大道至简

学必求其心得，业必贵于专精。

0%

Linux下SIPp测试Freeswitch

发表于 更新于 分类于

知识必备：Linux操作系统、Sip协议

安装

依赖包安装：

1
2
3
4
5
6
yum -y install gcc-c++
yum -y install ncurses-devel
yum -y install openssl-devel
# 为了安装libpcap，还需要安装以下两个开发包：
yum -y install flex
yum -y install bison

安装libcap

下载libcap ：http://www.tcpdump.org/release/

1
2
3
4
5
6
wget http://www.tcpdump.org/release/libpcap-1.9.1.tar.gz
tar zxvf libcapXXX.gz

login as root , go to folder libpcapXXX and install it. run:
./configure     
make && make install

安装libnet （可跳过）

下载libnet ：

1
2
3
4
5
6

tar zxvf libnetXXX.gz
# 切换root , 去libpnetXXX目录: 
./configure
. /make 
./make install

安装sipp

1
2
cd /home/dev
wget https://github.com/SIPp/sipp/releases/download/v3.6.0/sipp-3.6.0.tar.gz

安装方法：
支持PCAP 声音文件播放，而且支持密码验证支持：(支持407 auth验证支持)

1
2
3
4
5
6
# tar -zxvf sipp-xxx.tar
# cd sipp-xxx
#编译配置：
./configure --with-pcap --with-openssl
#编译&&安装：
make && make install
1
2
#确认是否安装成功：
sipp -v

第一行出现： SIPp v3.6.0-TLS-PCAP-RTPSTREAM 则安装成功

修改系统openfile限制

1
2
3
4
5
6
7
8
9
10
11
# 打开/etc/security/limits.conf 文件并添加如下内容:
* soft nofile 32768
* hard nofile 65535

# 打开/etc/pam.d/login 文件并添加如下内容：
session required /lib/security/pam_limits.so

#永久更改文件描述符所能支持的最大值：
ulimit -s unlimited
ulimit -a

freeswith配置修改

修改max-sessions和sessions-per-second

1
2
3
4
5
cd /etc/freeswitch/autoload_configs
vim switch.conf.xml
# 修改
<param name="max-sessions" value="100000"/>
<param name="sessions-per-second" value="10000"/>

修改拨号的正则

1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
1、cd /etc/freeswitch/dialplan
2、vim public.xml
改为：
<extension name="Balance_load">
        <condition field="destination_number" expression="^([2-6][0-9][0-9][0-9])$">
                <action application="export" data="dialed_extension=$1"/>
                <action application="set" data="sip_h_History-Info=${sip_history_info}"/>
                <action application="set" data="hangup_after_bridge=true"/>
                <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
        </condition>
      </extension>

3、vim default.xml
改为：
    <extension name="public_extensions">
      <condition field="destination_number" expression="^([2-6][0-9][0-9][0-9])$">
        <action application="transfer" data="$1 XML default"/>
      </condition>
    </extension>

添加defalut配置文件

1
2
3
cd /etc/freeswitch/directory/default
# 3000 5999为自己需要的用户
for i in `seq 2000 5999`; do sed -e "s/1000/$i/g" 1000.xml > $i.xml ; done

添加白名单无需鉴权

1
2
3
4
5
6
7
8
9
10
11
12
cd /etc/freeswitch/autoload_configs
vim acl.conf.xml

# 进入编辑模式修改
    <list name="domains" default="deny">
      <!-- domain= is special it scans the domain from the directory to build the ACL -->
      <node type="allow" domain="$${domain}"/>
      <!-- use cidr= if you wish to allow ip ranges to this domains acl. -->
      <node type="allow" cidr="192.168.200.0/24"/>
      <!--新增此行. -->
      <node type="allow" cidr="10.10.81.0/24"/>    
    </list>

SIPp常用参数

注册：

1
sipp -sf regclient_set_c_port.xml 10.3.18.174:5060 -i 10.3.18.172 -p 26000  -inf uac2000_aibus_4000.csv -r 500  -rp 1000 -l 500 -m 4000

呼叫：

1
sipp -sf caller_with_auth_aibus.xml 10.3.18.174:5060 -i 10.3.18.172 -p 26000 -inf uac2000_aibus_4000.csv -r 80  -rp 1000 -l 80 -m 1000 -d 30000 -oocsn ooc_default  -aa  -trace_msg -trace_err  -trace_screen

