voice-gateway/tests/pjsua/scripts-sipp/uas-answer-183-wrong-codec.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             -->
<!--                                                                    -->
<!-- Correct call cancellation on wrong SDP answer PR #3137             -->

<scenario name="UAS sending 183 with incompatible codec">
  <!-- 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>

  <send>
    <![CDATA[

      SIP/2.0 100 Trying
      [last_Via:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_CSeq:]
      Content-Length: 0

    ]]>
  </send>

  <send>
    <![CDATA[

      SIP/2.0 180 Ringing
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: sip:sipp@[local_ip]:[local_port]
      Content-Length: 0

    ]]>
  </send>

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

      SIP/2.0 183 Session Progress
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: sip:sipp@[local_ip]:[local_port]
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=- 3441953879 3441953879 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[media_ip_type] [media_ip]
      t=0 0
      m=audio [media_port] RTP/AVP 109 105
      a=rtpmap:109 EVS/16000
      a=fmtp:109 br=5.9-24.4; bw=nb-wb; max-red=220
      a=rtpmap:105 telephone-event/16000
      a=fmtp:105 0-15
      a=sendrecv

    ]]>
  </send>

  <!-- Wait for CANCEL -->
  <recv request="CANCEL" crlf="true">
  </recv>

  <send>
    <![CDATA[
      SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_CSeq:]
      Content-Length: 0

    ]]>
  </send>

  <send>
    <![CDATA[

      SIP/2.0 487 Request Terminated
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      CSeq: [cseq] INVITE
      Contact: <sip:sipp@[local_ip]:[local_port]>
      Content-Length: 0
    ]]>
  </send>

  <!-- Wait for CANCEL -->
  <recv request="ACK" crlf="true">
  </recv>

  <!-- 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>