Sipp工具实现呼叫中心的性能测试

近期要完成一个涉及呼叫的项目，需要支持访客直接联系客服的功能。

对于测试来说，也需要测试呼叫转发的性能，访客直接拨打接入号，由callserver这边添加路由转发到客服这边。

考虑过使用sipp与winsip两个工具，sipp的话可以自己定制流程，winsip的话可能适合配置型，但是也有很多流程可以配置的，由于以前项目也用过sipp来测试sip消息收发，呼叫也确定使用sipp来测试性能，当然，功能测试也可以用这个，不过功能测试一般直接装个eyebeam来拨就好了。。。

环境：单台服务器部署callserver，另外一台作为sipp的uac以及uas，可以同时运行uac和uas，开两个窗口就好了。

按照以往经验，测试环境系统能够支撑60个坐席，平均每个坐席通话时间60s，系统每秒处理的呼叫量为 60坐席/60s=1 CAPS。
设置3组测试数据：以60个坐席为基准，测试60个坐席以内，60个坐席以上，大于60坐席压力测试。
50坐席容量：0.83 CAPS 60s通话时长
100坐席容量：1.67 CAPS 60s通话时长
200坐席容量：3.33 CAPS 60s通话时长 

测试过程中，修改了通话时长为30s，然后通过修改每秒呼叫量来控制并发的呼叫。实际测试50个并发，一共拨了5000个呼叫没出现问题。但是到100个并发的时候，uac发送过去的invite，由于服务器资源不够，直接回404 not found了，没有直接转给uas。

脚本如下，当然，不是全部自己写的，大部分都是套用模板，然后根据项目需求修改了一下流程：

uac.xml:原本呼叫中心是发两个invite给uac的，取消了播工号，然后注释了一次invite，只发送一次invite。

评论反馈之前的脚本不对应，估计是因为之前是直接改了就copy上来了，没有验证，现在直接从服务器copy了一份正常执行的上来，里面注释了一次invite，应该是可以执行的。

<?xml version="1.0" encoding="ISO-8859-1" ?> 
<!DOCTYPE scenario (View Source for full doctype...)> 

<scenario name="Basic Sipstone UAC">

<send retrans="1000">
<![CDATA[

      INVITE sip:02099635005@[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:02099635005@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Contact: <sip:sipp@[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 response="180" optional="true" /> 
  <recv response="183" optional="true" /> 
<!--  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" /> 
<!--  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:02099635005@[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: "02099635005"<sip:02099635005@[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>
 
 <!--
 <recv request="INVITE" crlf="true">
  </recv>
<send retrans="500">
    <![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]>
      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 request="ACK"
        rtd="true"
        crlf="true">
 </recv>

 
  <nop>
  <action>
    <exec play_pcap_audio="/home/roger/uac/pcap/1.pcap"/>
  </action>
</nop>
-->
<!-- Pause 8 seconds, which is approximately the duration of the      -->
<!-- PCAP file                                                        -->
<pause milliseconds="20000" /> 

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

      BYE sip:02099635005@[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: "02099635005"<sip:02099635005@[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" /> 
<!--  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" /> 
  </scenario>

uas.xml:这个配置的流程，也就根据uac.xml来修改，需要对应请求和响应，基本上启动了就不怎么管了。

<?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 'uas' scenario.                       -->
<!--                                                                    -->

<scenario name="Basic UAS responder">
  <!-- 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 request="INVITE" crlf="true">
  </recv>

  <!-- The '[last_*]' keyword is replaced automatically by the          -->
  <!-- specified header if it was present in the last message received  -->
  <!-- (except if it was a retransmission). If the header was not       -->
  <!-- present or if no message has been received, the '[last_*]'       -->
  <!-- keyword is discarded, and all bytes until the end of the line    -->
  <!-- are also discarded.                                              -->
  <!--                                                                  -->
  <!-- If the specified header was present several times in the         -->
  <!-- message, all occurences are concatenated (CRLF seperated)        -->
  <!-- to be used in place of the '[last_*]' keyword.                   -->

  <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>


  <send retrans="500">
    <![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]>
      Max-Forwards: 70
      Subject: Performance Test     
      Content-Type: application/sdp
      Content-Length: [len]
	  
      v=0
      o=user2 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-sipp
      c=IN IP[local_ip_type] [local_ip]
      t=0 0
      m=audio 22222 RTP/AVP 8
      a=rtpmap:8 PCMA/8000
      a=rtpmap:101 telephone-event/8000
      a=fmtp:101 0-11,16

    ]]>
  </send>
  
 <recv request="ACK"
        rtd="true"
        crlf="true">
 </recv>
 
 <!--<pause milliseconds="20000" />  -->
  <recv request="BYE">
  </recv>

  <send>
    <![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>

  <!-- 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.               -->
  <pause milliseconds="500"/>


  <!-- 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"/>

</scenario>

执行脚本，需要用以下格式添加csv的文件，作为uac侧发起呼叫的号码资源或者用随机号码。

uac1.csv，指定手机号

uac2.csv：随机手机号

--------------------------------------------------------------------分割---------------------------------------------------------------------------------

10.32.121.20---安装sipp的服务器地址

10.32.121.11:7896---sip服务器地址

uas侧命令如下(需要先启动uas，不然uac启动了没对应的接收请求端了)：

./sipp -sn uas  -i 10.32.121.20 -p 5777 -sf /perf/sipp-script/uas.xml

说明：作为uas启动，5666为本地监听uac端的端口。

uac侧命令如下：

./sipp -sn uac 10.32.121.11:7896 -i 10.32.121.20 -p 5666 -m 500 -l 50 -r 3 -rp 1000 -sf  /perf/sipp-script/uac.xml -inf /perf/sipp-script/uac.csv -trace_msg

说明：作为uac启动，每秒一个访客呼入客服，呼叫500次后停止，最大同时并发50个，5666为本地监听uac端的端口；uac.csv 这个文件路径是前面用csv格式保存的uac读取的号码资源文件。

命令参数说明：

-sn uc:作为客户端执行

-sn us:作为服务端执行

172.16.71.21:5060--对方IP

-i: 本地ip

-p：本地端口

-m:发送N次呼叫后停止并退出

-l: 最大同时通话数量，表示最大并发数，即同时可接电话的坐席数量

-r 1 -rp 1000: 每秒发送一个呼叫

-sf:  场景文件路径：

-inf：CSV文件路径