参数解释

1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
-i: 本地ip，为'Contact:','Via:', 和 'From:' 头部信息设置本地IP，在脚本中用[local_ip]引入
-p:设置本地端口号，默认是由系统随机选择空闲的端口号；
-sf:引入脚本文件，根据需要模拟的呼叫流程编写
-inf:在通话场景中使用外部csv文件引入数据；文件的第一行说明了后面数据的读入方式，常用的有：顺序 	    (SEQUENTIAL), 随机 (RANDOM), 或用户(USER)顺序；第一行对应一个通话，它们由一个或多个’;’分隔数据字段，这些字段可以在xml场景文件中使用[field0], [field1], ...来调用；多个csv文件，可以同时使用（语法：-inf f1.csv -inf f2.csv ...）
xx:xx:xx:xx:5060:Freeswitch服务端IP及freeswitch使用的端口；
-r：并发数量
-rp:并发的时间，单位ms,例如：-r 800 -rp 1000，就是每秒800并发
-l:设置同时呼叫的最大数目；一旦达到此值，流量将被限制直到打的通话数下降；默认值3*call_duration(s)*rate
-m:通话总数，当设置的通话数完成时，停止测试并退出；
-d：自定义的通话时长，单位ms
-aa:针对INFO, UPDATE 和 NOTIFY消息，进行200 OK自动回复应答；
-oocsn：Load out-of-call scenario
-trace_msg:在<场景文件名>_<pid>_messages.log中显示发送和接收的SIP消息；调试时可增加，正试性能测试时，可取消，以免日志量太大影响本地性能；
-trace_screen:在退出SIPp时，把屏蔽上的统计信息写入<场景名>_<pid>_screens.log文件中；在后台模式（-bq选项）时，这对于得到最终状态报告很有用；
-trace_err:跟踪所有非期望的消息到<场景文件名>_<pid>_errors.log；

注册场景测试

测试前：准备uac.csv、uas.csv 和 regclient_set_c_port.xml

uac.csv

1
2
3
4
5
6
7
SEQUENTIAL
2000;2050;[authentication username=2000 password=1234]
2001;2051;[authentication username=2001 password=1234]
2002;2052;[authentication username=2002 password=1234]
2003;2053;[authentication username=2003 password=1234]
2004;2054;[authentication username=2004 password=1234]
...

注意：
1、csv格式创建后用Notepad++打开，否则改编编码导致失败
2、第一行代表执行顺序：包括SEQUENTIAL和RANDOM
3、[filed0]代表第1列，[filed1]代表第2列

uas.csv

1
2
3
4
5
6
7
SEQUENTIAL
2050;;[authentication username=2050 password=1234]
2051;;[authentication username=2051 password=1234]
2052;;[authentication username=2052 password=1234]
2053;;[authentication username=2053 password=1234]
2054;;[authentication username=2054 password=1234]
...

regclient_set_c_port.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
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="regclient">
    <Global variables="c_port" />

    <nop hide="true">
        <action>
            <assignstr assign_to="EXP" value="3600" />
        </action>
    </nop>

  <send>
    <![CDATA[
      REGISTER sip:[remote_ip] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[field0]@[remote_ip]>;tag=acknnkkg.[call_number]
      To: <sip:[field0]@[remote_ip]>
      Call-ID: [call_id]
      CSeq: 1 REGISTER
      Contact: <sip:[field0]@[local_ip]:[$c_port]>
      Max-Forwards: 70
      Subject: Reg Performance Test
      user-agent: SIPp client
      Expires: [$EXP]
      Content-Length: 0
          ]]>
  </send>


  <recv response="401" optional="true" auth="true" next="auth" >
  </recv>

  <recv response="403" optional="true" next="END">
  </recv>

  <recv response="404" optional="true" next="END">
  </recv>

  <recv response="200" next="END" timeout="5000">
  </recv>

  <label id="auth" />
  <send retrans="500">
    <![CDATA[
      REGISTER sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: [field0] <sip:[field0]@[remote_ip]:[remote_port]>;tag=[call_number]
      To: [field0] <sip:[field0]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 2 REGISTER
      Contact: sip:[field0]@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Reg Performance Test 
      user-agent: SIPp client
      Expires: [$EXP]
      [field2]
      Content-Length: 0

    ]]>
  </send>

  <recv response="200" next="END" timeout="5000">
  </recv>

  <label id="END"/>
  <nop hide="true">
  </nop>

<!--<Reference variables="microseconds,seconds" />-->

  <!-- Definition of the response time repartition table (unit is ms)   -->
  <ResponseTimeRepartition value="50, 200"/>

  <!-- Definition of the call length repartition table (unit is ms)     -->
  <CallLengthRepartition value="500, 5000"/>

</scenario>

主叫注册

1
sipp -sf regclient_set_c_port.xml 192.168.200.101:5060 -i 192.168.200.101 -p 26045  -inf uac500.csv -r 30  -rp 1000 -l 100 -m 500

被叫注册

1
sipp -sf regclient_set_c_port.xml 192.168.200.101:5060 -i 192.168.200.101 -p 26048 -inf uas500.csv  -r 30  -rp 1000 -l 100 -m 500

电话场景测试

测试前：准备被叫执行脚本callee_with_bye.xml和主叫呼叫脚本caller_with_auth.xml

callee_with_bye.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
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<scenario name="callee_with_bye">
<!--用于模拟局内被叫侧用户的正常业务流程
        媒体类型：PCMU
        呼叫挂机：主叫方（60秒超时后主动发BYE拆话）-->
        
<!--执行命令样例：sipp -sf callee_with_bye.xml -p 5068-->
                
<!--定义全局状态机，如果收到OPTIONS消息，则跳转至options标签处-->
<recv request="OPTIONS" optional="global" next="options">
</recv>
    
<recv request="INVITE">
<!--参数caller_num、callee_num和caller_tag用于主叫未挂机，BYE接收超时主动发BYE的流程-->
    <action>
        <ereg regexp="sip:(.*)@(.*)>;tag=(.*)"
              search_in="hdr"
              header="From: "
              check_it="true"
              assign_to="junk,caller_num,domain,caller_tag" >
        </ereg>    
        <ereg regexp="sip:(.*)@.*>"
              search_in="hdr"
              header="To: "
              check_it="true"
              assign_to="junk,callee_num" >
        </ereg>      
    </action>
</recv>
        
<!--增加间隔20ms，避免偶现系统不发送100响应的问题-->
<pause hide="true" milliseconds="20"/>  
    
<send>
    <![CDATA[
    SIP/2.0 100 Trying
    [last_Via:]
    [last_From:]
    [last_To:]
    [last_Call-ID:]
    [last_CSeq:]
    Contact: <sip:[local_ip]:[local_port];transport=[transport]>
    Content-Length: 0
    ]]>
    </send>

<!--增加间隔20ms，避免偶现系统不发送180响应的问题-->
<pause hide="true" milliseconds="20"/> 

<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>

<!--设置发送200后等待ACK的重传周期为1秒，如果1秒内收不到ACK则进行200的重传-->
<send retrans="1000" start_rtd="ack">
    <![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=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
    s=-
    c=IN IP[media_ip_type] [media_ip]
    t=0 0
    m=audio [media_port] RTP/AVP 0 8
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    ]]>
</send>
    
<!--设置等待ACK的超时定时器为30秒，如果30秒内收不到ACK则呼叫超时失败而结束-->    
<recv request="ACK" rtd="ack" timeout="30000" />

<!--使用rtp_stream循环播放PCMA音频
<nop hide="true">
    <action>
      <exec rtp_stream="pcap/g711a.pcap,-1,0"/>
    </action>
</nop>
-->
<!--使用rtp_stream循环播放PCMU音频
<nop hide="true">
    <action>
      <exec rtp_stream="pcap/g711u.pcap,-1,0"/>
    </action>
</nop>
-->

<!--使用play_pcap单次播放PCMA音频-->
<nop hide="true">
    <action>
        <exec play_pcap_audio="pcap/g711a.pcap"/> 
    </action>
</nop>
<!--使用play_pcap单次播放PCMU音频
<nop hide="true">
    <action>
        <exec play_pcap_audio="pcap/g711u.pcap"/> 
    </action>
</nop>
-->

<recv request="BYE" timeout="60000" ontimeout="send_bye"/>    
<send next="END">
    <![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>

<label id="options"/>
  <send next="END" >
    <![CDATA[
      SIP/2.0 200 OK
      [last_Via:]
      [last_Call-ID:]
      [last_From:]
      [last_To:];tag=telpo-options[call_number]
      [last_CSeq:]
      [last_Contact:]
      user-agent: SIPP version [sipp_version]
      subject: reg performance
      link-status: I am alive
      Content-Length: 0

    ]]>
</send> 
    
<!--主叫未挂机，BYE接收超时,被叫主动发BYE-->    
<label id="send_bye"/> 
<send start_rtd="bye">
    <![CDATA[
    BYE sip:[$caller_num]@[local_ip]:[local_port] SIP/2.0
    Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
    From: "[$caller_num]" <sip:[$caller_num]@[local_ip]>;tag=[call_number]
    To: "[$callee_num]"<sip:[$callee_num]@[local_ip]>;tag=[$caller_tag]
    Call-ID: [call_id]
    CSeq: 2 BYE
    Max-Forwards: 70
    Subject: normal call scenario 
    Content-Length: 0
    ]]>
</send>

