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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.     <send retrans="500">
  26.         <![CDATA[
  27.  
  28.      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  29.      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  30.      From: sipp <sip:+1000@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
  31.      To: [service] <sip:[service]@[remote_ip]:[remote_port]>
  32.      Call-ID: [call_id]
  33.      CSeq: 1 INVITE
  34.      Contact: sip:sipp@[local_ip]:[local_port]
  35.      Max-Forwards: 70
  36.      Subject: Performance Test
  37.      Content-Type: application/sdp
  38.      Content-Length: [len]
  39.  
  40.      v=0
  41.      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
  42.      s=-
  43.      c=IN IP[media_ip_type] [media_ip]
  44.      t=0 0
  45.      m=audio [media_port] RTP/AVP 0
  46.      a=rtpmap:0 PCMU/8000
  47.  
  48.    ]]>
  49.     </send>
  50.  
  51.     <recv response="100"
  52.          optional="true">
  53.     </recv>
  54.  
  55.     <recv response="180" optional="true">
  56.     </recv>
  57.  
  58.     <recv response="183" optional="true">
  59.     </recv>
  60.  
  61.     <!-- By adding rrs="true" (Record Route Sets), the route sets         -->
  62.     <!-- are saved and used for following messages sent. Useful to test   -->
  63.     <!-- against stateful SIP proxies/B2BUAs.                             -->
  64.     <recv response="200" rtd="true">
  65.         <!--
  66.        <action>
  67.            <ereg regexp="<(.*)>"
  68.                  search_in="hdr"
  69.                  header="Contact:"
  70.                  check_it="true"
  71.                  assign_to="0,1"/>
  72.        </action>
  73.        -->
  74.     </recv>
  75.  
  76.     <!-- Packet lost can be simulated in any send/recv message by         -->
  77.     <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
  78.     <!--
  79.    <send>
  80.        <![CDATA[
  81.  
  82.      ACK [$1] SIP/2.0
  83.      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  84.      From: sipp <sip:+1000@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
  85.      To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
  86.      Call-ID: [call_id]
  87.      CSeq: 1 ACK
  88.      Contact: sip:sipp@[local_ip]:[local_port]
  89.      Max-Forwards: 70
  90.      Subject: Performance Test
  91.      Content-Length: 0
  92.      X-Contact: [$0]
  93.  
  94.    ]]>
  95.    </send>
  96.    -->
  97.  
  98.     <!-- This delay can be customized by the -d command-line option       -->
  99.     <!-- or by adding a 'milliseconds = "value"' option here.             -->
  100.     <!--
  101.    <pause/>
  102.    -->
  103.  
  104.     <!-- The 'crlf' option inserts a blank line in the statistics report. -->
  105.     <!--
  106.    <send retrans="500">
  107.        <![CDATA[
  108.  
  109.      BYE [$1] SIP/2.0
  110.      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  111.      From: sipp <sip:+1000@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
  112.      To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
  113.      Call-ID: [call_id]
  114.      CSeq: 2 BYE
  115.      Contact: sip:sipp@[local_ip]:[local_port]
  116.      Max-Forwards: 70
  117.      Subject: Performance Test
  118.      Content-Length: 0
  119.      X-Contact: [$0]
  120.  
  121.    ]]>
  122.    </send>
  123.  
  124.    <recv response="200" crlf="true">
  125.    </recv>
  126.    -->
  127.  
  128.     <!-- definition of the response time repartition table (unit is ms)   -->
  129.     <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
  130.  
  131.     <!-- definition of the call length repartition table (unit is ms)     -->
  132.     <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
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