From Jose Lopes, 2 Years ago, written in XML.
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  1. <?xml version="1.0" encoding="ISO-8859-1" ?>
  2. <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3.  
  4. <!-- This program is free software; you can redistribute it and/or      -->
  5. <!-- modify it under the terms of the GNU General Public License as     -->
  6. <!-- published by the Free Software Foundation; either version 2 of the -->
  7. <!-- License, or (at your option) any later version.                    -->
  8. <!--                                                                    -->
  9. <!-- This program is distributed in the hope that it will be useful,    -->
  10. <!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
  11. <!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
  12. <!-- GNU General Public License for more details.                       -->
  13. <!--                                                                    -->
  14. <!-- You should have received a copy of the GNU General Public License  -->
  15. <!-- along with this program; if not, write to the                      -->
  16. <!-- Free Software Foundation, Inc.,                                    -->
  17. <!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
  18. <!--                                                                    -->
  19. <!--                 Sipp default 'uac' scenario.                       -->
  20. <!--                                                                    -->
  21.  
  22. <scenario name="Basic Sipstone UAC">
  23.   <!-- In client mode (sipp placing calls), the Call-ID MUST be         -->
  24.   <!-- generated by sipp. To do so, use [call_id] keyword.              -->
  25.   <!--
  26.      Content-Type: application/sdp
  27.  -->
  28.   <send retrans="500">
  29.     <![CDATA[
  30.  
  31.      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  32.      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  33.      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
  34.      To: sut <sip:[service]@[remote_ip]:[remote_port]>
  35.      Call-ID: [call_id]
  36.      CSeq: 1 INVITE
  37.      Contact: sip:sipp@[local_ip]:[local_port]
  38.      Max-Forwards: 70
  39.      Subject: Performance Test
  40.      Content-Type: application/sdp
  41.      Content-Length: [len]
  42.  
  43.      v=0
  44.      o=user1 53655765 53655765 IN IP[local_ip_type] [local_ip]
  45.      s=-
  46.      c=IN IP[media_ip_type] [media_ip]
  47.      t=0 0
  48.      m=audio [media_port] RTP/AVP 0
  49.      a=rtpmap:0 PCMU/8000
  50.  
  51.    ]]>
  52.   </send>
  53.  
  54.   <recv response="100"
  55.        optional="true">
  56.   </recv>
  57.  
  58.   <recv response="180" optional="true">
  59.   </recv>
  60.  
  61.   <recv response="183" optional="true">
  62.   </recv>
  63.  
  64.   <!-- By adding rrs="true" (Record Route Sets), the route sets         -->
  65.   <!-- are saved and used for following messages sent. Useful to test   -->
  66.   <!-- against stateful SIP proxies/B2BUAs.                             -->
  67.   <recv response="200" rtd="true">
  68.   </recv>
  69.  
  70.   <!-- Packet lost can be simulated in any send/recv message by         -->
  71.   <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
  72.   <send>
  73.     <![CDATA[
  74.  
  75.      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  76.      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  77.      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
  78.      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
  79.      Call-ID: [call_id]
  80.      CSeq: 1 ACK
  81.      Contact: sip:sipp@[local_ip]:[local_port]
  82.      Max-Forwards: 70
  83.      Subject: Performance Test
  84.      Content-Length: 0
  85.  
  86.    ]]>
  87.   </send>
  88.  
  89.   <!-- This delay can be customized by the -d command-line option       -->
  90.   <!-- or by adding a 'milliseconds = "value"' option here.             -->
  91.   <pause milliseconds="2000"/>
  92.  
  93.   <send retrans="500">
  94.     <![CDATA[
  95.  
  96.      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  97.      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  98.      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
  99.      To: sut <sip:[service]@[remote_ip]:[remote_port]>
  100.      Call-ID: [call_id]
  101.      CSeq: 2 INVITE
  102.      Contact: sip:sipp@[local_ip]:[local_port]
  103.      Max-Forwards: 70
  104.      Subject: Performance Test
  105.      Content-Type: application/sdp
  106.      Content-Length: [len]
  107.  
