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Sipp를 Windows에 설치하기

1. Sipp에서 pcap을 사용하기 위해서는 WinPcap을 받아 미리 Sipp 설치하기 전에 먼저 설치가 되어있어야 합니다.
 현재 최신버전인 WinPcap 4.1.2 버전을 받아 설치를 합니다.
 설치 버젼 : http://www.winpcap.org/install/bin/WinPcap_4_1_2.exe

2. WinPcap 홈페이지에서 받은 WpdPack_4_1_2.exe 파일을 실행하여 설치합니다.

3. Sipp 홈페이지에서 윈도우 설치용 최신 버전 파일을 받아 설치합니다.
 윈도우용 설치 버전 : http://sourceforge.net/projects/sipp/files/sipp/3.2/sipp-win32-3.2-setup.exe/download

4. 받은 파일 sipp-win32-3.2-setup.exe을 실행하여 설치합니다.

5. C:\Program Files\Sipp_3.2안에 sipp.exe 파일을 실행하면 됩니다.


Sipp 홈페이지 : http://sipp.sourceforge.net/

WinPcap 홈페이지 : http://www.winpcap.org/default.htm

sipp-win32-3.2-setup.exe

WinPcap_4_1_2.exe

Asterisk Stress 테스트 인 Sipp Test시 녹취가 가능하도록 Pcap 까지 플레이 해주는 방법

1. 환경설정

sip.conf
[general]
progressinband=yes ;와이파이 컬러링 문제 때문에 필요  <- 주석 처리

;;클라이언트 단 Peer 설정
[sippuac]  
type=peer
;secret=1111
context=sipptest
username=sippuac
host=XXX.XXX.XXX.XXX
port=5065
dtmfmode=rfc2833
insecure=no
canreinvite=no
nat=no
qualify=no

;;서버 단 Peer 설정
[sippuas]
type=peer
context=sipptest
username=sippuas
host=XXX.XXX.XXX.XXX
port=5066
;dtmfmode=rfc2833
insecure=no
;canreinvite=no
nat=no
qualify=no

extensions.conf
[sipptest]
exten=>s,1,Noop(SIPp Test)
exten=>s,n,Noop(=========================SIPp Test)
;;exten=>s,n,AGI(agi://localhost/managertest.agi)
;exten=>s,n,Dial(SIP/9805&SIP/9806&SIP/9807&SIP/9809,8,twWR)
exten=>s,n,Dial(SIP/9805&SIP/9806&SIP/9807&SIP/9809,8,R)
exten=>s,n,Dial(SIP/sippuas,8)
exten=>s,n,Hangup

2. 실행 명령어
클라이언트 단 실행 명령어
C:\Program Files\Sipp_3.1>sipp -s s -sf pcap.xml -p 5065 -i XXX.XXX.XXX.XXX xxx.xxx.xxx.xxx -r 5 -rp 1000 -nr -mi XXX.XXX.XXX.XXX -mp 6006

서버단 실행 명령어
C:\Program Files\Sipp_3.1>sipp -sn uas -p 5066 -mi XXX.XXX.XXX.XXX -mp 6003 -i XXX.XXX.XXX.XXX -rtp_echo -nr

3. 실행 시나리오 ( 서버는 기본, 클라이언트는 이름부분 수정)
----------------------------- pcap.xml -----------------------------------
<?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:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sippuas <sip:sippuas@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
      To: sippuac <sip:[service]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Contact: sip:sippuas@[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 8 101
      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:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sippuas <sip:sippuas@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
      To: sippuac <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Contact: sip:sippuas@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>

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

  <!-- Pause 8 seconds, which is approximately the duration of the      -->
  <!-- PCAP file                                                        -->
  <pause milliseconds="20000"/>

  <!-- 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:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sippuas <sip:sippuas@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
      To: sippuac <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 2 BYE
      Contact: sip:sippuas@[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>

