uac-media-recv-bye.xml

<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="MobileUC 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=[branch]
      From: +27877520000 <sip:+27877520000@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
      To: [service] <sip:[service]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Contact: sip:+27877520000@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Call Generator
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[local_ip_type] [local_ip]
      t=0 0
      m=audio [auto_media_port] RTP/AVP 8 101
      a=rtpmap:8 PCMA/8000
      a=rtpmap:101 telephone-event/8000
      a=fmtp:101 0-11,16
	  a=ptime:30

    ]]>
  </send>

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

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

  <recv response="183" optional="true">
  </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" rrs="true">
  <action>
  <ereg regexp="([0-9]{1,3}\.){3}[0-9]{1,3}:[0-9]*"  search_in="hdr" header="Contact:" check_it="true" assign_to="1" />
  </action>
  </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:[$1] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
      From: +27877520000 <sip:+27877520000@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
      To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      [routes]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Max-Forwards: 70
      Content-Length: 0

    ]]>
  </send>


  <!-- Play a pre-recorded PCAP file (RTP stream)-->
  <nop>
    <action>
      <exec play_pcap_audio="pcap/g711a.pcap"/>
    </action>
  </nop>

  <recv request="BYE">
  </recv>
  <!-- Send 200OK back -->
  <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>

  <!-- 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>