From fuad, 1 Year ago, written in XML.
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Viewing differences between and multi-reinvite.xml
  1. <?xml version="1.0" encoding="ISO-8859-1" ?>
  2. <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3.  
  4. <!-- UAS with multiple reinvite      -->
  5.  
  6. <scenario name="Basic UAS responder">
  7.  
  8.   <recv request="INVITE" crlf="true">
  9.     <action>
  10.       <ereg regexp=".*" search_in="hdr" header="From" assign_to="remote_from"/>
  11.       <ereg regexp=".*sip:([^;@]*)" search_in="hdr" header="Contact:" assign_to="duh,ct"/>
  12.      </action>
  13.   </recv>
  14.   <Reference variables="duh"/>
  15.  
  16.   <send>
  17.     <![CDATA[
  18.  
  19.      SIP/2.0 100 trying -- your call is important to us
  20.      [last_Via:]
  21.      [last_From:]
  22.      [last_To:];tag=[pid]SIPpTag01[call_number]
  23.      [last_Call-ID:]
  24.      [last_CSeq:]
  25.      Content-Length: 0
  26.  
  27.    ]]>
  28.   </send>
  29.  
  30.   <pause milliseconds="1590"/>
  31.  
  32.   <send>
  33.     <![CDATA[
  34.  
  35.      SIP/2.0 183 Session Progress
  36.      [last_Via:]
  37.      [last_From:]
  38.      [last_To:];tag=[pid]SIPpTag01[call_number]
  39.      [last_Call-ID:]
  40.      [last_CSeq:]
  41.      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
  42.      Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
  43.      Content-Disposition: session; handling=required
  44.      Content-Type: application/sdp
  45.      Content-Length: [len]
  46.  
  47.      v=0
  48.      o=- 833252837 32403 IN IP[local_ip_type] [local_ip]
  49.      s=-
  50.      c=IN IP[media_ip_type] [media_ip]
  51.      t=0 0
  52.      m=audio [media_port] RTP/AVP 8 101
  53.      a=rtpmap:8 PCMA/8000
  54.      a=rtpmap:101 telephone-event/8000
  55.      a=fmtp:101 0-15
  56.      a=maxptime:20
  57.      a=sendrecv
  58.  
  59.    ]]>
  60.   </send>
  61.  
  62.   <pause milliseconds="10000"/>
  63.  
  64.   <send retrans="500">
  65.     <![CDATA[
  66.  
  67.      SIP/2.0 200 OK
  68.      [last_Via:]
  69.      [last_From:]
  70.      [last_To:];tag=[pid]SIPpTag01[call_number]
  71.      [last_Call-ID:]
  72.      [last_CSeq:]
  73.      Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay,  multipart/mixed
  74.      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
  75.      Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
  76.      Supported: replaces
  77.      Content-Disposition: session; handling=required
  78.      Content-Type: application/sdp
  79.      Content-Length: [len]
  80.  
  81.      v=0
  82.      o=- 833252837 32403 IN IP[local_ip_type] [local_ip]
  83.      s=-
  84.      c=IN IP[media_ip_type] [media_ip]
  85.      t=0 0
  86.      m=audio [media_port] RTP/AVP 8 101
  87.      a=rtpmap:8 PCMA/8000
  88.      a=rtpmap:101 telephone-event/8000
  89.      a=fmtp:101 0-15
  90.      a=maxptime:20
  91.      a=sendrecv
  92.  
  93.    ]]>
  94.   </send>
  95.  
  96.   <recv request="ACK"
  97.        optional="false"
  98.        rtd="true"
  99.        crlf="true">
  100.   </recv>
  101.  
  102.   <send retrans="500">
  103.     <![CDATA[
  104.  
  105.      INVITE sip:[$ct]@[remote_ip]:[remote_port] SIP/2.0
  106.      Via: SIP/2.0/[transport] [local_ip]:[local_port]
  107.      From: [service] <sip:[service]@[local_ip]:[local_port]>;tag=[pid]SIPpTag01[call_number]
  108.      To[$remote_from]
  109.      [last_Call-ID:]
  110.      CSeq: 31993 INVITE
  111.      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
  112.      Content-Type: application/sdp
  113.      Content-Length: [len]
  114.  
