1:<?xml version="1.0" encoding="ISO-8859-1" ?>
   2:<!DOCTYPE scenario SYSTEM "sipp.dtd">
   3:
   4:<!-- This program is free software; you can redistribute it and/or      -->
   5:<!-- modify it under the terms of the GNU General Public License as     -->
   6:<!-- published by the Free Software Foundation; either version 2 of the -->
   7:<!-- License, or (at your option) any later version.                    -->
   8:<!--                                                                    -->
   9:<!-- This program is distributed in the hope that it will be useful,    -->
  10:<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
  11:<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
  12:<!-- GNU General Public License for more details.                       -->
  13:<!--                                                                    -->
  14:<!-- You should have received a copy of the GNU General Public License  -->
  15:<!-- along with this program; if not, write to the                      -->
  16:<!-- Free Software Foundation, Inc.,                                    -->
  17:<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
  18:<!--                                                                    -->
  19:<!--                 Sipp default 'uas' scenario.                       -->
  20:<!--                                                                    -->
  21:
  22:<scenario name="Basic UAS responder">
  23:  <!-- By adding rrs="true" (Record Route Sets), the route sets         -->
  24:  <!-- are saved and used for following messages sent. Useful to test   -->
  25:  <!-- against stateful SIP proxies/B2BUAs.                             -->
  26:  <recv request="INVITE" crlf="true">
  27:  </recv>
  28:
  29:  <!-- The '[last_*]' keyword is replaced automatically by the          -->
  30:  <!-- specified header if it was present in the last message received  -->
  31:  <!-- (except if it was a retransmission). If the header was not       -->
  32:  <!-- present or if no message has been received, the '[last_*]'       -->
  33:  <!-- keyword is discarded, and all bytes until the end of the line    -->
  34:  <!-- are also discarded.                                              -->
  35:  <!--                                                                  -->
  36:  <!-- If the specified header was present several times in the         -->
  37:  <!-- message, all occurences are concatenated (CRLF seperated)        -->
  38:  <!-- to be used in place of the '[last_*]' keyword.                   -->
  39:
  40:  <send>
  41:    <![CDATA[
  42:
  43:      SIP/2.0 180 Ringing
  44:      [last_Via:]
  45:      [last_From:]
  46:      [last_To:];tag=[call_number]
  47:      [last_Call-ID:]
  48:      [last_CSeq:]
  49:      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
  50:      Content-Length: 0
  51:
  52:    ]]>
  53:  </send>
  54:
  55:  <send retrans="500">
  56:    <![CDATA[
  57:
  58:      SIP/2.0 200 OK
  59:      [last_Via:]
  60:      [last_From:]
  61:      [last_To:];tag=[call_number]
  62:      [last_Call-ID:]
  63:      [last_CSeq:]
  64:      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
  65:      Content-Type: application/sdp
  66:      Content-Length: [len]
  67:
  68:      v=0
  69:      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
  70:      s=-
  71:      c=IN IP[media_ip_type] [media_ip]
  72:      t=0 0
  73:      m=audio [media_port] RTP/AVP 0
  74:      a=rtpmap:0 PCMU/8000
  75:
  76:    ]]>
  77:  </send>
  78:
  79:  <recv request="ACK"
  80:        optional="true"
  81:        rtd="true"
  82:        crlf="true">
  83:  </recv>
  84:
  85:  <recv request="BYE">
  86:  </recv>
  87:
  88:  <send>
  89:    <![CDATA[
  90:
  91:      SIP/2.0 200 OK
  92:      [last_Via:]
  93:      [last_From:]
  94:      [last_To:]
  95:      [last_Call-ID:]
  96:      [last_CSeq:]
  97:      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
  98:      Content-Length: 0
  99:
 100:    ]]>
 101:  </send>
 102:
 103:  <!-- Keep the call open for a while in case the 200 is lost to be     -->
 104:  <!-- able to retransmit it if we receive the BYE again.               -->
 105:  <pause milliseconds="4000"/>
 106:
 107:
 108:  <!-- definition of the response time repartition table (unit is ms)   -->
 109:  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
 110:
 111:  <!-- definition of the call length repartition table (unit is ms)     -->
 112:  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
 113:
 114:</scenario>
 115: