【SIPp】Linux-SIPp3.6.0 测试FreeSwitch

最新推荐文章于 2024-03-15 09:49:11 发布

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最新推荐文章于 2024-03-15 09:49:11 发布

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文章标签： linux centos 运维

本文链接：https://blog.csdn.net/weixin_45616369/article/details/104477275

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Linux下SIPp测试Freeswitch

知识必备：Linux操作系统、Sip协议

一、安装

1．依赖包安装：

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

2、安装libcap

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

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

3．安装libnet （可跳过）

下载libnet ：


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

4、安装sipp

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)

# tar -zxvf sipp-xxx.tar
# cd sipp-xxx
#编译配置：
./configure --with-pcap --with-openssl
#编译&&安装：
make && make install

#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

5、修改系统openfile限制

# 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

6、freeswith配置修改

①、修改max-sessions和sessions-per-second

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

②、修改拨号的正则

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>

③、添加defalut配置文件

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

④、添加白名单无需鉴权

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>

二、SIPp常用参数

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：

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：

-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；

三、注册场景测试

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

1、uac.csv

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

2、uas.csv

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]
...

3、regclient_set_c_port.xml

<?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

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

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

四、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

1、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>    
        <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>

2、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 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>

3、测试呼叫的步骤：

执行先后顺序：

# 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