<?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.                       -->
<!--                                                                    -->
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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             -->


<!--                                                                    -->
<!--   Session timer where UAS doesn't indicate support for UPDATE.     -->
<!--   In this case, UAC MUST use re-INVITE with SDP.                   -->

<scenario name="Basic UAS responder">
  <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:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]> 
      Require: timer
      Session-Expires: 90;refresher=uac
      Content-Type: application/sdp
      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:101 telephone-event/8000
      a=fmtp:101 0-16
    ]]>
  </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:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]> 
      Require: timer
      Session-Expires: 90;refresher=uac
      Content-Type: application/sdp
      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:101 telephone-event/8000
      a=fmtp:101 0-16
    ]]>
  </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>