<recv response="200" rtd="bye">
</recv> 

<label id="END"/>

<Reference variables="junk,domain" />

<!-- definition of the response time repartition table (unit is ms)-->
<ResponseTimeRepartition value="50, 200"/>

<!-- definition of the call length repartition table (unit is ms)-->
<CallLengthRepartition value="500, 1000, 10000"/>

</scenario>

caller_with_auth.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
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<scenario name="caller_with_auth">
<!--执行命令样例：sipp -sf caller_with_auth.xml xx.x.x.xx:5060 -p 5066 -inf caller.csv -m 1 -d 10000 -oocsn ooc_default-->
<!--发送INVITE消息，设定重传定时器为1000ms，同时启动定时器invite-->
<send retrans="1000" start_rtd="invite">
    <![CDATA[
      INVITE sip:[field1]@[remote_ip] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: "[field0]" <sip:[field0]@[remote_ip]>;tag=[call_number]zhg8
      To: "[field1]"<sip:[field1]@[remote_ip]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Contact: <sip:[field0]@[local_ip]:[local_port]>
      User-Agent: SIPp client mode version [sipp_version]
      Allow: INVITE,PRACK,ACK,UPDATE,CANCEL,BYE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY
      Max-Forwards: 70
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=SIPp [pid][call_number] 8[pid][call_number]8 IN IP[local_ip_type] [local_ip]
      s=SIPp Normal Call Test
      t=0 0
      m=audio [media_port] RTP/AVP 0 8
      c=IN IP[media_ip_type] [media_ip]
      a=rtpmap:0 PCMU/8000
      a=rtpmap:8 PCMA/8000
      a=ptime:20
      a=sendrecv
    ]]>
     </send>

<recv response="100" optional="true">
</recv>

<!-- <recv response="401" auth="true"> -->
<!-- </recv> -->

<!-- 部分呼叫鉴权可能为407 -->
<!-- <recv response="407" option="true" auth="true">
</recv>

<send>
    <![CDATA[
      ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
      From: "[field0]" <sip:[field0]@[remote_ip]>;tag=[call_number]zhg8
      To: "[field1]"<sip:[field1]@[remote_ip]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Contact: <sip:[field0]@[local_ip]:[local_port]>
      Max-Forwards: 70
      Subject: normal call scenario 
      user-agent: SIPp client mode version [sipp_version]
      Content-Length: 0
    ]]>
  </send>

<send retrans="1000" start_rtd="invite">
    <![CDATA[
        INVITE sip:[field1]@[remote_ip] SIP/2.0
        [last_Via:]
        From: "[field0]" <sip:[field0]@[remote_ip]>;tag=[call_number]zhg8
        To: "[field1]"<sip:[field1]@[remote_ip]>
        Call-ID: [call_id]
        CSeq: 2 INVITE
        [field2]
        Contact: <sip:[field0]@[local_ip]:[local_port]>
        User-Agent: SIPp client mode version [sipp_version]
        Allow: INVITE,PRACK,ACK,UPDATE,CANCEL,BYE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY
        Max-Forwards: 70
        Content-Type: application/sdp
        Content-Length: [len]

        v=0
        o=SIPp [pid][call_number] 8[pid][call_number]8 IN IP[local_ip_type] [local_ip]
        s=SIPp Normal Call Test
        t=0 0
        m=audio [media_port] RTP/AVP 0 8
        c=IN IP[media_ip_type] [media_ip]
        a=rtpmap:0 PCMU/8000
        a=rtpmap:8 PCMA/8000
        a=ptime:20
        a=sendrecv
    
    ]]>
</send>


<!--1xx响应均为可选接收消息，且接收到临时响应后，即可停止invite定时器的计时-->
<!--收到4xx/5xx错误响应后，直接进入呼叫失败-->
<!-- <recv response="100" optional="true" rtd="invite">
</recv>

<recv response="183" optional="true" rtd="invite" next="normal">
</recv>

<recv response="403" optional="true" rtd="invite" next="err_ack">
</recv>

<recv response="407" optional="true" rtd="invite" next="err_ack">
</recv>

<recv response="415" optional="true" rtd="invite" next="err_ack">
</recv>

<recv response="480" optional="true" rtd="invite" next="err_ack">
</recv>

<recv response="486" optional="true" rtd="invite" next="err_ack">
</recv>

<recv response="500" optional="true" rtd="invite" next="err_ack">
</recv>

<recv response="503" optional="true" rtd="invite" next="err_ack">
</recv> -->
 -->
<recv response="180"  optional="true" rtd="invite" next="normal">
</recv>

