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

<scenario name="UAS">
  <!-- You must supply a file containing data for injection -->
  <!-- field0 : dummy index -->
  <!-- field1 : local username -->
  <!-- field2 : local domain -->
  <!-- field3 : authentication -->
  <!-- Sample
SEQUENTIAL
0;1001;test.com;[authentication username=1001 password=1001]
  -->

  <!-- arguments to sipp must include: -->
  <!-- -i : local_ip -->
  <!-- -p : local_port -->
  <!-- -sf : scenario file -->
  <!-- -inf : injection data file -->
  <!--  if no registration, remote host(192.168.2.146) is not needed -->
  <!-- Ex.: sipp -i 192.168.2.122 -p 6060 -sf uas.xml -inf data_s.txt 192.168.2.146 -->

  <send retrans="500">
    <![CDATA[
    REGISTER sip:[field2] SIP/2.0
    Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
    Max-Forwards: 20
    From: "[field1]" <sip:[field1]@[field2]>;tag=[call_number]
    To: "[field1]" <sip:[field1]@[field2]>
    Call-ID: [call_id]
    CSeq: [cseq] REGISTER
    Contact: <sip:[field1]@[local_ip]:[local_port]>
    Expires: 300
    User-Agent: SIPP_S
    Content-Length: 0
    ]]>
  </send>

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

  <send retrans="500">
    <![CDATA[
    REGISTER sip:[field2] SIP/2.0
    Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
    Max-Forwards: 20
    From: "[field1]" <sip:[field1]@[field2]>;tag=[call_number]
    To: "[field1]" <sip:[field1]@[field2]>
    Call-ID: [call_id]
    CSeq: [cseq] REGISTER
    Contact: <sip:[field1]@[local_ip]:[local_port]>
    Expires: 300
    User-Agent: SIPP_S
    [field3]
    Content-Length: 0
    ]]>
  </send>

  <recv response="200" />

  <recv request="NOTIFY" option="true" />

  <!-- 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" rrs="true" crlf="true" />

  <send>
    <![CDATA[
    SIP/2.0 100 Trying
    [last_To:];tag=[call_number]
    [last_From:]
    [last_Call-ID:]
    [last_CSeq:]
    Contact: <sip:[field1]@[local_ip]:[local_port]>
    [last_Via:]
    User-Agent: SIPP_S
    Content-Length: 0
    ]]>
  </send>

  <send>
    <![CDATA[
    SIP/2.0 180 Ringing
    [last_To:];tag=[call_number]
    [last_From:]
    [last_Call-ID:]
    [last_CSeq:]
    [last_Contact:]
    [last_Via:]
    User-Agent: SIPP_S
    Content-Length: 0
    ]]>
  </send>

  <send retrans="500">
    <![CDATA[
    SIP/2.0 200 OK
    [last_Via:]
    [last_To:];tag=[call_number]
    [last_From:]
    [last_Call-ID:]
    [last_CSeq:]
    [last_Contact:]
    User-Agent: SIPP_S
    Content-Type: application/sdp
    Content-Length: [len]

    v=0
    o=CMI-SIPUA 61838 0 IN IP[local_ip_type] [local_ip]
    s=SIP CALL
    c=IN IP[media_ip_type] [media_ip]
    t=0 0
    m=audio [media_port] RTP/AVP 0
    a=rtpmap:0 PCMU/8000
    a=sendrecv
    ]]>
  </send>

  <!-- The [last_Via:] above will cause all Via Headers received in the last message to be added at that point in the SIP message, but they will be combined in a single Header (separated by commas). So the message will not be exactly the same as generated by a real CSP -->

  <recv request="INVITE" optional="true" />
  <!-- the above is used to ignore duplicated INVITE sent by OpenSER -->

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

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

  <send>
      <![CDATA[
        SIP/2.0 200 OK
        [last_Via:]
        [last_From:]
        [last_To:]
        [last_Call-ID:]
        [last_CSeq:]
        Contact: <sip:[field1]@[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.               -->
  <timewait 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>