sipp模拟freepbx分机测试(SIP协议调试)

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

https://github.com/SIPp

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