sipp模拟电信运营商VoIP终端测试(SIP协议调试)

三大运营商和其他众多通信业务厂商都可能有SIP服务器，用来支持语音对讲，多媒体调度等功能，他们的平台可能不是标准的SIP协议会话。

为了应对没完没了的对接各个厂商的平台，这里再整理了一套协议脚本，毕竟全都是没有意义的无用功，标准化的SIP会话就是最好的。

感谢西安的枫林晨曦，帮忙抓包，整理了这套脚本。

 

1、先熟悉一下SIP的各种请求方法

INVITE，ACK，BYE，CANCEL，OPTIONS，REGISTER，PRACK，SUBSCRIBE，NOTIFY，PUBLISH，INFO，REFER，MESSAGE，UPDATE

SIP request methods

https://en.wikipedia.org/wiki/List_of_SIP_request_methods

 

2、调试协议，少不了要抓包分析数据，手机app抓包，最简单，最靠谱的就是在电脑上装个wifi热点，让手机连上这个热点，在电脑上抓取这个wifi网卡的数据。

有的电脑网卡能模拟wifi AP，如果不支持，就买个wifi网卡吧

Android抓包方法(三)之Win7笔记本Wifi热点+WireShark工具

https://www.cnblogs.com/findyou/p/3491065.html

 

3、各请求流程的协议脚本

不一定能直接用，一般都需要调整，因为每家都可能有差异，按照厂商给的协议文档，或者抓包信息来调整。

虽然抓包就什么都有了，但是我这里还是把运营商的信息屏蔽了，毕竟签了保密协议，免得被找茬。

不熟悉协议可以参考https://github.com/saghul/sipp-scenarios

1）regclient_set_c_port.xml

<?xml version="1.0" encoding="utf-8" ?>
<!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 regclient_set_c_port.xml SIP_Proxy_IP:SIP_Proxy_Port -i 172.16.0.6 -p 5088 -inf callee.csv -set c_port 5088 -m 1-->
    <Global variables="c_port" />
    
    <nop hide="true">
        <action>
            <!--设置EXP的值为3600-->
            <assignstr assign_to="EXP" value="3600" />
            <assignstr assign_to="DOMAIN" value="运营商域名" />
        </action>
    </nop>
    
  <send>
    <![CDATA[
      REGISTER sip:[$DOMAIN] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[field0]@[$DOMAIN]>;tag=acknnkkg.[call_number]
      To: <sip:[field0]@[$DOMAIN]>
      Call-ID: [call_id]
      CSeq: 1 REGISTER
      Contact: <sip:[field0]@[local_ip]:[$c_port];line=79169130b56d431>
      Max-Forwards: 70
      Subject: Reg Performance Test made by wangwei
      user-agent: SIPp client
      Digest username="sip:[field0]@[$DOMAIN]", realm="[$DOMAIN]", uri="sip:[$DOMAIN]"
      Expires: [$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:[$DOMAIN] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[field0]@[local_ip]:[$c_port];line=79169130b56d431>
      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>

 

2）publish.xml

<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="publish_client">
<!---->
<!--执行命令样例：sipp -sf publish.xml SIP_Proxy_IP:SIP_Proxy_Port -i 172.16.0.6 -p 5088 -inf callee.csv  -m 1-->
    
    <nop hide="true">
        <action>
            <!--设置EXP的值为3600-->
            <assignstr assign_to="EXP" value="3600" />
            <assignstr assign_to="DOMAIN" value="运营商域名" />
        </action>
    </nop>
    
  <send>
    <![CDATA[
        PUBLISH sip:[field0]@[$DOMAIN] SIP/2.0
        Via: SIP/2.0/UDP [local_ip]:[local_port];rport;branch=[branch]
        From: <sip:[field0]@[$DOMAIN]>;tag=acknnkkg.[call_number]
        To: <sip:[field0]@[$DOMAIN]>
        Call-ID: [call_id]
        CSeq: 2 PUBLISH
        Max-Forwards: 70
        User-Agent: SIPp client
        Expires: [$EXP]
        Event: poc-settings
        Accept-Contact: 请查找运营商文档字段
        Supported: 100rel,eventlist,timer,multiple-refer
        Content-Type: 请查找运营商文档字段
        Content-Length:[len]

