uas-media-wav.xml

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

<scenario name="UAS - Media">
  <recv request="INVITE" crlf="true"></recv>

  <send>
    <![CDATA[

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

    ]]>
  </send>

  <pause distribution="uniform" min="1000" max="3000"/>
 
  <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]>
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=sipp 87308505 1 IN IP[local_ip_type] [local_ip]
      s=-
      t=0 0
      m=audio [media_port] RTP/AVP 8 101
      c=IN IP[media_ip_type] [media_ip]
      a=rtpmap:8 PCMA/8000
      a=rtpmap:101 telephone-event/8000
      a=fmtp:101 0-15
      a=ptime:20

    ]]>
  </send>

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

  <nop>
    <action>
        <exec rtp_stream="/root/sipp/wav/g711a-client.wav" />
    </action>
  </nop>

  <recv request="BYE">
  </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-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="2000"/>

</scenario>