<label id="normal"/>
<recv response="200" rtd="invite">
    <action>
        <ereg regexp="m=audio ([0-9]*)"
            search_in="msg"
            check_it="true"
            assign_to="junk,callee_media_port" />
    </action>
</recv>

<nop hide="true">
    
</nop>

<send>
    <![CDATA[
        ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
        Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
        From: "[field0]" <sip:[field0]@[remote_ip]>;tag=[call_number]zhg8
        To: "[field1]"<sip:[field1]@[remote_ip]>[peer_tag_param]
        Call-ID: [call_id]
        CSeq: 2 ACK
        Contact: <sip:[field0]@[local_ip]:[local_port]>
        Max-Forwards: 70
        Subject: normal call scenario
        user-agent: SIPp client mode version [sipp_version]
        Content-Length: 0
    ]]>
</send>

<!--使用rtp_stream循环播放PCMA音频
<nop hide="true">
    <action>
      <exec rtp_stream="pcap/g711a.pcap,-1,0"/>
    </action>
</nop>
-->
<!--使用rtp_stream循环播放PCMU音频
<nop hide="true">
    <action>
      <exec rtp_stream="pcap/g711u.pcap,-1,0"/>
    </action>
</nop>
-->

<!--使用play_pcap单次播放PCMA音频-->
<nop hide="true">
    <action>
        <exec play_pcap_audio="pcap/g711a.pcap"/> 
    </action>
</nop>
<!--使用play_pcap单次播放PCMU音频
<nop hide="true">
    <action>
        <exec play_pcap_audio="pcap/g711u.pcap"/> 
    </action>
</nop>
-->

<!--媒体流传输完毕后，暂停发送BYE结束呼叫，在执行命令中增加参数-d 指定暂停时间：如-d 10000暂停10秒-->
<pause />

<send start_rtd="bye">
    <![CDATA[
        BYE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
        Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
        From: "[field0]" <sip:[field0]@[remote_ip]>;tag=[call_number]zhg8
        To: "[field1]"<sip:[field1]@[remote_ip]>[peer_tag_param]
        Call-ID: [call_id]
        CSeq: 3 BYE
        Max-Forwards: 70
        Subject: normal call scenario
        Content-Length: 0
    ]]>
</send>

<recv response="200" rtd="bye" next="END">
</recv>

<!--异常结束，复用err_ack流程-->
<label id="err_ack"/>

<send>
    <![CDATA[
        ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
        [last_Via:]
        From: "[field0]" <sip:[field0]@[remote_ip]>;tag=[call_number]zhg8
        To: "[field1]"<sip:[field1]@[remote_ip]>[peer_tag_param]
        [last_Call-ID:]
        CSeq: 2 ACK
        Max-Forwards: 70
        Subject: normal call scenario
        user-agent: SIPp client mode version [sipp_version]
        Content-Length: 0
    ]]>
</send>

<!--正常结束-->
<label id="END"/>
<nop hide="true">
</nop>

<!--如果存在定义了但未被使用的变量，可以在下面语句的双引号中增加，避免运行时报错-->
<Reference variables="junk,callee_media_port" />
    
<!--definition of the response time repartition table (unit is ms)   -->
<ResponseTimeRepartition value="50, 200,1000,2000,4000,10000"/>

<!--definition of the call length repartition table (unit is ms)     -->
<CallLengthRepartition value="500, 1000, 10000"/>

</scenario>

测试呼叫的步骤：

执行先后顺序

1
2
3
4
5
6
7
8
9
10
11
# 1、启动主叫注册
sipp -sf regclient_set_c_port.xml 192.168.200.101:5060 -i 192.168.200.101 -p 26045  -inf uac500.csv -r 30  -rp 1000 -l 100 -m 500

# 2、启动被叫注册
sipp -sf regclient_set_c_port.xml 192.168.200.101:5060 -i 192.168.200.101 -p 26048 -inf uas500.csv  -r 30  -rp 1000 -l 100 -m 500

#3、等待被叫注册结束，启动被叫
sipp -sf callee_with_bye.xml -i 192.168.200.101 -p 26048 -trace_err 

# 4、等待被叫执行后，执行主叫
sipp -sf caller_with_auth.xml 192.168.200.101:5060 -i 192.168.200.101 -p 26045 -inf uac500.csv -r 500  -rp 1000 -l 5000 -m 5000 -d 60000 -oocsn ooc_default -trace_err

性能分析

并发数（CC） = 平均通话时长 * CPS