1 <?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 <!-- Sipp default 'branchc' scenario. -->
20 <!-- -->
21 <scenario name="basiccall">
22 <send retrans="500">
23 <![CDATA[
24
25 INVITE sip:[field3]@[field1] SIP/2.0
26 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
27 From: <sip:[field0]@[field1]>;tag=[call_number]
28 To: <sip:[field3]@[field1]>
29 Call-ID: [call_id]
30 CSeq: 1 INVITE
31 Contact: sip:[field0]@[local_ip]:[local_port]
32 Max-Forwards: 7
33 Subject: Performance Test
34 User-Agent: Sipp
35 X-cid: [call_id]
36 Content-Type: application/sdp
37 Content-Length: [len]
38
39 v=0
40 o=[local_ip] 53655765 2353687637 IN IP[local_ip_type] [local_ip]
41 s=-
42 c=IN IP[local_ip_type] [local_ip]
43 t=0 0
44 m=audio [rtpstream_audio_port] RTP/AVP 8 0 101
45 a=rtpmap:0 PCMU/8000
46 a=rtpmap:101 telephone-event/8000
47 a=fmtp:101 0-16
48 a=sendrecv
49 ]]>
50 </send>
51
52 <recv response="100" optional="true">
53 </recv>
54
55 <recv response="407" auth="true">
56 <action>
57 <ereg regexp="branch=(.*)"
58 search_in="hdr"
59 header="Via: "
60 assign_to="8"/>
61 </action>
62 </recv>
63
64 <!-- By adding rrs="true" (Record Route Sets), the route sets -->
65 <!-- are saved and used for following messages sent. Useful to test-->
66 <!-- against stateful SIP proxies/B2BUAs. -->
67 <!-- Packet lost can be simulated in any send/recv message by -->
68 <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.-->
69 <send>
70 <![CDATA[
71
72 ACK sip:[field3]@[field1]:[remote_port] SIP/2.0
73 Via: SIP/2.0/[transport] [local_ip]:[local_port];[$8]
74 From: <sip:[field0]@[field1]>;tag=[call_number]
75 To: <sip:[field3]@[field1]>[peer_tag_param]
76 Call-ID: [call_id]
77 CSeq: 1 ACK
78 Contact: sip:[field0]@[local_ip]:[local_port]
79 Max-Forwards: 7
80 User-Agent: Sipp
81 X-cid: [call_id]
82 Subject: Performance Test
83 Content-Length: 0
84 ]]>
85 </send>
86
87
88 <send retrans="500">
89 <![CDATA[
90
91 INVITE sip:[field3]@[field1]:[remote_port] SIP/2.0
92 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
93 From: <sip:[field0]@[field1]>;tag=[call_number]
94 To: <sip:[field3]@[field1]>
95 Call-ID: [call_id]
96 CSeq: 2 INVITE
97 Contact: <sip:[field0]@[local_ip]:[local_port]>; expires=600
98 [field2]
99 Max-Forwards: 7
100 Subject: Performance Test
101 User-Agent: Sipp
102 X-cid: [call_id]
103 Content-Type: application/sdp
104 Content-Length: [len]
105 Expires: 600
106
107 v=0
108 o=[local_ip] 53655765 2353687637 IN IP[local_ip_type] [local_ip]
109 s=-
110 c=IN IP[local_ip_type] [local_ip]
111 t=0 0
112 m=audio [rtpstream_audio_port] RTP/AVP 8 0 101
113 a=rtpmap:0 PCMU/8000
114 a=rtpmap:101 telephone-event/8000
115 a=fmtp:101 0-16
116 a=sendrecv
117 ]]>
118 </send>
119
120
121 <recv response="100" optional="true">
122 </recv>
123
124 <recv response="183" optional="true">
125 </recv>
126
127 <recv response="200">
128 </recv>
129
130 <!-- Packet lost can be simulated in any send/recv message by -->
131 <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
132 <send>
133 <![CDATA[
134
135 ACK sip:[field3]@[field1]:[remote_port] SIP/2.0
136 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
137 From: <sip:[field0]@[field1]>;tag=[call_number]
138 To: <sip:[field3]@[field1]>[peer_tag_param]
139 Route: <sip:10.100.125.17:5060;lr>
140 Call-ID: [call_id]
141 CSeq: 2 ACK
142 Contact: sip:[field0]@[local_ip]:[local_port]
143 Max-Forwards: 7
144 Subject: Performance Test
145 User-Agent: Sipp
146 X-cid: [call_id]
147 Content-Length: 0
148 ]]>
149 </send>
150
151 <!--
152 <nop>
153 <action>
154 <exec play_pcap_audio="g711u.pcap"/>
155 </action>
156 </nop>
157 -->
158
159 <nop>
160 <action>
161 <exec rtp_stream="hahaha.wav,-1,0" />
162 </action>
163 </nop>
164
165
166
167 <!-- <pause distribution="normal" mean="80000" stdev="1000"/> -->
168 <pause milliseconds="56000"/>
169
170
171 <!-- The 'crlf' option inserts a blank line in the statistics report.-->
172 <send retrans="2000">
173 <![CDATA[
174
175 BYE sip:[field3]@[field1]:[remote_port]; SIP/2.0
176 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
177 From: <sip:[field0]@[field1]>;tag=[call_number]
178 To: <sip:[field3]@[field1]>[peer_tag_param]
179 Route: <sip:10.100.125.17:5060;lr>
180 Call-ID: [call_id]
181 CSeq: 2 BYE
182 Contact: sip:[field0]@[local_ip]:[local_port]
183 Max-Forwards: 7
184 Subject: Performance Test
185 User-Agent: Sipp
186 X-cid: [call_id]
187 Content-Length: 0
188 ]]>
189 </send>
190
191 <recv response="200" crlf="true">
192 </recv>
193
194 <ResponseTimeRepartition value="1000, 2000, 3000, 5000, 10000, 32000"/>
195 <CallLengthRepartition value="1000, 5000, 10000, 15000, 20000, 25000, 30000"/>
196 </scenario>
197
使用pcap 模式发送RTP 包，效率并不高，会有大量的软中断，不能支持大并发测试，
因为使用pcap 发送RTP的模式中， SIPP client 协商出的RTP 端口号是一个（不管有多少路），也即auto_media_port
没有为每一路分配单独的RTP 端口。
而且为了操作方便，每一路通话，SIPP 都创建了一个RAW socket， 这些socket 都绑定在一个端口上，
在高并发中，例如2000路并发，就有2000个raw socket 绑定在一个端口收发，这个软中断非常高。
为什么软中断比较高，可以参考raw socket 接收数据流程理解。
所以使用rtp_stream 方式会好很多。使用这种方式， 每一路都有自己的单独的RTP 端口。还有一个好处是
如果使用pcap的话， 这个wireshark 包大小需要自己控制， sipp并不能循环播放这个pcap包，但是使用rtp_stream
的话，就可以测试长时间通话了。
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