forked from pbertera/SIPp-by-example
-
Notifications
You must be signed in to change notification settings - Fork 0
/
uas-hold.xml
158 lines (129 loc) · 3.56 KB
/
uas-hold.xml
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
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic UAS responder">
<recv request="INVITE" crlf="true">
</recv>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=sipp 87308505 1 IN IP[local_ip_type] [local_ip]
s=-
t=0 0
m=audio [media_port] RTP/AVP 8 101
c=IN IP[media_ip_type] [media_ip]
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
]]>
</send>
<recv request="ACK" crlf="true">
</recv>
<nop display=">>>> Sending media <<<<<" crlf="true">
<action>
<exec play_pcap_audio="../pcap/g711a.pcap"/>
</action>
</nop>
<!-- the call hold is received with a=sendonly -->
<recv request="INVITE" rtd="true" crlf="true">
</recv>
<nop display=">>>>> Received hold re-INVITE <<<<" crlf="true">
</nop>
<!-- answer with SDP with a=recvonly -->
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=sipp 87308505 2 IN IP[local_ip_type] [local_ip]
s=-
t=0 0
m=audio [media_port] RTP/AVP 8 101
c=IN IP[media_ip_type] [media_ip]
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=recvonly
]]>
</send>
<recv request="ACK">
</recv>
<!-- receive the call retrieve with a=sendrecv -->
<recv request="INVITE" rtd="true" crlf="true">
</recv>
<nop display=">>>>> Received retreive re-INVITE <<<<<" crlf="true">
</nop>
<!-- accept the re-INVITE with a=sendrecv -->
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=sipp 87308505 2 IN IP[local_ip_type] [local_ip]
s=-
t=0 0
m=audio [media_port] RTP/AVP 8 101
c=IN IP[media_ip_type] [media_ip]
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
]]>
</send>
<recv request="ACK">
</recv>
<recv request="BYE">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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="2000"/>
</scenario>