uac_g711a.xml

<?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 'uac' scenario with pcap (rtp) play           --><!--                                                                    --><scenario name="UAC with media">  <!-- 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:[field1]@[remote_ip]:[remote_port] SIP/2.0      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]      From: sipp <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]      To: sut <sip:[field1]@[remote_ip]:[remote_port]>      Call-ID: [call_id]      CSeq: 1 INVITE      Contact: sip:[field0]@[local_ip]:[local_port]      Max-Forwards: 70      Subject: Performance Test      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    ]]>  </send>  <recv response="100" optional="true">  </recv>  <recv response="180" 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" crlf="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:[field1]@[remote_ip]:[remote_port] SIP/2.0      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]      From: sipp <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]      To: sut <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]      Call-ID: [call_id]      CSeq: 1 ACK      Contact: sip:[field0]@[local_ip]:[local_port]      Max-Forwards: 70      Subject: Performance Test      Content-Length: 0    ]]>  </send>  <!-- Play a pre-recorded PCAP file (RTP stream)                       -->  <nop>    <action>      <exec play_pcap_audio="g711a.pcap"/>    </action>  </nop>  <!-- Pause 90 seconds, which is approximately the duration of the      -->  <!-- PCAP file                                                        -->  <pause milliseconds="90000"/>  <!-- The 'crlf' option inserts a blank line in the statistics report. -->  <send retrans="500">    <![CDATA[      BYE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]      From: sipp <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]      To: sut <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]      Call-ID: [call_id]      CSeq: 2 BYE      Contact: sip:[field0]@[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>