uac_wait_for_hangup.xml

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

<scenario name="Basic Sipstone UAC">
  <send retrans="500">
    <![CDATA[
      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK[call_number]
      From: "[field0]" <sip:[field0]@[field1]>;tag=[call_number]
      To: <sip:[service]@[field1]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Contact: <sip:[field0]@[local_ip]:[local_port]>
      Content-Type: application/sdp
      Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER
      Max-Forwards: 70
      User-Agent: SIPp
      Allow-Events: talk,hold,conference,refer,check-sync
      Content-Length: [len]

      v=0
      o=- 20102 20102 IN IP[local_ip_type] [local_ip]
      s=SDP data
      c=IN IP[media_ip_type] [media_ip]
      t=0 0
      m=audio [media_port] RTP/AVP 0 8 18 9 101
      a=rtpmap:0 PCMU/8000
      a=rtpmap:8 PCMA/8000
      a=rtpmap:18 G729/8000
      a=fmtp:18 annexb=no
      a=rtpmap:9 G722/8000
      a=fmtp:101 0-15
      a=rtpmap:101 telephone-event/8000
      a=ptime:20
      a=sendrecv
    ]]>
  </send>

  <recv response="100" optional="true"></recv>
  <recv response="101" optional="true"></recv>
  <recv response="180" optional="true"></recv>
  <recv response="183" optional="true"></recv>
  <recv response="404" optional="true" next="5"></recv>
  <recv response="480" optional="true" next="5"></recv>
  <recv response="488" optional="true" next="5"></recv>
  <recv response="502" optional="true" next="5"></recv>
  <recv response="503" optional="true" next="6"></recv>
  <recv response="486" optional="true" next="6"></recv>
  <recv response="407" optional="true" next="6"></recv>

  <!-- 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 response="200" rtd="true"></recv>

  <!-- Packet lost can be simulated in any send/recv message by         -->
  <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
  <send>
    <![CDATA[

      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      [last_Via:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_Contact:]
      CSeq: 2 ACK
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>

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

  <recv response="200" crlf="true" next="6"></recv>

  <label id="5"/>
  <send>
    <![CDATA[

      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      [last_Via:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_Contact:]
      CSeq: 2 ACK
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>


  <label id="6"/>

  <!-- 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>