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 'uac' scenario with pcap (rtp) play           -->
  20:<!--                                                                    -->
  21:
  22:<scenario name="UAC with media">
  23:  <!-- In client mode (sipp placing calls), the Call-ID MUST be         -->
  24:  <!-- generated by sipp. To do so, use [call_id] keyword.                -->
  25:  <send retrans="500">
  26:    <![CDATA[
  27:
  28:      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  29:      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  30:      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
  31:      To: sut <sip:[service]@[remote_ip]:[remote_port]>
  32:      Call-ID: [call_id]
  33:      CSeq: 1 INVITE
  34:      Contact: sip:sipp@[local_ip]:[local_port]
  35:      Max-Forwards: 70
  36:      Subject: Performance Test
  37:      Content-Type: application/sdp
  38:      Content-Length: [len]
  39:
  40:      v=0
  41:      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
  42:      s=-
  43:      c=IN IP[local_ip_type] [local_ip]
  44:      t=0 0
  45:      m=audio [auto_media_port] RTP/AVP 8
  46:      a=rtpmap:8 PCMA/8000
  47:      a=rtpmap:101 telephone-event/8000
  48:      a=fmtp:101 0-11,16
  49:
  50:    ]]>
  51:  </send>
  52:
  53:  <recv response="100" optional="true">
  54:  </recv>
  55:
  56:  <recv response="180" optional="true">
  57:  </recv>
  58:
  59:  <!-- By adding rrs="true" (Record Route Sets), the route sets         -->
  60:  <!-- are saved and used for following messages sent. Useful to test   -->
  61:  <!-- against stateful SIP proxies/B2BUAs.                             -->
  62:  <recv response="200" rtd="true" crlf="true">
  63:  </recv>
  64:
  65:  <!-- Packet lost can be simulated in any send/recv message by         -->
  66:  <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
  67:  <send>
  68:    <![CDATA[
  69:
  70:      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  71:      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  72:      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
  73:      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
  74:      Call-ID: [call_id]
  75:      CSeq: 1 ACK
  76:      Contact: sip:sipp@[local_ip]:[local_port]
  77:      Max-Forwards: 70
  78:      Subject: Performance Test
  79:      Content-Length: 0
  80:
  81:    ]]>
  82:  </send>
  83:
  84:  <!-- Play a pre-recorded PCAP file (RTP stream)                       -->
  85:  <nop>
  86:    <action>
  87:      <exec play_pcap_audio="pcap/g711a.pcap"/>
  88:    </action>
  89:  </nop>
  90:
  91:  <!-- Pause 8 seconds, which is approximately the duration of the      -->
  92:  <!-- PCAP file                                                        -->
  93:  <pause milliseconds="8000"/>
  94:
  95:  <!-- Play an out of band DTMF '1'                                     -->
  96:  <nop>
  97:    <action>
  98:      <exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/>
  99:    </action>
 100:  </nop>
 101:
 102:  <pause milliseconds="1000"/>
 103:
 104:  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
 105:  <send retrans="500">
 106:    <![CDATA[
 107:
 108:      BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
 109:      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 110:      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
 111:      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
 112:      Call-ID: [call_id]
 113:      CSeq: 2 BYE
 114:      Contact: sip:sipp@[local_ip]:[local_port]
 115:      Max-Forwards: 70
 116:      Subject: Performance Test
 117:      Content-Length: 0
 118:
 119:    ]]>
 120:  </send>
 121:
 122:  <recv response="200" crlf="true">
 123:  </recv>
 124:
 125:  <!-- definition of the response time repartition table (unit is ms)   -->
 126:  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
 127:
 128:  <!-- definition of the call length repartition table (unit is ms)     -->
 129:  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
 130:
 131:</scenario>
 132: