uas_video_non_secure.xml

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

<scenario name="Video UAS non-secure">

  <recv request="INVITE" crlf="true">
  </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-Type: application/sdp
      Content-Length: [len]

      v=0
      o=user1 53655765 2353687637 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 18
      a=rtpmap:18 G729/8000 
      a=fmtp:18 annexb=no 
      m=video 17280 RTP/AVP 119
      a=rtpmap:119 H264/90000
      a=fmtp:119 profile-level-id=ABCDEF;sprop-parameter-sets=Z00AKAoWVAPAEPI=,aGFLjyA=;packetization-mode=1


    ]]>
  </send>

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

  <pause milliseconds="1000"/>


  <recv request="BYE"
        rtd="true"
        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>

  <!-- 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>