mrcp_uac_recog 3.2 KB

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  1. <?xml version="1.0" encoding="ISO-8859-1" ?>
  2. <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3. <scenario name="MRCP Recognizer Resource UAC">
  4. <!-- In client mode (sipp placing calls), the Call-ID MUST be -->
  5. <!-- generated by sipp. To do so, use [call_id] token. -->
  6. <send retrans="500">
  7. <![CDATA[
  8. INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  9. Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  10. From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
  11. To: sut <sip:[service]@[remote_ip]:[remote_port]>
  12. Call-ID: [call_id]
  13. CSeq: 1 INVITE
  14. Contact: sip:sipp@[local_ip]:[local_port]
  15. Max-Forwards: 70
  16. Subject: Performance Test
  17. Content-Type: application/sdp
  18. Content-Length: [len]
  19. v=0
  20. o=user1 53655765 2353687637 IN IP4 [local_ip]
  21. s=-
  22. c=IN IP4 [media_ip]
  23. t=0 0
  24. m=application 9 TCP/MRCPv2 1
  25. a=setup:active
  26. a=connection:new
  27. a=resource:speechrecog
  28. a=cmid:1
  29. m=audio [media_port] RTP/AVP 0 8
  30. a=sendonly
  31. a=mid:1
  32. ]]>
  33. </send>
  34. <recv response="100"
  35. optional="true">
  36. </recv>
  37. <recv response="180" optional="true">
  38. </recv>
  39. <!-- By adding rrs="true" (Record Route Sets), the route sets -->
  40. <!-- are saved and used for following messages sent. Useful to test -->
  41. <!-- against stateful SIP proxies/B2BUAs. -->
  42. <recv response="200" rtd="true">
  43. </recv>
  44. <!-- Packet lost can be simulated in any send/recv message by -->
  45. <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
  46. <send>
  47. <![CDATA[
  48. ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  49. Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  50. From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
  51. To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
  52. Call-ID: [call_id]
  53. CSeq: 1 ACK
  54. Contact: sip:sipp@[local_ip]:[local_port]
  55. Max-Forwards: 70
  56. Subject: Performance Test
  57. Content-Length: 0
  58. ]]>
  59. </send>
  60. <!-- This delay can be customized by the -d command-line option -->
  61. <!-- or by adding a 'milliseconds = "value"' option here. -->
  62. <pause/>
  63. <!-- The 'crlf' option inserts a blank line in the statistics report. -->
  64. <send retrans="500">
  65. <![CDATA[
  66. BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  67. Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  68. From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
  69. To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
  70. Call-ID: [call_id]
  71. CSeq: 2 BYE
  72. Contact: sip:sipp@[local_ip]:[local_port]
  73. Max-Forwards: 70
  74. Subject: Performance Test
  75. Content-Length: 0
  76. ]]>
  77. </send>
  78. <recv response="200" crlf="true">
  79. </recv>
  80. <!-- definition of the response time repartition table (unit is ms) -->
  81. <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
  82. <!-- definition of the call length repartition table (unit is ms) -->
  83. <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
  84. </scenario>