SIPp Test Scenario

From Open CSTA

<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.                
-->
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<send retrans="500">


      INVITE 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]SIPpTag09[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Contact: sip:sipp@[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         
-->
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<!--
 are saved and used for following messages sent. Useful to test   
-->
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<!--
 against stateful SIP proxies/B2BUAs.                             
-->
<recv response="200" rtd="true" crlf="true">
  </recv>
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<!--
 Packet lost can be simulated in any send/recv message by         
-->
−
<!--
 by adding the 'lost = "10"'. Value can be [1-100] percent.       
-->
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<send>


      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]SIPpTag09[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>
−
<!--
 Play a pre-recorded PCAP file (RTP stream)                       
-->
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<nop>
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<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
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<!--
 Pause 8 seconds, which is approximately the duration of the      
-->
−
<!--
 PCAP file                                                        
-->
<pause milliseconds="8000"/>
−
<!--
 The 'crlf' option inserts a blank line in the statistics report. 
-->
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<send retrans="500">


      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]SIPpTag09[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"/>
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<!--
 definition of the call length repartition table (unit is ms)     
-->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>