用Sipp 对Asterisk 进行性能测试的工作笔记-2

xml 文件 . 本文草本出自狼山客人, 保留其相关注释.

<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<!-- This program is free software; you can redistribute it and/or      -->
<!-- modify it under the terms of the GNU General Public License as     -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version.                    -->
<!--                                                                    -->
<!-- This program is distributed in the hope that it will be useful,    -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
<!-- GNU General Public License for more details.                       -->
<!--                                                                    -->
<!-- You should have received a copy of the GNU General Public License  -->
<!-- along with this program; if not, write to the                      -->
<!-- Free Software Foundation, Inc.,                                    -->
<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
<!--                                                                    -->
<!--                 Sipp default 'uac' scenario.                       -->
<!--                                                                    -->
<scenario name="Basic Sipstone UAC">

<!--狼山客人：下面这一块表示SIPp发送一个INVITE数据包到SIP server(Wavesplitter的MSP-16)-->
  <send>
    <![CDATA[
   
      INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch];rport
      From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
      To: [field1] <sip:[field1]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]              
      Cseq: 1 INVITE                  
      Contact: sip:[field0]@[local_ip]:[local_port]
      Max-Forwards: 70                
      Subject: Performance Test       
      Content-Type: application/sdp   
      Content-Length: [len]           

      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-
      t=0 0
      c=IN IP[media_ip_type] [media_ip]
      m=audio [media_port] RTP/AVP 0
      a=rtpmap:0 G726/8000

    ]]>
  </send>

<!--狼山客人：下面这一块表示SIPp在等待SIP server返回一个100的数据包-->

  <recv response="100"> optional="true"
  </recv>

<!--狼山客人：下面这一块表示SIPp在等待SIP server返回一个200的数据包，如果收到，说明ViVoice公司的VENUS NW800视频电话已经接听了，用户已经提起话筒-->

  <recv response="200">
  </recv>

<!--狼山客人：下面这一块表示SIPp开始通话-->

  <!-- Packet lost can be simulated in any send/recv message by         -->
  <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
  <send>
    <![CDATA[

      ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sipp <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
      To: sut <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Contact: sip:[field0]@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>

 <!--狼山客人：下面这一块表示SIPp开始发送语音RTP stream，在VENUS NW800视频电话上可以听到不知道哪国的老外的声音-->

  <!-- Play a pre-recorded PCAP file (RTP stream)       -->
  <nop>
    <action>
      <exec play_pcap_audio="pcap/g711a.pcap"/>
    </action>
  </nop>

<!--狼山客人：暂停10秒钟(10000)，一小时， 3600000， 等待播放语音完毕-->
<pause milliseconds="300000"/>

<!--狼山客人：下面这一块发送BYE信号，这是挂断电话信号-->

    <send retrans="500">
     <![CDATA[

      BYE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port]
      From: sipp  <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
      To: sip <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      Cseq: 2 BYE
      Contact: sip:[field0]@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
   </send>

<!--狼山客人：下面这一块表示挂断完毕-->

   <recv response="200">
   </recv>

</scenario>

里面的业务逻辑需要自己确认, 最好的方法是通过抓包(比如ethereal), 分析其工作流. 不同情况下, 返回的信息有所不同, 需要相应调整. 

上面是个呼入后,播放IVR 的案例, 比较简单.

下面是register 的例子(要测试通话, 就要先注册分机号)

<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<!-- This program is free software; you can redistribute it and/or      -->
<!-- modify it under the terms of the GNU General Public License as     -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version.                    -->
<!--                                                                    -->
<!-- This program is distributed in the hope that it will be useful,    -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
<!-- GNU General Public License for more details.                       -->
<!--                                                                    -->
<!-- You should have received a copy of the GNU General Public License  -->
<!-- along with this program; if not, write to the                      -->
<!-- Free Software Foundation, Inc.,                                    -->
<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
<!--                                                                    -->
<!--                 Sipp default 'branchc' scenario.                   -->
<!--                                                                    -->

<scenario name="Basic Sipstone UAC">
  <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];rport
      From: [field0] <sip:[field0]@[remote_ip]:[remote_port]>;tag=[call_number]
      To: [field0] <sip:[field0]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: [cseq] REGISTER
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-d87543-717507386-1--d87543-;rport     
      Contact: <sip:[field0]@[local_ip]:[local_port]>
      Expires: 1200
      Max-Forwards: 70
      User-Agent: eyeBeam release 3004t stamp 16741
      Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,NOTIFY,MESSAGE,SUBSCRIBE,INFO
      Content-Length: 0

    ]]>
  </send>
 
  <recv response="100"> optional="true"
  </recv>
 <recv response="200" crlf="true">
  </recv>

  </recv>
</scenario>

Register 后,  先把被叫号码启动, 进入守护状态, 等到有电话呼叫(INVITE) 后, 震铃接通, 并发送数据. 产生通话效果.

