PsapCaller_register.xml 3.27 KB
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<?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="TC_PSAP_SIP_INVITE_BV_02_REGISTER">
  <!-- 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.                             -->

  <send retrans="1000">
    <![CDATA[

      REGISTER sip:[remote_ip] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
      To: caller <sip:[service]@127.0.0.1:5080>
      Call-ID: [call_id]
      CSeq: 1 REGISTER
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Expires: 1800
      User-Agent: SIPp/Linux
      Content-Length: 0

    ]]>
  </send>

  <recv response="401" auth="true">
  </recv>

  <send retrans="1000">
    <![CDATA[

      REGISTER sip:[remote_ip] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
      To: caller <sip:[service]@127.0.0.1:5080>
      Call-ID: [call_id]
      CSeq: 2 REGISTER
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Expires: 1800
      User-Agent: SIPp/Linux
      Content-Length: 0
    ]]>
  </send>

  <recv response="200" rtd="true">
  </recv>

  <pause milliseconds="12000"/>

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  <recv request="BYE" timeout="10000">
  </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>

  <!-- 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>