  115.      v=0
  116.      o=- 833252837 32404 IN IP[local_ip_type] [local_ip]
  117.      s=-
  118.      c=IN IP[media_ip_type] [media_ip]
  119.      t=0 0
  120.      m=audio [media_port] RTP/AVP 0 101
  121.      a=rtpmap:0 PCMU/8000
  122.      a=rtpmap:101 telephone-event/8000
  123.      a=fmtp:101 0-15
  124.      a=maxptime:20
  125.      a=sendrecv
  126.  
  127.    ]]>
  128.    </send>
  129.  
  130.    <recv response="100" optional="true">
  131.    </recv>
  132.  
  133.    <recv response="200">
  134.    </recv>
  135.    
  136.    <send>
  137.     <![CDATA[
  138.     ACK sip:[$ct]@[remote_ip]:[remote_port] SIP/2.0
  139.     Via: SIP/2.0/[transport] [local_ip]:[local_port]
  140.     From: [service] <sip:[service]@[local_ip]:[local_port]>;tag=[pid]SIPpTag01[call_number]
  141.     To[$remote_from]
  142.     [last_Call-ID:]
  143.     CSeq: 31993 ACK
  144.     Contact: <sip:[local_ip]:[local_port];transport=[transport]>
  145.     Max-Forwards: 70
  146.    ]]>
  147.    </send>
  148.  
  149.    <!-- uncomment to test with 100ms pause bween reinvites -->
  150.    <!-- <pause milliseconds="100"/> -->
  151.  
  152.    <send retrans="500">
  153.     <![CDATA[
  154.  
  155.      INVITE sip:[$ct]@[remote_ip]:[remote_port] SIP/2.0
  156.      Via: SIP/2.0/[transport] [local_ip]:[local_port]
  157.      From: [service] <sip:[service]@[local_ip]:[local_port]>;tag=[pid]SIPpTag01[call_number]
  158.      To[$remote_from]
  159.      [last_Call-ID:]
  160.      CSeq: 31994 INVITE
  161.      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
  162.      Content-Type: application/sdp
  163.      Content-Length: [len]
  164.  
  165.      v=0
  166.      o=- 833252837 32405 IN IP[local_ip_type] [local_ip]
  167.      s=-
  168.      c=IN IP[media_ip_type] [media_ip]
  169.      t=0 0
  170.      m=audio [media_port] RTP/AVP 8 101
  171.      a=rtpmap:8 PCMA/8000
  172.      a=rtpmap:101 telephone-event/8000
  173.      a=fmtp:101 0-15
  174.      a=maxptime:20
  175.      a=sendrecv
  176.  
  177.    ]]>
  178.    </send>
  179.  
  180.    <recv response="100" optional="true">
  181.    </recv>
  182.  
  183.    <recv response="200">
  184.    </recv>
  185.    
  186.    <send>
  187.     <![CDATA[
  188.     ACK sip:[$ct]@[remote_ip]:[remote_port] SIP/2.0
  189.     Via: SIP/2.0/[transport] [local_ip]:[local_port]
  190.     From: [service] <sip:[service]@[local_ip]:[local_port]>;tag=[pid]SIPpTag01[call_number]
  191.     To[$remote_from]
  192.     [last_Call-ID:]
  193.     CSeq: 31994 ACK
  194.     Contact: <sip:[local_ip]:[local_port];transport=[transport]>
  195.     Max-Forwards: 70
  196.    ]]>
  197.    </send>
  198.  
  199.   <recv request="BYE">
  200.   </recv>
  201.  
  202.   <send>
  203.     <![CDATA[
  204.  
  205.      SIP/2.0 200 OK
  206.      [last_Via:]
  207.      [last_From:]
  208.      [last_To:]
  209.      [last_Call-ID:]
  210.      [last_CSeq:]
  211.      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
  212.      Content-Length: 0
  213.  
  214.    ]]>
  215.   </send>
  216.  
  217.   <!-- Keep the call open for a while in case the 200 is lost to be     -->
  218.   <!-- able to retransmit it if we receive the BYE again.               -->
  219.   <timewait milliseconds="4000"/>
  220.  
  221.  
  222.   <!-- definition of the response time repartition table (unit is ms)   -->
  223.   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
  224.  
  225.   <!-- definition of the call length repartition table (unit is ms)     -->
  226.   <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
  227.  
  228. </scenario>
  229.