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.    <!-- uncomment to test with 100ms pause bween reinvites -->
150.    <!-- <pause milliseconds="100"/> -->
151.  
152.    <send retrans="500">
153.     <![CDATA[
154.  
155.      INVITE sip:[$ct]@[remote_ip]:[remote_port] SIP/2.0
156.      Via: SIP/2.0/[transport] [local_ip]:[local_port]
157.      From: [service] <sip:[service]@[local_ip]:[local_port]>;tag=[pid]SIPpTag01[call_number]
158.      To[$remote_from]
159.      [last_Call-ID:]
160.      CSeq: 31994 INVITE
161.      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
162.      Content-Type: application/sdp
163.      Content-Length: [len]
164.  
165.      v=0
166.      o=- 833252837 32405 IN IP[local_ip_type] [local_ip]
167.      s=-
168.      c=IN IP[media_ip_type] [media_ip]
169.      t=0 0
170.      m=audio [media_port] RTP/AVP 8 101
171.      a=rtpmap:8 PCMA/8000
172.      a=rtpmap:101 telephone-event/8000
173.      a=fmtp:101 0-15
174.      a=maxptime:20
175.      a=sendrecv
176.  
177.    ]]>
178.    </send>
179.  
180.    <recv response="100" optional="true">
181.    </recv>
182.  
183.    <recv response="200">
184.    </recv>
185.    
186.    <send>
187.     <![CDATA[
188.     ACK sip:[$ct]@[remote_ip]:[remote_port] SIP/2.0
189.     Via: SIP/2.0/[transport] [local_ip]:[local_port]
190.     From: [service] <sip:[service]@[local_ip]:[local_port]>;tag=[pid]SIPpTag01[call_number]
191.     To[$remote_from]
192.     [last_Call-ID:]
193.     CSeq: 31994 ACK
194.     Contact: <sip:[local_ip]:[local_port];transport=[transport]>
195.     Max-Forwards: 70
196.    ]]>
197.    </send>
198.  
199.   <recv request="BYE">
200.   </recv>
201.  
202.   <send>
203.     <![CDATA[
204.  
205.      SIP/2.0 200 OK
206.      [last_Via:]
207.      [last_From:]
208.      [last_To:]
209.      [last_Call-ID:]
210.      [last_CSeq:]
211.      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
212.      Content-Length: 0
213.  
214.    ]]>
215.   </send>
216.  
217.   <!-- Keep the call open for a while in case the 200 is lost to be     -->
218.   <!-- able to retransmit it if we receive the BYE again.               -->
219.   <timewait milliseconds="4000"/>
220.  
221.  
222.   <!-- definition of the response time repartition table (unit is ms)   -->
223.   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
224.  
225.   <!-- definition of the call length repartition table (unit is ms)     -->
226.   <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
227.  
228. </scenario>
229.