  108.      v=0
  109.      o=user1 53655765 53655766 IN IP[local_ip_type] [local_ip]
  110.      s=-
  111.      c=IN IP[media_ip_type] [media_ip]
  112.      t=0 0
  113.      m=audio [media_port] RTP/AVP 0
  114.      a=rtpmap:0 PCMU/8000
  115.      a=sendonly
  116.  
  117.    ]]>
  118.   </send>
  119.  
  120.   <recv response="100"
  121.        optional="true">
  122.   </recv>
  123.  
  124.  
  125.   <!-- By adding rrs="true" (Record Route Sets), the route sets         -->
  126.   <!-- are saved and used for following messages sent. Useful to test   -->
  127.   <!-- against stateful SIP proxies/B2BUAs.                             -->
  128.   <recv response="200" rtd="true">
  129.   </recv>
  130.  
  131.   <!-- Packet lost can be simulated in any send/recv message by         -->
  132.   <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
  133.   <send>
  134.     <![CDATA[
  135.  
  136.      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  137.      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  138.      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
  139.      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
  140.      Call-ID: [call_id]
  141.      CSeq: 2 ACK
  142.      Contact: sip:sipp@[local_ip]:[local_port]
  143.      Max-Forwards: 70
  144.      Subject: Performance Test
  145.      Content-Length: 0
  146.  
  147.    ]]>
  148.   </send>
  149.  
  150.   <pause milliseconds="2000"/>
  151.  
  152.   <send retrans="500">
  153.     <![CDATA[
  154.  
  155.      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  156.      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  157.      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
  158.      To: sut <sip:[service]@[remote_ip]:[remote_port]>
  159.      Call-ID: [call_id]
  160.      CSeq: 3 INVITE
  161.      Contact: sip:sipp@[local_ip]:[local_port]
  162.      Max-Forwards: 70
  163.      Subject: Performance Test
  164.      Content-Type: application/sdp
  165.      Content-Length: [len]
  166.  
  167.      v=0
  168.      o=user1 53655765 53655767 IN IP[local_ip_type] 10.112.51.43
  169.      s=-
  170.      c=IN IP[media_ip_type] 10.112.51.43
  171.      t=0 0
  172.      m=audio [media_port] RTP/AVP 0
  173.      a=rtpmap:0 PCMU/8000
  174.      a=sendrcv
  175.  
  176.    ]]>
  177.   </send>
  178.  
  179.   <recv response="100"
  180.        optional="true">
  181.   </recv>
  182.  
  183.  
  184.   <!-- By adding rrs="true" (Record Route Sets), the route sets         -->
  185.   <!-- are saved and used for following messages sent. Useful to test   -->
  186.   <!-- against stateful SIP proxies/B2BUAs.                             -->
  187.   <recv response="200" rtd="true">
  188.   </recv>
  189.  
  190.   <!-- Packet lost can be simulated in any send/recv message by         -->
  191.   <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
  192.   <send>
  193.     <![CDATA[
  194.  
  195.      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  196.      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  197.      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
  198.      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
  199.      Call-ID: [call_id]
  200.      CSeq: 3 ACK
  201.      Contact: sip:sipp@[local_ip]:[local_port]
  202.      Max-Forwards: 70
  203.      Subject: Performance Test
  204.      Content-Length: 0
  205.  
  206.    ]]>
  207.   </send>
  208.  
  209.   <pause milliseconds="2000"/>
  210.  
  211.  
  212.  
  213.  
  214.   <send retrans="500">
  215.     <![CDATA[
  216.  
  217.      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  218.      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  219.      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
  220.      To: sut <sip:[service]@[remote_ip]:[remote_port]>
  221.      Call-ID: [call_id]
  222.      CSeq: 4 INVITE
  223.      Contact: sip:sipp@[local_ip]:[local_port]
  224.      Max-Forwards: 70
  225.      Subject: Performance Test
  226.      Content-Type: application/sdp
  227.      Content-Length: [len]
  228.  
  229.      v=0
  230.      o=user1 53655765 53655768 IN IP[local_ip_type] [local_ip]
  231.      s=-
  232.      c=IN IP[media_ip_type] [media_ip]
  233.      t=0 0
  234.      m=audio [media_port] RTP/AVP 0
  235.      a=rtpmap:0 PCMU/8000
  236.  