<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<!-- This program is free software; you can redistribute it and/or      -->
<!-- modify it under the terms of the GNU General Public License as     -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version.                    -->
<!--                                                                    -->
<!-- This program is distributed in the hope that it will be useful,    -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
<!-- GNU General Public License for more details.                       -->
<!--                                                                    -->
<!-- You should have received a copy of the GNU General Public License  -->
<!-- along with this program; if not, write to the                      -->
<!-- Free Software Foundation, Inc.,                                    -->
<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
<!--                                                                    -->
<!--                 Sipp default 'branchs' scenario.                   -->
<!--                                                                    -->

<scenario name="branch_server">
  <!-- Set variable 3 if the ua is of the form ua2... -->
  <recv request="INVITE" crlf="true">
  </recv>

  <!-- send 180 then trying if variable 3 is set -->
  <send next="1">
    <![CDATA[

      SIP/2.0 180 Ringing
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      [last_Via:]     
      Contact: <sip:[field0]@[local_ip]:[local_port]>
      Content-Length: 0

    ]]>
  </send>

  <label id="1"/>

  <send retrans="500">
    <![CDATA[

      SIP/2.0 200 OK
      [last_From:]
      [last_To:];tag=800000[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      [last_Via:]     
      Contact: <sip:[field0]@[local_ip]:[local_port]>
      Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,NOTIFY,MESSAGE,SUBSCRIBE,INFO
      Content-Type: application/sdp
      Content-Length: 136

      v=0
      o=- 2372001 2372043 IN IP4 [local_ip]
      s=eyeBeam
      c=IN IP4 [local_ip]
      t=0 0
      c=IN IP4 [media_ip]
      m=audio [media_port] RTP/AVP 0
      a=fmtp:101 0-15
      a=rtpmap:100 speex/16000
      a=rtpmap:101 telephone-event/8000
      a=sendrecv     

    ]]>
  </send>

  <recv request="ACK"
        rtd="true"
        crlf="true">
  </recv>

  <label id="2"/>
  <nop>
    <action>
      <exec play_pcap_audio="pcap/g711a.pcap"/>
    </action>
  </nop>
  <pause milliseconds="7000" next="2" />
 
  <!-- Keep the call open for a while in case the 200 is lost to be     -->
  <!-- able to retransmit it if we receive the BYE again.               -->
   <!-- definition of the response time repartition table (unit is ms)   -->
<!--  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/> -->

  <!-- definition of the call length repartition table (unit is ms)     -->
<!--  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> -->
 
  <ResponseTimeRepartition value="20, 40, 60, 80, 100, 200, 400, 800"/>
  <CallLengthRepartition value="20, 100, 200, 1000, 2000, 10000, 20000"/>
</scenario>

被叫准备好以后, 主叫开始呼叫

<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<!-- This program is free software; you can redistribute it and/or      -->
<!-- modify it under the terms of the GNU General Public License as     -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version.                    -->
<!--                                                                    -->
<!-- This program is distributed in the hope that it will be useful,    -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
<!-- GNU General Public License for more details.                       -->
<!--                                                                    -->
<!-- You should have received a copy of the GNU General Public License  -->
<!-- along with this program; if not, write to the                      -->
<!-- Free Software Foundation, Inc.,                                    -->
<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
<!--                                                                    -->
<!--                 Sipp 'uac' scenario with pcap (rtp) play           -->
<!--                                                                    -->

<scenario name="UAC with media">
  <!-- In client mode (sipp placing calls), the Call-ID MUST be         -->
  <!-- generated by sipp. To do so, use [call_id] keyword.                -->
  <send retrans="500">
    <![CDATA[

      INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch];rport
      From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
      To: [field1] <sip:[field1]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      Cseq: 1 INVITE
      Contact: sip:[field0]@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[local_ip_type] [local_ip]
      t=0 0
      m=audio [auto_media_port] RTP/AVP 8
      a=rtpmap:8 PCMA/8000
      a=rtpmap:101 telephone-event/8000
      a=fmtp:101 0-11,16

    ]]>
  </send>

  <recv response="100" optional="true">
  </recv>
<!--
  <recv response="180" optional="true">
  </recv>
-->
  <!-- By adding rrs="true" (Record Route Sets), the route sets         -->
  <!-- are saved and used for following messages sent. Useful to test   -->
  <!-- against stateful SIP proxies/B2BUAs.                             -->
  <recv response="200" rtd="true" crlf="true">
  </recv>

  <!-- Packet lost can be simulated in any send/recv message by         -->
  <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
  <send>
    <![CDATA[

      ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sipp <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
      To: sut <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Contact: sip:[field0]@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>

  <!-- Play a pre-recorded PCAP file (RTP stream)                       -->
  <label id="1"/>
  <nop>
    <action>
      <exec play_pcap_audio="pcap/g711a.pcap"/>
    </action>
  </nop>
   <pause milliseconds="7000" next="1" />
 
 
  <!-- Play an out of band DTMF '1'                                     -->
  <nop>
    <action>
      <exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/>
    </action>
  </nop>

  <pause milliseconds="1000"/>

  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
  <send retrans="500">
    <![CDATA[

      BYE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port]
      From: sipp  <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
      To: sip <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      Cseq: 2 BYE
      Contact: sip:[field0]@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>

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

  <!-- definition of the response time repartition table (unit is ms)   -->
  <!-- <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/> -->

  <!-- definition of the call length repartition table (unit is ms)     -->
  <!-- <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> -->

 <ResponseTimeRepartition value="20, 40, 60, 80, 100, 200, 400, 800"/>
  <CallLengthRepartition value="20, 100, 200, 1000, 2000, 10000, 20000"/>

</scenario>

编码对性能影响:

      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[local_ip_type] [local_ip]
      t=0 0
      m=audio [auto_media_port] RTP/AVP 8
      a=rtpmap:8 PCMA/8000      //语音发送, 修改PCMA 为G726 等, 不过这个需要asterisk server 支持
      a=rtpmap:101 telephone-event/8000
      a=fmtp:101 0-11,16

通话录音性能:

extensions.conf 中选择通话中录音的方式就可以了. 录音文件保存在: /var/spool/asterisk/monitor

测试结果:

IVR: 500 路
一对一通话: 500 路
录音: 300 路
编解码转换对性能影响较大, 具体和编码方式相关.

有问题可以和我联系: [e-mail]