SIPp压力测试Meeting的场景分析

撰写了基于SIP的meeting压力测试脚本。可以实现同时50个人同时进入meeting，然后说话，12秒后同时退出聊天室，系统等待8秒，再次重复50个人进入会议室，可以实现对meeting的压力测试。

    压力测试流程图如下：

    invite--------------->meeting

    407<------------------meeting

    invite with auth----->meeting

    100<------------------meeting

    user PIN<-------------meeting RTP

    DTMF 1 and #---------->meeting RTP

    RTP------------------->meeting

    pause 18 sec

    Bye------------------->meeting

    200 ok<----------------meeting

    xml文档如下：

<?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]
      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="407" auth="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.                             -->
  <!-- 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: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
      To: [field1] <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>
  
  <send retrans="500">
  <![CDATA[

      INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port]
      From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
      To: [field1] <sip:[field1]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 2 INVITE
      Contact: sip:[field0]@[local_ip]:[local_port]
      [field2]
      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 [auto_media_port] RTP/AVP 0
      a=rtpmap:0 PCMU/8000
    ]]>
 </send>

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

 <recv response="180" optional="true">
 </recv>
  
 <recv response="200" rtd="true" crlf="true">
 </recv>
 
  <!-- Play a pre-recorded PCAP file (RTP stream)                       -->
  <!-- Pause 8 seconds, which is approximately the duration of the      -->
  <!-- PCAP file                                                        -->
  <pause milliseconds="2000"/>

  <!-- Play an out of band DTMF '1'  '#'                               -->
  <nop>
    <action>
      <exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/>
    </action>
  </nop>
  <pause milliseconds="950"/>
  <nop>
    <action>
      <exec play_pcap_audio="pcap/dtmf_2833_pound.pcap"/>
    </action>
  </nop>
  <pause milliseconds="1500"/>
  <nop>
    <action>
      <exec play_pcap_audio="pcap/g711a.pcap"/>
    </action>
  </nop>
  <pause milliseconds="12000"/>
  <!-- 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];branch=[branch]
      From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
      To: [field1] <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>
  <pause milliseconds="8000"/>
  <!-- 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>

 

测试csv批量文件如下：

3001;904;[authentication username=3001 password=3001]
3002;904;[authentication username=3002 password=3002]
3003;904;[authentication username=3003 password=3003]
3004;904;[authentication username=3004 password=3004]
3005;904;[authentication username=3005 password=3005]
3006;904;[authentication username=3006 password=3006]
3007;904;[authentication username=3007 password=3007]
3008;904;[authentication username=3008 password=3008]
3009;904;[authentication username=3009 password=3009]
3010;904;[authentication username=3010 password=3010]
3011;904;[authentication username=3011 password=3011]
3012;904;[authentication username=3012 password=3012]
3013;904;[authentication username=3013 password=3013]
3014;904;[authentication username=3014 password=3014]
3015;904;[authentication username=3015 password=3015]
3016;904;[authentication username=3016 password=3016]
3017;904;[authentication username=3017 password=3017]
3018;904;[authentication username=3018 password=3018]
3019;904;[authentication username=3019 password=3019]
3020;904;[authentication username=3020 password=3020]

注意：这里的904号码为meeting接入号。这里我只写了20个，当然您可以写50，100都没关系。

测试方法与步骤：

(1):首先采用eyebeam呼入904号码，进入后根据密码PIN输入1#，进入会议室，然后听背景音乐

(2):采用sipp压力呼入904号码

    ./sipp -sf meeting.cml -sf meeting.csv -p 6718 -i 192.168.0.254 -m 100000 192.168.0.254 -l 50

(3):此时会听到一批号码登入会议室，然后可以听到一人说话，因为meeting的会议声音是采用叠加技术的，所以会出现很大的回声，之后陆续用户exit，完全退出后会继续听到音乐，因为只有你一个人在会议室了，这说明人已经全部走光了。然后会重复之前的动作

(4):注意 192.168.0.254是发起者呼叫机；192.168.0.194是被测试机。SIpp必须安装在254服务器上才能发起呼叫，另外我这里所指的meeting是指asterisk的conference功能，所以如果有其他软交换要测试会议的话，只需要稍微修改xml文件即可。