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2014-12-29 15:50:53


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

  3. <!-- This program is free software; you can redistribute it and/or -->
  4. <!-- modify it under the terms of the GNU General Public License as -->
  5. <!-- published by the Free Software Foundation; either version 2 of the -->
  6. <!-- License, or (at your option) any later version. -->
  7. <!-- -->
  8. <!-- This program is distributed in the hope that it will be useful, -->
  9. <!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
  10. <!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
  11. <!-- GNU General Public License for more details. -->
  12. <!-- -->
  13. <!-- You should have received a copy of the GNU General Public License -->
  14. <!-- along with this program; if not, write to the -->
  15. <!-- Free Software Foundation, Inc., -->
  16. <!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
  17. <!-- -->
  18. <!-- Sipp default 'uas' scenario. -->
  19. <!-- -->

  20. <scenario name="Out-of-call UAS">

  21.   <recv request="MESSAGE|INVITE" regexp_match="true">
  22.   <action>
  23.   <ereg regexp="MESSAGE" search_in="hdr" header="CSeq:" assign_to="1"/>
  24.   <ereg regexp="INVITE" search_in="hdr" header="CSeq:" assign_to="2"/>
  25.   </action>
  26.   </recv>

  27.   <nop next="1" test="1"/>
  28.   <nop next="2" test="2"/>

  29.   <label id="1"/>
  30.   <send next="9">
  31.     <![CDATA[
  32.       SIP/2.0 200 OK
  33.       [last_Via:]
  34.       [last_From:]
  35.       [last_To:]
  36.       [last_Call-ID:]
  37.       [last_CSeq:]
  38.       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
  39.       Content-Length: 0

  40.     ]]>
  41.   </send>

  42.   <label id="2"/>
  43.   <send retrans="500" next="9">
  44.     <![CDATA[

  45.       SIP/2.0 200 OK
  46.       [last_Via:]
  47.       [last_From:]
  48.       [last_To:];tag=[call_number]
  49.       [last_Call-ID:]
  50.       [last_CSeq:]
  51.       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
  52.       Content-Type: application/sdp
  53.       Content-Length: [len]

  54.       v=0
  55.       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
  56.       s=-
  57.       c=IN IP[media_ip_type] [media_ip]
  58.       t=0 0
  59.       m=audio [media_port] RTP/AVP 0
  60.       a=rtpmap:0 PCMU/8000

  61.     ]]>
  62.   </send>

  63.   <label id="9"/>

  64.   <!-- Keep the call open for a while in case the 200 is lost to be -->
  65.   <!-- able to retransmit it if we receive the BYE again. -->
  66.   <timewait milliseconds="4000"/>


  67.   <!-- definition of the response time repartition table (unit is ms) -->
  68.   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>

  69.   <!-- definition of the call length repartition table (unit is ms) -->
  70.   <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>

  71. </scenario>

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