voice-gateway-service/pjproject/tests/pjsua/scripts-sipp/strict-route.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     -->
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<!-- 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="Strict route test">
  <recv request="INVITE" crlf="true">
  </recv>

  <send>
    <![CDATA[

      SIP/2.0 100 Trying
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
    ]]>
  </send>

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

      SIP/2.0 407 Proxy Authenticate
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Proxy-Authenticate: DIGEST realm="test", nonce="12345", algorithm=MD5
    ]]>
  </send>

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

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

  <send>
    <![CDATA[
      SIP/2.0 100 Trying
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
    ]]>
  </send>

  <send>
    <![CDATA[
      SIP/2.0 180 Ringing
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
    ]]>
  </send>

  <send>
    <![CDATA[

      SIP/2.0 183 progress
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:target@[local_ip]>
      Record-route: <sip:proxy@[local_ip]:[local_port]>
      Content-Type: application/sdp

      v=0
      o=- 3442013205 3442013205 IN IP4 [local_ip]
      s=pjsip
      c=IN IP4 [local_ip]
      t=0 0
      m=audio 4002 RTP/AVP 0
      a=rtpmap:0 PCMU/8000
    ]]>
  </send>

  <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:target@[local_ip]>
      Record-route: <sip:proxy@[local_ip]:[local_port];maddr=[local_ip]>
      Content-Type: application/sdp

      v=0
      o=- 3442013205 3442013205 IN IP4 [local_ip]
      s=pjsip
      c=IN IP4 [local_ip]
      t=0 0
      m=audio 4002 RTP/AVP 0
      a=rtpmap:0 PCMU/8000
    ]]>
  </send>

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

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

  <send>
    <![CDATA[
      SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
    ]]>
  </send>


  <!-- 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="1000"/>


  <!-- 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>