invite-without-auth-expected-to-receive-busy-486.xml

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

<!-- You will need to compile SIPp with OpenSSL support 'make ossl' to use this call scenario -->

<!-- Execute this script with SIPp using the following command assuming your UAS is 10.0.0.10 -->
<!-- Replace 10.0.0.10 with your SIP proxy’s address. The command will generate 10 calls (-r) per 10000 -->
<!-- milliseconds (-rp), max 100 concurrent calls (-l) and make a max of 100000 calls (-m) -->

<!-- ./sipp 10.0.0.10 -sf invite-auth-sdp-nomedia.xml -inf user-accounts.csv -m 100000 -l 100 -r 10 -rp 10000 -->

<scenario name="UAC INVITE with Auth and SDP">
  <send retrans="500" start_txn="invite">
    <![CDATA[

      INVITE sip:[field3]@[field1]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid]SIPpTag00[call_number]
      To: sut <sip:[field3]@[field1]:[remote_port]>
      Call-ID: [call_id]
      CSeq: [cseq] INVITE
      Contact: <sip:sipp@[local_ip]:[local_port]>
      Max-Forwards: 70
      Subject: Performance Test
      User-Agent: SIPp Tester UAC
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[media_ip_type] [media_ip]
      t=0 0
      m=audio [media_port] RTP/AVP 8
      a=rtpmap:8 PCMA/8000
      a=rtpmap:101 telephone-event/8000
      a=fmtp:101 0-11,16


    ]]>
  </send>

  <recv response="100" optional="true" rrs="true" response_txn="invite">
  </recv>

  <recv response="486" rrs="true" response_txn="invite">
  </recv>

  <send ack_txn="invite">
    <![CDATA[

      ACK sip:[field3]@[field1] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
      From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid]SIPpTag00[call_number]
      [last_To]
      [routes]
      Call-ID: [call_id]
      CSeq: [cseq] ACK
      Contact: <sip:sipp@[local_ip]:[local_port]>
      Max-Forwards: 70
      Subject: Performance Test
      User-Agent: SIPp Tester UAC
      Content-Length: 0

    ]]>
  </send>

  <!-- 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>