# 녹음 또는 녹화 되어 있는 Pcap을 플레이.
# 활용 방안은 오디오 같은 경우 일정 음성을 들려주고 Hangup 한다던가, 비디오의 경우 무료 전화 서비스시 일정 광고를 보여준 후 , 통화를 하도록 설정 할 수 있음.
# 이후 고민 사항....
 (1) 전화번호나 어떤 특정 조건에 따른 다른 오디오나 비디오가 나오도록 해야할것
 (2) 현재 데몬을 여러개 띄우고 각기 다른 번호를 이용하면 가능하나 비효율적

1. Asterisk 환경 설정
sip.conf
;; Server 만 설정, 1111로 전화만 받는 peer 등록
[1111]
type=peer
context=outbound
username=1111
host=XXX.XXXX.XXX.XXX
port=5067
dtmfmode=rfc2833
insecure=very
canreinvite=no
nat=yes
qualify=no
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=h263
allow=h263a
videosupport=yes

Dialplan의 경우 그냥 Dial을 이용했으므로 크게 변경 사항 없음. ( 각 내선번호의 Peer로 등록하였기 때문에 )

2. SIPP 설정
가. 시나리오  XML
<?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="Modify UAS PCAP PLAY kimos">
  <!-- 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=[pid]SIPpTag01[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=[pid]SIPpTag01[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=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[media_ip_type] [media_ip]
      t=0 0
      m=audio [media_port] RTP/AVP 0 8 101
      a=fmtp:101 0-15
      a=rtpmap:101 telephone-event/8000
      m=video [media_port] RTP/AVP 34
      a=fmtp:34 QCIF=1;CIF=1
      a=rtpmap:34 H263/90000

    ]]>
  </send>

  <recv request="ACK"
        optional="false"
        rtd="true"
        crlf="true">
  </recv>
 
  <!-- Play a pre-recorded PCAP file (RTP stream)                       -->
  <nop>
    <action>
      <exec play_pcap_video="pcap/b.pcap"/>
    </action>
  </nop>

  <!-- Pause 8 seconds, which is approximately the duration of the      -->
  <!-- PCAP file                                                        -->
  <pause milliseconds="80000"/>

  <send retrans="500">
    <![CDATA[

      BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 2 BYE
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>

  <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.               -->
  <timewait milliseconds="4000"/>

  <!-- 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>

 # 시나리오에 쓰인 PCAP 파일들

b.pcap

orgvideo.pcap

video123.pcap

1.Asterisk 설정 환경 파일 설정 ( /etc/asterisk/ )
sip.conf 설정 ( 유저 등록 )
;; 클라이언트 peer 등록
[sipp]
type=peer
username=sippuac
;; sipp 실행하는 PC의 IP와 Port
host=192.168.0.3
port=5065
context=sipptest
dtmfmode=rfc2833
insecure=very
canreinvite=no
nat=yes
qualify=no ;; <== 이부분이 빠지면 실행되지 않음.. 정확이는 잘...

;; 서버 peer 등록
[sippuas]
type=peer
context=sipptest
username=sippuas
;; sipp 실행하는 PC의 IP와 Port
host=192.168.0.3
port=5067
dtmfmode=rfc2833
insecure=very
canreinvite=no
nat=yes
qualify=no

extensions.conf 설정 ( Dialplan 정의 )
[sipptest]
exten=>s,1,Dial(SIP/sippuas,20 )

2.Sipp 설정
sipp 실행명령어
Clent =>
sipp -s s -sn uac -p 5065 -i 192.168.0.3 192.168.0.2 -nr -r 1 -rp 1000
 -s : service name
 -sn : 기본 지원하는 XML 시나리오
 -p : port
 -i : local ip
 -nr : Disable retransmission in UDP mode ( UDP 재전송 모드 해제 )
 -r : rate(cps), 초당 콜수
 -rp : 위의 초 시간을 정함, 단위는 ms. 1000 = 1초임.
Server =>
sipp -sn uas -p 5067 -mp 6003 -i 123.140.245.80 -nr
 -mp : Media Port