voice-gateway-service/pjproject/tests/pjsua/scripts-sipp/uas-422-then-200-bad-se.xml

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

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<!-- 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.                    -->
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<!-- GNU General Public License for more details.                       -->
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<!-- 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="Basic UAS responder">
  <!-- 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 retrans="500">
    <![CDATA[

      SIP/2.0 422 Session Timer too small
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]> 
      Min-SE:  5400
      Content-Length: 0

	]]>
  </send>

  <recv request="ACK"
        optional="true"
        rtd="true"
        crlf="true"> 
  </recv> 
 

  <recv request="INVITE" crlf="true">
  </recv>

  <send retrans="500"> 
    <![CDATA[

      SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
      Allow-Events: telephone-event
      Contact: <sip:[local_ip]:[local_port];transport=[transport]> 
      Supported: replaces
      Session-Expires:  3600;refresher=uas
      Require: timer
      Content-Type: application/sdp
      Content-Disposition: session;handling=required
      Content-Length: [len]
 
      v=0
      o=Some-UserAgent 68 210 IN IP4 [local_ip]
      s=SIP Call
      c=IN IP4 [local_ip]
      t=0 0
      m=audio 17294 RTP/AVP 0 101
      c=IN IP4 [local_ip]
      a=rtpmap:0 PCMU/8000
      a=rtpmap:101 telephone-event/8000
      a=fmtp:101 0-16
      a=ptime:20 

    ]]>
  </send> 
  
  <recv request="ACK"
        rtd="true"
        crlf="true"> 
  </recv> 


  <!-- 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.               -->
  <pause 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>