  237.    ]]>
  238.   </send>
  239.  
  240.   <recv response="100"
  241.        optional="true">
  242.   </recv>
  243.  
  244.  
  245.   <!-- By adding rrs="true" (Record Route Sets), the route sets         -->
  246.   <!-- are saved and used for following messages sent. Useful to test   -->
  247.   <!-- against stateful SIP proxies/B2BUAs.                             -->
  248.   <recv response="200" rtd="true">
  249.   </recv>
  250.  
  251.   <!-- Packet lost can be simulated in any send/recv message by         -->
  252.   <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
  253.   <send>
  254.     <![CDATA[
  255.  
  256.      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  257.      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  258.      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
  259.      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
  260.      Call-ID: [call_id]
  261.      CSeq: 4 ACK
  262.      Contact: sip:sipp@[local_ip]:[local_port]
  263.      Max-Forwards: 70
  264.      Subject: Performance Test
  265.      Content-Length: 0
  266.  
  267.    ]]>
  268.   </send>
  269.  
  270.   <recv request="INVITE" crlf="true">
  271.   </recv>
  272.  
  273.   <send retrans="500">
  274.   <![CDATA[
  275.    SIP/2.0 200 OK
  276.    [last_Via:]
  277.    [last_From:]
  278.    [last_To:];tag=[pid]SIPpTag01[call_number]
  279.    [last_Call-ID:]
  280.    [last_CSeq:]
  281.    Contact: <sip:[local_ip]:[local_port];transport=[transport]>
  282.    Content-Type: application/sdp
  283.    Content-Length: [len]
  284.    v=0
  285.    o=user1 53655765 53655769 IN IP[local_ip_type] [local_ip]
  286.    s=-
  287.    c=IN IP[media_ip_type] [media_ip]
  288.    t=0 0
  289.    m=audio [media_port] RTP/AVP 0
  290.    a=rtpmap:0 PCMU/8000
  291.    a=recvonly
  292.  ]]>
  293. </send>
  294.  
  295. <recv request="ACK"
  296.      rtd="true"
  297.      crlf="true">
  298. </recv>
  299.  
  300.  
  301. <recv request="INVITE" crlf="true">
  302. </recv>
  303.  
  304. <send retrans="500">
  305. <![CDATA[
  306.  SIP/2.0 200 OK
  307.  [last_Via:]
  308.  [last_From:]
  309.  [last_To:];tag=[pid]SIPpTag01[call_number]
  310.  [last_Call-ID:]
  311.  [last_CSeq:]
  312.  Contact: <sip:[local_ip]:[local_port];transport=[transport]>
  313.  Content-Type: application/sdp
  314.  Content-Length: [len]
  315.  v=0
  316.  o=user1 53655765 53655770 IN IP[local_ip_type] [local_ip]
  317.  s=-
  318.  c=IN IP[media_ip_type] [media_ip]
  319.  t=0 0
  320.  m=audio [media_port] RTP/AVP 0
  321.  a=rtpmap:0 PCMU/8000
  322. ]]>
  323. </send>
  324.  
  325. <recv request="ACK"
  326.    rtd="true"
  327.    crlf="true">
  328. </recv>
  329.  
  330.  
  331.   <!-- The 'crlf' option inserts a blank line in the statistics report. -->
  332.   <recv request="BYE" crlf="true">
  333.   </recv>
  334.  
  335.   <send>
  336.     <![CDATA[
  337.  
  338.      SIP/2.0 200 OK
  339.      [last_Via:]
  340.      [last_From:]
  341.      [last_To:]
  342.      [last_Call-ID:]
  343.      [last_CSeq:]
  344.      [last_Contact:]
  345.      Content-Length: 0
  346.    ]]>
  347.   </send>
  348.  
  349.   <!-- definition of the response time repartition table (unit is ms)   -->
  350.   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
  351.  
  352.   <!-- definition of the call length repartition table (unit is ms)     -->
  353.   <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
  354.  
  355. </scenario>
  356.