        <?xml version="1.0" encoding="UTF-8"?>
        <poc-settings xmlns="请查找运营商文档字段" xsi:schemaLocation="请查找运营商文档字段">
        <entity id="sip:[field0]@[$DOMAIN]">
        <isb-settings>
        <incoming-session-barring active="false" />
        </isb-settings>
        <am-settings>
        <answer-mode>automatic</answer-mode>
        </am-settings>
        <ipab-settings>
        <incoming-personal-alert-barring active="false" />
        </ipab-settings>
        <sss-settings>
        <simultaneous-sessions-support active="true" />
        </sss-settings>
        </entity>
        </poc-settings>
          ]]>
  </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>

 

3）poc.xml

<?xml version="1.0" encoding="utf-8" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<scenario name="caller_with_auth">

<nop hide="true">
    <action>
        <!--设置EXP的值为3600-->
        <assignstr assign_to="POCID" value="C127375" />
        <assignstr assign_to="EXP" value="120" />
        <assignstr assign_to="DOMAIN" value="运营商域名" />
    </action>
</nop>


<!--执行命令样例：sudo sipp -sf poc.xml SIP_Proxy_IP:SIP_Proxy_Port -i 172.16.0.6 -p 5088 -inf callee.csv -m 1 -d 60000 -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>
    <![CDATA[
        INVITE sip:[$POCID]&[field1]@[$DOMAIN];session=chat SIP/2.0
        Via: SIP/2.0/UDP [local_ip]:[local_port];rport;branch=[branch]
        From: <sip:[field0]@[$DOMAIN]>;tag=4140059
        To: <sip:[$POCID]&[field1]@[$DOMAIN];session=chat>
        Call-ID:[call_id]
        CSeq: 1 INVITE
        Contact: <sip:[field0]@[local_ip]:[local_port]>;请查找运营商文档字段
        Allow: INVITE,ACK,CANCEL,BYE,REGISTER,PRACK,PUBLISH,REFER,SUBSCRIBE,NOTIFY,MESSAGE
        P-Preferred-Identity: <sip:[field0]@[$DOMAIN]>
        Session-Expires: [$EXP]
        Supported: replaces, 100rel, timer
        Max-Forwards: 70
        User-Agent: SIPp client mode
        Accept-Contact: 请查找运营商文档字段
        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
        c=IN IP[media_ip_type] [media_ip]
        t=0 0
        m=audio [media_port] RTP/AVP 106
        a=rtpmap:106 AMR/8000
        a=fmtp:106 mode-set=0,1,2,3,4,5,6,7; octet-align=1
        a=ptime:200
        m=application 10667 UDP TBCP
        a=fmtp:TBCP queuing=0; tb_priority=1; poc_sess_priority=0
    ]]>
     </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">
</recv>

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

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

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

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

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

<recv response="503" optional="true" next="err_ack">
</recv>

<recv response="180"  optional="true" next="normal">
</recv>

<label id="normal"/>
<!--<recv response="200">
</recv>-->

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

<send>
    <![CDATA[
        ACK sip:[$POCID]&[field1]@[$DOMAIN];session=chat SIP/2.0
        Via: SIP/2.0/UDP [local_ip]:[local_port];rport;branch=[branch]
        Route: <sip:[remote_ip];lr>
        From: <sip:[field0]@[$DOMAIN]>;tag=4140059
        To: <sip:[$POCID]&[field1]@[$DOMAIN];session=chat>;tag=9500414
        Call-ID: [call_id]
        CSeq: 1 ACK
        Contact: <sip:[field0]@[local_ip]:[local_port]>;请查找运营商文档字段
        Max-Forwards: 70
        User-Agent: SIPp client mode
        Content-Length: 0
    ]]>
</send>

