用sipp对Asterisk进行性能测试（转）

测试目标:

1.  IVR支持多少路
2. 一对一通话,支持多少路
3. 不同编解码的性能影响.
4.通话中,录音,支持多少路.

测试工具: sipphttp://sipp.sourceforge.net/

辅助工具: Xlite

SIP rfc:http://www.ietf.org/rfc/rfc3261.txt

RTP for AVhttp://www.ietf.org/rfc/rfc3551.txt

环境:

CPU: xeon 51101.6G*2 ,1 GMEM物理机
Asterisk1.4.7

Asterisk基本操作:

启动: safe_asterisk,或者asterisk -vvvc

如果是后台启动,连接监控: astersisk -r

关闭:在控制栏输入stop now

Asterisk配置:

关注两个配置文件(/etc/asterisk):

sip.conf         // sip分机号设置      
extensions.conf   // dail plan设置,控制呼入后是什么动作

sip.conf添加2000个分机号,以便模拟1000人呼叫(呼叫,应答)

[1000]
type=friend
host=dynamic
context=incoming   //和extensions.conf中对应
canreinvite=no   //如果设置为yes,双方通话信息会直接进行,而不通过asterisk.设置成no,表示所有交互都通过Asterisk.

[1001]
type=friend
host=dynamic
context=incoming
canreinvite=no

extensions.conf 这里列举了多种呼叫计划,包括IVR, 拨号通话,通话录音等.

[incoming]
;play hello world forever
exten => _XXXX,1,answer()
exten => _XXXX,2,playback(hello-world)
exten => _XXXX,3,goto(OneToOne,_XXXX,1)

;[typetest]
;exten => 1111,1,Wait(2)
;exten => 1111,2,Record(/tmp/asterisk-recording:gsm)
;exten => 1111,3,Hangup
;exten => 1112,1,Wait(2)
;exten => 1112,n,Playback(/tmp/asterisk-recording)
;exten => 1112,n,Hangup

;[typetest2]
;exten => _XXXX,1,answer()
;exten => _XXXX,2,dial(sip/${EXTEN},10,r)

;[typetest3]
;exten => 999,1,answer()
;exten => 999,2,dial(sip/${EXTEN},10,r)
;exten => 999,1,Meetme(1234,i,123456)

;[OneToOne]
;exten => _XXXX,1,answer()
;exten => _XXXX,2,mixmonitor(test${EXTEN}.wav|av(0)V(0))
;exten => _XXXX,3,dial(sip/${EXTEN},10,r)
;exten => _XXXX,4,Hangup
;exten => _XXXX,3,Record(/tmp/asterisk-recording${EXTEN}:gsm)

;[IVR]
;exten => _XXXX,1,answer()
;exten => _XXXX,2,playback(hello-world)
;exten => _XXXX,3,goto(IVR,_XXXX,2)

Sipp基本操作:

sipp涉及三个文档(以呼入后,就不停播放IVR语音):

*.bat  批处理命令,方便调用,其中-m参数表示

sipp -sf a16.xml -inf a16.csv -p 5062 -m 200 -i 172.16.3.19910.0.1.4:5060 -trace_err

*.xml   //具体的操作,核心部分,下篇具体介绍xml文件

*.csv  // xml里面引用的参数,以便发起不同呼叫

SEQUENTIAL  
2001;1002;   
2003;1003;
2200;1200;

xml文件 .

<?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">

<!--：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 536557652353687637 INIP[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>

<!--：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"> ōptional="true"
  </recv>

<!--：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>

<!--：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>

 <!--：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>

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

<!--：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>

<!--：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"> ōptional="true"
  </recv>
 <recv response="200" crlf="true">
  </recv>

  </recv>
</scenario>

Register后, 先把被叫号码启动,