<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<scenario name="Re-Invite problem 1">

<recv request="INVITE" crlf="true" rrs="true">
</recv>



<send>
<![CDATA[

SIP/2.0 100 Trying
[last_Via:]
[last_Call-ID:]
[last_From:]
[last_To:]
[last_CSeq:]
Content-Length: 0

]]>
</send>

<send retrans="500">
<![CDATA[

SIP/2.0 200 OK
[last_Via:]
[last_Call-ID:]
[last_From:]
[last_To:];tag=[call_number]
[last_CSeq:]
[last_Record-Route]
Contact: <sip:bansallaptop@[local_ip]:[local_port];user=phone>
Content-Type: application/sdp
Content-Length: [len]

v=0
o=HuaweiSoftX3000 6644052 6644052 IN IP[local_ip_type] [local_ip]
s=Sip Call
c=IN IP4 127.0.0.1
t=0 0
m=audio 8000 RTP/AVP 0 103
a=rtpmap:0 PCMU/8000
a=rtpmap:103 telephone-event/8000

]]>
</send>

<recv request="ACK"
      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:[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="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>