<!--<pause hide="true" milliseconds="500"/> 

<send>
    <![CDATA[
        SUBSCRIBE sip:[$POCID]&[field1]@[$DOMAIN];session=chat SIP/2.0
        Via: SIP/2.0/UDP [local_ip]:[local_port];rport;branch=[branch]
        From: <sip:[field0]@[$DOMAIN]>;tag=4628763
        To: <sip:[$POCID]&[field1]@[$DOMAIN];session=chat>
        Call-ID: [call_id]
        CSeq: 2 SUBSCRIBE
        Contact: <sip:[field0]@[local_ip]:[local_port]>
        Max-Forwards: 70
        User-Agent: SIPp client mode
        Expires: [$EXP]
        Event: conference
        Accept-Contact:请查找运营商文档字段
        Content-Length: 0
    ]]>
</send>

<recv response="200">
</recv>-->

<pause hide="true" milliseconds="500"/> 

<!--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>
-->

<!--使用play_pcap单次播放amr音频-->
<nop hide="true">
    <action>
        <exec play_pcap_audio="pcap/amr.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>
    <![CDATA[
        SUBSCRIBE sip:[$POCID]&[field1]@[$DOMAIN] SIP/2.0
        Via: SIP/2.0/UDP [local_ip]:[local_port];rport;branch=[branch]
        From: <sip:[field0]@[$DOMAIN]>;tag=4628763
        To: <sip:[$POCID]&[field1]@[$DOMAIN];session=chat>[peer_tag_param]
        Call-ID: [call_id]
        CSeq: 3 SUBSCRIBE
        Contact: <sip:[field0]@[local_ip]:[local_port]>
        Max-Forwards: 70
        User-Agent: SIPp client mode
        Accept: 请查找运营商文档字段
        Expires: 0
        Event: conference
        Accept-Contact: 请查找运营商文档字段
        Content-Length: 0
    ]]>
</send>


<recv response="200">
</recv>-->




<send start_rtd="bye">
    <![CDATA[
        BYE sip:[$POCID]&[field1]@[$DOMAIN];session=chat SIP/2.0
        Via: SIP/2.0/UDP [local_ip]:[local_port];rport;branch=[branch]
        Route: <sip:[remote_ip];lr>
        From: <sip:[field0]@[$DOMAIN]>;tag=4140059
        To: <sip:[$POCID]&[field1]@[$DOMAIN];session=chat>;tag=9500414
        Call-ID: [call_id]
        CSeq: 4 BYE
        Contact: <sip:[field0]@[local_ip]:[local_port]>
        Max-Forwards: 70
        User-Agent: SIPp client mode
        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:[$POCID]&[field1]@[$DOMAIN];session=chat SIP/2.0
        [last_Via:]
        From: <sip:[field0]@[$DOMAIN]>;tag=[call_number]zhg8
        To: <sip:[$POCID]&[field1]@[$DOMAIN];session=chat>[peer_tag_param]
        [last_Call-ID:]
        CSeq: 1 ACK
        Contact: <sip:[field0]@[local_ip]:[local_port]>;请查找运营商文档字段
        Max-Forwards: 70
        User-Agent: SIPp client mode
        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>

 

4） subscribe.xml

<?xml version="1.0" encoding="utf-8" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<scenario name="subscribe">
<Global variables="c_port" />

<!--执行命令样例：sipp -sf subscribe.xml SIP_Proxy_IP:SIP_Proxy_Port -i 172.16.0.6 -p 5077 -set c_port 5088 -inf callee.csv -m 1 -d 40000-->

<nop hide="true">
    <action>
        <!--设置EXP的值为3600-->
        <assignstr assign_to="POCID" value="C127375" />
        <assignstr assign_to="EXP" value="120" />
        <assignstr assign_to="DOMAIN" value="运营商域名" />
    </action>
</nop>

