1. <?xml version="1.0" encoding="ISO-8859-1" ?>
2. <!DOCTYPE scenario SYSTEM "sipp.dtd">
3.  
4. <!-- UAS with multiple reinvite      -->
5.  
6. <scenario name="Basic UAS responder">
7.  
8.   <recv request="INVITE" crlf="true">
9.     <action>
10.       <ereg regexp=".*" search_in="hdr" header="From" assign_to="remote_from"/>
11.       <ereg regexp=".*sip:([^;@]*)" search_in="hdr" header="Contact:" assign_to="duh,ct"/>
12.      </action>
13.   </recv>
14.   <Reference variables="duh"/>
15.  
16.   <send>
17.     <![CDATA[
18.  
19.      SIP/2.0 100 trying -- your call is important to us
20.      [last_Via:]
21.      [last_From:]
22.      [last_To:];tag=[pid]SIPpTag01[call_number]
23.      [last_Call-ID:]
24.      [last_CSeq:]
25.      Content-Length: 0
26.  
27.    ]]>
28.   </send>
29.  
30.   <pause milliseconds="1590"/>
31.  
32.   <send>
33.     <![CDATA[
34.  
35.      SIP/2.0 183 Session Progress
36.      [last_Via:]
37.      [last_From:]
38.      [last_To:];tag=[pid]SIPpTag01[call_number]
39.      [last_Call-ID:]
40.      [last_CSeq:]
41.      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
42.      Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
43.      Content-Disposition: session; handling=required
44.      Content-Type: application/sdp
45.      Content-Length: [len]
46.  
47.      v=0
48.      o=- 833252837 32403 IN IP[local_ip_type] [local_ip]
49.      s=-
50.      c=IN IP[media_ip_type] [media_ip]
51.      t=0 0
52.      m=audio [media_port] RTP/AVP 8 101
53.      a=rtpmap:8 PCMA/8000
54.      a=rtpmap:101 telephone-event/8000
55.      a=fmtp:101 0-15
56.      a=maxptime:20
57.      a=sendrecv
58.  
59.    ]]>
60.   </send>
61.  
62.   <pause milliseconds="10000"/>
63.  
64.   <send retrans="500">
65.     <![CDATA[
66.  
67.      SIP/2.0 200 OK
68.      [last_Via:]
69.      [last_From:]
70.      [last_To:];tag=[pid]SIPpTag01[call_number]
71.      [last_Call-ID:]
72.      [last_CSeq:]
73.      Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay,  multipart/mixed
74.      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
75.      Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
76.      Supported: replaces
77.      Content-Disposition: session; handling=required
78.      Content-Type: application/sdp
79.      Content-Length: [len]
80.  
81.      v=0
82.      o=- 833252837 32403 IN IP[local_ip_type] [local_ip]
83.      s=-
84.      c=IN IP[media_ip_type] [media_ip]
85.      t=0 0
86.      m=audio [media_port] RTP/AVP 8 101
87.      a=rtpmap:8 PCMA/8000
88.      a=rtpmap:101 telephone-event/8000
89.      a=fmtp:101 0-15
90.      a=maxptime:20
91.      a=sendrecv
92.  
93.    ]]>
94.   </send>
95.  
96.   <recv request="ACK"
97.        optional="false"
98.        rtd="true"
99.        crlf="true">
100.   </recv>
101.  
102.   <send retrans="500">
103.     <![CDATA[
104.  
105.      INVITE sip:[$ct]@[remote_ip]:[remote_port] SIP/2.0
106.      Via: SIP/2.0/[transport] [local_ip]:[local_port]
107.      From: [service] <sip:[service]@[local_ip]:[local_port]>;tag=[pid]SIPpTag01[call_number]
108.      To[$remote_from]
109.      [last_Call-ID:]
110.      CSeq: 31993 INVITE
111.      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
112.      Content-Type: application/sdp
113.      Content-Length: [len]
114.  
