blob: fe5169bb816c74ea6779f5598e0d168f8f780df5 [file] [log] [blame]
Tristan Matthews0a329cc2013-07-17 13:20:14 -04001<?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
20<!-- -->
21<!-- Session timer where UAS doesn't indicate support for UPDATE. -->
22<!-- In this case, UAC MUST use re-INVITE with SDP. -->
23
24<scenario name="Basic UAS responder">
25 <recv request="INVITE" crlf="true">
26 </recv>
27
28 <send retrans="500">
29 <![CDATA[
30
31 SIP/2.0 200 OK
32 [last_Via:]
33 [last_From:]
34 [last_To:];tag=[call_number]
35 [last_Call-ID:]
36 [last_CSeq:]
37 Contact: <sip:[local_ip]:[local_port];transport=[transport]>
38 Require: timer
39 Session-Expires: 90;refresher=uac
40 Content-Type: application/sdp
41 Content-Length: [len]
42
43 v=0
44 o=Some-UserAgent 68 210 IN IP4 [local_ip]
45 s=SIP Call
46 c=IN IP4 [local_ip]
47 t=0 0
48 m=audio 17294 RTP/AVP 0 101
49 c=IN IP4 [local_ip]
50 a=rtpmap:101 telephone-event/8000
51 a=fmtp:101 0-16
52 ]]>
53 </send>
54
55 <recv request="ACK"
56 optional="true"
57 rtd="true"
58 crlf="true">
59 </recv>
60
61 <recv request="INVITE" crlf="true">
62 </recv>
63
64 <send retrans="500">
65 <![CDATA[
66
67 SIP/2.0 200 OK
68 [last_Via:]
69 [last_From:]
70 [last_To:];tag=[call_number]
71 [last_Call-ID:]
72 [last_CSeq:]
73 Contact: <sip:[local_ip]:[local_port];transport=[transport]>
74 Require: timer
75 Session-Expires: 90;refresher=uac
76 Content-Type: application/sdp
77 Content-Length: [len]
78
79 v=0
80 o=Some-UserAgent 68 210 IN IP4 [local_ip]
81 s=SIP Call
82 c=IN IP4 [local_ip]
83 t=0 0
84 m=audio 17294 RTP/AVP 0 101
85 c=IN IP4 [local_ip]
86 a=rtpmap:101 telephone-event/8000
87 a=fmtp:101 0-16
88 ]]>
89 </send>
90
91 <recv request="ACK"
92 rtd="true"
93 crlf="true">
94 </recv>
95
96
97 <!-- Keep the call open for a while in case the 200 is lost to be -->
98 <!-- able to retransmit it if we receive the BYE again. -->
99 <pause milliseconds="4000"/>
100
101 <!-- definition of the response time repartition table (unit is ms) -->
102 <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
103
104 <!-- definition of the call length repartition table (unit is ms) -->
105 <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
106
107</scenario>
108