<send>
    <![CDATA[
        SUBSCRIBE sip:[$POCID]&[field1]@[$DOMAIN];session=chat SIP/2.0
        Via: SIP/2.0/UDP [local_ip]:[local_port];rport;branch=[branch]
        From: <sip:[field0]@[$DOMAIN]>;tag=4629583
        To: <sip:[$POCID]&[field1]@[$DOMAIN];session=chat>
        Call-ID: [call_id]
        CSeq: 2 SUBSCRIBE
        Contact: <sip:[field0]@[local_ip]:[$c_port]>
        Max-Forwards: 70
        User-Agent: SIPp client mode
        Expires: [$EXP]
        Event: conference
        Accept-Contact: 请查找运营商文档字段
        Content-Length: 0
    ]]>
</send>

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

<pause />

<send>
    <![CDATA[
        SUBSCRIBE sip:[$POCID]&[field1]@[$DOMAIN] SIP/2.0
        Via: SIP/2.0/UDP [local_ip]:[local_port];rport;branch=[branch]
        From: <sip:[field0]@[$DOMAIN]>;tag=4629583
        To: <sip:[$POCID]&[field1]@[$DOMAIN];session=chat>[peer_tag_param]
        Call-ID: [call_id]
        CSeq: 3 SUBSCRIBE
        Contact: <sip:[field0]@[local_ip]:[$c_port]>
        Max-Forwards: 70
        User-Agent: SIPp client mode
        Accept: 请查找运营商文档字段
        Expires: 0
        Event: conference
        Accept-Contact: 请查找运营商文档字段
        Content-Length: 0
    ]]>
</send>


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



<!--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>

 

5) sip里的rtcp操作, 抢占讲话权限
https://wenku.baidu.com/view/854dd3e55ef7ba0d4a733bed.html

TBCP 消息简要概述

https://blog.csdn.net/wunderup/article/details/5136441

 

6) deregclient_set_c_port.xml

<?xml version="1.0" encoding="utf-8" ?>
<!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 deregclient_set_c_port.xml SIP_Proxy_IP:SIP_Proxy_Port -i 172.16.0.6 -p 5088 -inf callee.csv -set c_port 5088 -m 1-->
    <Global variables="c_port" />
    
    <nop hide="true">
        <action>
            <!--设置EXP的值为3600-->
            <assignstr assign_to="EXP" value="0" />
            <assignstr assign_to="DOMAIN" value="运营商域名" />
        </action>
    </nop>
    
  <send>
    <![CDATA[
      REGISTER sip:[$DOMAIN] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[field0]@[$DOMAIN]>;tag=acknnkkg.[call_number]
      To: <sip:[field0]@[$DOMAIN]>
      Call-ID: [call_id]
      CSeq: 1 REGISTER
      Contact: <sip:[field0]@[local_ip]:[$c_port];line=79169130b56d431>
      Max-Forwards: 70
      Subject: Reg Performance Test made by wangwei
      user-agent: SIPp client
      Digest username="sip:[field0]@[$DOMAIN]", realm="[$DOMAIN]", uri="sip:[$DOMAIN]"
      Expires: [$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:[$DOMAIN] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[field0]@[local_ip]:[$c_port];line=79169130b56d431>
      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、sipp xml正则表达式获取接收的信息

<recv response="200">

    <action> 
    <ereg regexp="\r\n\r\n(.*)" search_in="msg" assign_to="sdp_info" />
    <!--
    <ereg regexp=".*" search_in="msg" body="" assign_to="1" />
    <ereg regexp=".*" search_in="hdr" header="CSeq:" check_it="true" assign_to="2" />
    <exec command="echo [$1] >> from_list.log"/>-->
    <exec command="echo '[$sdp_info]' >> from_list.log"/>
    </action>

</recv>