sipp压测fresswitch之呼叫篇

根据之前写的一篇微博关于sipp压测fresswitch之账号注册的后续

在用户注册完成后，此时uac和uas都有了自己账号，流程类似（uac是小明）（uas是小红），现在小明给小红拨打电话需要进行几个步骤

大概流程是这样，现在需要两个脚本（呼叫+接听）和两个csv文件（呼叫手机号+接听手机号）
例如：1000—2000是呼叫手机号，3000—4000是接听手机号，那么就是1000呼叫3000这么对应着来。
用上之前注册的csv对应上就可以了

uac.csv
SEQUENTIAL
1000;3000;[authentication username=1000 password=1000]
1001;3001;[authentication username=1001 password=1001]
1002;3002;[authentication username=1002 password=1002]
1003;3003;[authentication username=1003 password=1003]

uas.csv
SEQUENTIAL
3000;3000;[authentication username=3000 password=3000]
3001;3001;[authentication username=3001 password=3001]
3002;3002;[authentication username=3002 password=3002]
3003;3003;[authentication username=3003 password=3003]

呼叫call_uac.xml

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

<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];received=192.168.149.47:7160;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=FreeSWITCH 1574118064 1574118065 IN IP4 47.100.53.252
        s=FreeSWITCH
        c=IN IP4 47.100.53.252
        t=0 0
        m=audio 24714 RTP/AVP 18 101
        7a=rtpmap:18 G729/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20
        a=sendrecv



    ]]>
  </send>

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

  <recv response="180" optional="true">
  </recv>
 
  <recv response="183" 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: [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>

  <!-- pause milliseconds="10000"/  -- >


  <!-- The 'crlf' option inserts a blank line in the statistics report. -->

  <pause milliseconds="60000"/>

  <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>

  <!-- 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>

接听：call_uas.xml

<?xml version="1.0" encoding="ISO-8859-1" ?>

<!DOCTYPE scenario SYSTEM "sipp.dtd">

<scenario name="Basic UAS responder">

<recv request="INVITE" crlf="true">
</recv>
<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]>
      Content-Type: application/sdp
      Content-Length: [len]

	v=0
	o=FreeSWITCH 1574118064 1574118065 IN IP4 47.100.53.252
	s=FreeSWITCH
	c=IN IP4 47.100.53.252
	t=0 0
	m=audio 24714 RTP/AVP 18 101
	7a=rtpmap:18 G729/8000
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16
	a=ptime:20
	a=sendrecv
    ]]>
</send>
<recv request="ACK" optional="true" rtd="true" crlf="true">
  </recv>
<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>
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>

<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>

先启动uas
命令行：uas
sipp -sf call_uas.xml -inf uas_reg.csv 192.168.149**** -m 100000 -i 192.168.149.**** -aa -l 400
命令行：uac
sipp -sf call_uac.xml -inf uac_reg.csv 192.168.149.**** -m 100000 -i 192.168.149.**** -aa -l 400

前面的ip是fresswitch所在ip，后面-i的是本机sipp所在ip

注：如有错误不足地方还请指点修改。

																		2020/3/27