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

<!-- This program is free software; you can redistribute it and/or      -->
<!-- modify it under the terms of the GNU General Public License as     -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version.                    -->
<!--                                                                    -->
<!-- This program is distributed in the hope that it will be useful,    -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
<!-- GNU General Public License for more details.                       -->
<!--                                                                    -->
<!-- You should have received a copy of the GNU General Public License  -->
<!-- along with this program; if not, write to the                      -->
<!-- Free Software Foundation, Inc.,                                    -->
<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
<!--                                                                    -->
<!--                 Sipp default 'uac' scenario.                       -->
<!--                                                                    -->

<scenario name="Basic Sipstone UAC">
  <!-- In client mode (sipp placing calls), the Call-ID MUST be         -->
  <!-- generated by sipp. To do so, use [call_id] keyword.                -->
  <send retrans="500">
    <![CDATA[
      INVITE sip:test@conference.sip2sip.info SIP/2.0
      From: <sip:2233437105@sip2sip.info>;tag=e27b3
      To: <sip:test@conference.sip2sip.info>
      Call-Id: scb2f9c3f297d0197b29f4591169fcc23
      Cseq: 25077 INVITE
      Session-Expires: 1800
      Min-Expires: 90
      Content-Type: application/sdp
      Content-Length: 375
      Expires: 180
      Date: Tue, 01 Mar 2011 04:19:12 GMT
      Max-Forwards: 69
      User-Agent: wxCommunicator
      Accept-Language: en
      Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,INFO,PRACK,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REGISTER,REFER
      Supported: replaces,timer,100rel,from-change,norefersub
      Via: SIP/2.0/UDP 173.170.11.84:5060;branch=z9hG4bK-83c6ce099e7d;rport
      Contact: <sip:2233437105@173.170.11.84:5060>

      v=0
      o=sipX 5 10 IN IP4 192.168.1.138
      s=call
      c=IN IP4 192.168.1.138
      t=0 0
      m=audio 9000 RTP/AVP 96 97
      a=candidate:0 t UDP 1.0 192.168.1.138 9000
      a=candidate:0 t UDP 1.0 192.168.1.138 9001
      a=candidate:1 t UDP 0.5 5.210.195.106 9000
      a=candidate:1 t UDP 0.5 5.210.195.106 9001
      a=rtpmap:96 telephone-event/8000/1
      a=rtpmap:97 speex/32000/1
      a=fmtp:97 mode=4
      a=ptime:20

    ]]>
  </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" 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
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>

  <!-- This delay can be customized by the -d command-line option       -->
  <!-- or by adding a 'milliseconds = "value"' option here.             -->
  <pause/>

  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
  <send retrans="500">
    <![CDATA[

      BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 2 BYE
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>

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

  <!-- 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>