• 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="
    
    (.*)" 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>
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  • 原文地址:https://www.cnblogs.com/dong1/p/10250789.html
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