-1998/90000
72
a=rtpmap:104 H264/90000
83
<recv response="180" optional="true">
86
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
87
<!-- are saved and used for following messages sent. Useful to test -->
88
<!-- against stateful SIP proxies/B2BUAs. -->
89
<recv response="200" rtd="true">
92
<!-- Packet lost can be simulated in any send/recv message by -->
93
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
97
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
98
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
99
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
100
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
103
Contact: sip:sipp@[local_ip]:[local_port]
105
Subject: Performance Test
111
<!-- This delay can be customized by the -d command-line option -->
112
<!-- or by adding a 'milliseconds = "value"' option here. -->
113
<pause milliseconds="2000"/>
118
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
119
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
120
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
121
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
124
Contact: sip:sipp@[local_ip]:[local_port]
126
Subject: Performance Test
127
Content-Type: application/sdp
128
Content-Length: [len]
131
o=Tester 234 124 IN IP4 89.208.145.194
133
c=IN IP4 89.208.145.194
135
m=audio 17424 RTP/AVP 111 0 18 101
136
a=rtpmap:111 SPEEX/16000
138
a=rtpmap:18 G729/8000
139
a=rtpmap:101 telephone-event/8000
142
m=video 0 RTP/AVP 34 103 104
149
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
150
<!-- are saved and used for following messages sent. Useful to test -->
151
<!-- against stateful SIP proxies/B2BUAs. -->
152
<recv response="200" rtd="true">
155
<!-- Packet lost can be simulated in any send/recv message by -->
156
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
160
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
161
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
162
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
163
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
166
Contact: sip:sipp@[local_ip]:[local_port]
168
Subject: Performance Test
175
<pause milliseconds="2000"/>
178
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
182
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
183
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
184
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
185
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
188
Contact: sip:sipp@[local_ip]:[local_port]
190
Subject: Performance Test
196
<recv response="200" crlf="true">
200
<!-- definition of the response time repartition table (unit is ms) -->
201
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
203
<!-- definition of the call length repartition table (unit is ms) -->
204
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>