115.      v=0
116.      o=- 833252837 32404 IN IP[local_ip_type] [local_ip]
117.      s=-
118.      c=IN IP[media_ip_type] [media_ip]
119.      t=0 0
120.      m=audio [media_port] RTP/AVP 0 101
121.      a=rtpmap:0 PCMU/8000
122.      a=rtpmap:101 telephone-event/8000
123.      a=fmtp:101 0-15
124.      a=maxptime:20
125.      a=sendrecv
126.  
127.    ]]>
128.    </send>
129.  
130.    <recv response="100" optional="true">
131.    </recv>
132.  
133.    <recv response="200">
134.    </recv>
135.    
136.    <send>
137.     <![CDATA[
138.     ACK sip:[$ct]@[remote_ip]:[remote_port] SIP/2.0
139.     Via: SIP/2.0/[transport] [local_ip]:[local_port]
140.     From: [service] <sip:[service]@[local_ip]:[local_port]>;tag=[pid]SIPpTag01[call_number]
141.     To[$remote_from]
142.     [last_Call-ID:]
143.     CSeq: 31993 ACK
144.     Contact: <sip:[local_ip]:[local_port];transport=[transport]>
145.     Max-Forwards: 70
146.    ]]>
147.    </send>
148.  
149.    <pause milliseconds="100"/>
150.  
151.    <send retrans="500">
152.     <![CDATA[
153.  
154.      INVITE sip:[$ct]@[remote_ip]:[remote_port] SIP/2.0
155.      Via: SIP/2.0/[transport] [local_ip]:[local_port]
156.      From: [service] <sip:[service]@[local_ip]:[local_port]>;tag=[pid]SIPpTag01[call_number]
157.      To[$remote_from]
158.      [last_Call-ID:]
159.      CSeq: 31994 INVITE
160.      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
161.      Content-Type: application/sdp
162.      Content-Length: [len]
163.  
164.      v=0
165.      o=- 833252837 32405 IN IP[local_ip_type] [local_ip]
166.      s=-
167.      c=IN IP[media_ip_type] [media_ip]
168.      t=0 0
169.      m=audio [media_port] RTP/AVP 8 101
170.      a=rtpmap:8 PCMA/8000
171.      a=rtpmap:101 telephone-event/8000
172.      a=fmtp:101 0-15
173.      a=maxptime:20
174.      a=sendrecv
175.  
176.    ]]>
177.    </send>
178.  
179.    <recv response="100" optional="true">
180.    </recv>
181.  
182.    <recv response="200">
183.    </recv>
184.    
185.    <send>
186.     <![CDATA[
187.     ACK sip:[$ct]@[remote_ip]:[remote_port] SIP/2.0
188.     Via: SIP/2.0/[transport] [local_ip]:[local_port]
189.     From: [service] <sip:[service]@[local_ip]:[local_port]>;tag=[pid]SIPpTag01[call_number]
190.     To[$remote_from]
191.     [last_Call-ID:]
192.     CSeq: 31994 ACK
193.     Contact: <sip:[local_ip]:[local_port];transport=[transport]>
194.     Max-Forwards: 70
195.    ]]>
196.    </send>
197.  
198.   <recv request="BYE">
199.   </recv>
200.  
201.   <send>
202.     <![CDATA[
203.  
204.      SIP/2.0 200 OK
205.      [last_Via:]
206.      [last_From:]
207.      [last_To:]
208.      [last_Call-ID:]
209.      [last_CSeq:]
210.      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
211.      Content-Length: 0
212.  
213.    ]]>
214.   </send>
215.  
216.   <!-- Keep the call open for a while in case the 200 is lost to be     -->
217.   <!-- able to retransmit it if we receive the BYE again.               -->
218.   <timewait milliseconds="4000"/>
219.  
220.  
221.   <!-- definition of the response time repartition table (unit is ms)   -->
222.   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
223.  
224.   <!-- definition of the call length repartition table (unit is ms)     -->
225.   <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
226.  
227. </scenario>