sipp模拟freepbx分机测试(SIP协议调试)
2024-10-21 02:43:17
1、sipp的安装
1) 在centos 7.2下安装
yum install make gcc gcc-c++ ncurses ncurses.x86_64 ncurses-devel ncurses-devel.x86_64 openssl lksctp-tools libnet libpcap libpcap-devel libpcap.x86_64 libpcap-devel.x86_64 gsl gsl-devel
cd /root/
wget http://sourceforge.net/projects/sipp/files/sipp/3.4/sipp-3.3.990.tar.gz/download
tar -zxvf sipp-3.3.990.tar.gz
cd sipp-3.3.990/
./configure --with-sctp --with-pcap --with-openssl make && sudo make install
sipp -v
2) 在ubuntu14.04 下有些差异
http://sipp.sourceforge.net/doc3.3/reference.html
http://sipp.sourceforge.net/doc/reference.html
tar xvf sipp-3.5.2.tar.gz
cd sipp-3.5.2
sudo apt-get install libsctp-dev lksctp-tools
sudo apt-get install libncurses5-dev
sudo apt-get install libpcap-dev libssl-dev
sudo apt-get install build-essential
./build.sh --with-pcap --with-sctp --with-openssl
make
make install
2、sipp拨号测试
先在freepbx上创建好两个分机103和104
最好是在linux下创建xml脚本文件，因为windows操作之后文档的编码格式可能会改变，xml对编码格式非常敏感。在windows和linux两边编辑容易出问题。
1) 主叫账户
vi caller.csv
SEQUENTIAL
103;104;[authentication username=103 password=103]
2) 被叫账号
vi callee.csv
SEQUENTIAL
104;;[authentication username=104 password=104]
3) 注册脚本
vi regclient_set_c_port.xml
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="regclient">
<!--本脚本特为持续性测试使用，如单次使用，建议-p 与-set c_port的端口设为相同-->
<!--执行命令样例：sipp -sf regc_set_c_port.xml 172.31.231.220:5060 -i 172.31.231.23 -p 5077 -inf caller.csv -m 1 -set c_port 5066-->
<Global variables="c_port" /> <nop hide="true">
<action>
<!--设置EXP的值为3600-->
<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]:[$c_port]>
Max-Forwards: 70
Subject: Reg Performance Test made by wangwei
user-agent: SIPp client
Expires: [$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>
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[field0]@[local_ip]:[$c_port]>
Max-Forwards: 70
Subject: Reg Performance Test made by wangwei
user-agent: SIPp client
Expires: [$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>
4) 被叫脚本
vi callee_with_bye.xml
<?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 regexp="<sip:(.*)@.*>"
search_in="hdr"
header="To: "
check_it="true"
assign_to="junk,callee_num" />
</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 test made by wangwei
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 by wangwei
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>
5) 主叫脚本
vi caller_with_auth.xml
<?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 47.106.93.236:5060 -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="401" 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 by wangwei
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
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: 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=sendrecv
a=ptime:20
]]>
</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="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 by wangwei
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 by wangwei
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 by wangwei
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>
6) 按以下步骤运行
#主叫注册
sipp -sf regclient_set_c_port.xml 47.106.xx.xxx:5060 -i 192.168.247.152 -p 5066 -inf caller.csv -set c_port 5066 -m 1
#被叫注册
sipp -sf regclient_set_c_port.xml 47.106.xx.xxx:5060 -i 192.168.247.152 -p 5088 -inf callee.csv -set c_port 5088 -m 1
#被叫
sipp -sf callee_with_bye.xml -i 192.168.247.152 -p 5088 -trace_err
#主叫
sudo sipp -sf caller_with_auth.xml 47.106.xx.xxx:5060 -i 192.168.247.152 -p 5066 -inf caller.csv -m 1 -d 10000 -oocsn ooc_default -trace_err
7) 登陆会议室
我在freepbx上创建的会议室号是2018，没有密码，直接拨号就能登进去
所以将主叫账户caller.csv改成登陆会议室就行
SEQUENTIAL
103;2018;[authentication username=103 password=103]
3、其他拓展应用
具体应用看手册 SIPp3.4中文参考手册.pdf https://files.cnblogs.com/files/dong1/SIPp3.4%E4%B8%AD%E6%96%87%E5%8F%82%E8%80%83%E6%89%8B%E5%86%8C.pdf
1) SIPP通过next指定id实现循环
https://blog.csdn.net/voip3261/article/details/10335925
参考 https://blog.csdn.net/netluoriver/article/details/21786301
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