1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
|
<?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 'uas' scenario. -->
<!-- -->
<scenario name="Sending re-INVITE and ACK (with same branch) without SDP (#1045)">
<!-- 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 request="INVITE" crlf="true">
<action>
<ereg regexp=".*" search_in="hdr" header="From" assign_to="3"/>
<ereg regexp="sip:(.*)>" search_in="hdr" header="Contact" assign_to="4,5"/>
<assign assign_to="4" variable="5" />
</action>
</recv>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: sip:sipp@[local_ip]:[local_port]
Content-Type: application/sdp
Content-Length: [len]
v=0
o=- 3441953879 3441953879 IN IP4 192.168.0.15
s=pjmedia
c=IN IP4 192.168.0.15
t=0 0
m=audio 4004 RTP/AVP 0
]]>
</send>
<recv request="ACK" crlf="true">
</recv>
<pause milliseconds="2000"/>
<send retrans="500">
<![CDATA[
INVITE sip:[$5] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-same-branch
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To[$3]
Call-ID: [call_id]
Cseq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Content-Length: 0
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="200" rtd="true">
</recv>
<send>
<![CDATA[
ACK sip:[$5] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-same-branch
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To[$3]
Call-ID: [call_id]
Cseq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Content-Length: 0
]]>
</send>
<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the BYE again. -->
<pause milliseconds="4000"/>
<!-- 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>
|
