<?xml version="1.0" encoding="iso-8859-2" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<scenario name="UAC REGISTER + INVITE + call">

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

      REGISTER sip:[remote_ip] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[field0]@[field1]>;tag=[call_number]
      To: <sip:[field0]@[field1]>
      Call-ID: [call_id]
      CSeq: [cseq] REGISTER
      Contact: sip:[field0]@[local_ip]:[local_port]
      Max-Forwards: 100
      Expires: 120
      User-Agent: SIPp/Win32
      Content-Length: 0

    ]]>
  </send>

  <recv response="401" auth="true">
  </recv>

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

      REGISTER sip:[remote_ip] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[field0]@[field1]>;tag=[call_number]
      To: <sip:[field0]@[field1]>
      Call-ID: [call_id]
      CSeq: [cseq] REGISTER
      Contact: sip:[field0]@[local_ip]:[local_port]
      [field2]
      Max-Forwards: 100
      Expires: 120
      User-Agent: SIPp/Win32
      Content-Length: 0

    ]]>
  </send>

  <recv response="200">
  </recv>

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

      INVITE sip:[field3]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sipp <sip:[field0]@[field1]>;tag=[call_number]
      To: <sip:[field3]@[field1]:[remote_port]>
      Call-ID: [call_id]
      CSeq: [cseq] INVITE
      Contact: sip:[field0]@[local_ip]:[local_port]
      [field2]
      Max-Forwards: 100
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=16002 0 0 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[media_ip_type] [media_ip]
      t=0 0
      m=audio [rtpstream_audio_port] RTP/AVP 0 18 9 103 8 101
      a=crypto:[cryptotag1audio] [cryptosuiteaescm128sha1801audio] inline:[cryptokeyparams1audio]
      a=rtcp:[rtpstream_audio_port+1]
      a=sendrecv
      a=rtpmap:0 PCMU/8000
      a=rtpmap:18 G729/8000
      a=rtpmap:9 G722/16000
      a=fmtp:9 bitrate=64000
      a=rtpmap:103 G7221/16000
      a=fmtp:103 bitrate=32000
      a=rtpmap:8 PCMA/8000
      a=rtpmap:101 telephone-event/8000
      a=fmtp:101 0-11,16

    ]]>
  </send>

  <recv response="100">
  </recv>

  <recv response="180">
  </recv>

  <recv response="180">
  </recv>

  <recv response="200">
  </recv>

  <recv response="200">
  </recv>

  <nop>
    <action>
      <exec rtp_echo="startaudio,0,PCMU/8000" />
    </action>
  </nop>

  <nop>
    <action>
      <exec rtp_echo="updateaudio,0,PCMU/8000" />
    </action>
  </nop>

  <send>
    <![CDATA[
      SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_CSeq:]
      Server: VIRTUAL Mitel-3300-ICP 12.0.1.99
      Content-Length: 0
    ]]>
  </send>

  <nop>
    <action>
      <exec rtp_echo="stopaudio,0,PCMU/8000" />
    </action>
  </nop>

  <!-- 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>