uas-timer-reinvite.xml 3.3 KB

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  1. <?xml version="1.0" encoding="ISO-8859-1" ?>
  2. <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3. <!-- This program is free software; you can redistribute it and/or -->
  4. <!-- modify it under the terms of the GNU General Public License as -->
  5. <!-- published by the Free Software Foundation; either version 2 of the -->
  6. <!-- License, or (at your option) any later version. -->
  7. <!-- -->
  8. <!-- This program is distributed in the hope that it will be useful, -->
  9. <!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
  10. <!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
  11. <!-- GNU General Public License for more details. -->
  12. <!-- -->
  13. <!-- You should have received a copy of the GNU General Public License -->
  14. <!-- along with this program; if not, write to the -->
  15. <!-- Free Software Foundation, Inc., -->
  16. <!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
  17. <!-- -->
  18. <!-- Session timer where UAS doesn't indicate support for UPDATE. -->
  19. <!-- In this case, UAC MUST use re-INVITE with SDP. -->
  20. <scenario name="Basic UAS responder">
  21. <recv request="INVITE" crlf="true">
  22. </recv>
  23. <send retrans="500">
  24. <![CDATA[
  25. SIP/2.0 200 OK
  26. [last_Via:]
  27. [last_From:]
  28. [last_To:];tag=[call_number]
  29. [last_Call-ID:]
  30. [last_CSeq:]
  31. Contact: <sip:[local_ip]:[local_port];transport=[transport]>
  32. Require: timer
  33. Session-Expires: 90;refresher=uac
  34. Content-Type: application/sdp
  35. Content-Length: [len]
  36. v=0
  37. o=Some-UserAgent 68 210 IN IP4 [local_ip]
  38. s=SIP Call
  39. c=IN IP4 [local_ip]
  40. t=0 0
  41. m=audio 17294 RTP/AVP 0 101
  42. c=IN IP4 [local_ip]
  43. a=rtpmap:101 telephone-event/8000
  44. a=fmtp:101 0-16
  45. ]]>
  46. </send>
  47. <recv request="ACK"
  48. optional="true"
  49. rtd="true"
  50. crlf="true">
  51. </recv>
  52. <recv request="INVITE" crlf="true">
  53. </recv>
  54. <send retrans="500">
  55. <![CDATA[
  56. SIP/2.0 200 OK
  57. [last_Via:]
  58. [last_From:]
  59. [last_To:];tag=[call_number]
  60. [last_Call-ID:]
  61. [last_CSeq:]
  62. Contact: <sip:[local_ip]:[local_port];transport=[transport]>
  63. Require: timer
  64. Session-Expires: 90;refresher=uac
  65. Content-Type: application/sdp
  66. Content-Length: [len]
  67. v=0
  68. o=Some-UserAgent 68 210 IN IP4 [local_ip]
  69. s=SIP Call
  70. c=IN IP4 [local_ip]
  71. t=0 0
  72. m=audio 17294 RTP/AVP 0 101
  73. c=IN IP4 [local_ip]
  74. a=rtpmap:101 telephone-event/8000
  75. a=fmtp:101 0-16
  76. ]]>
  77. </send>
  78. <recv request="ACK"
  79. rtd="true"
  80. crlf="true">
  81. </recv>
  82. <!-- Keep the call open for a while in case the 200 is lost to be -->
  83. <!-- able to retransmit it if we receive the BYE again. -->
  84. <pause milliseconds="4000"/>
  85. <!-- definition of the response time repartition table (unit is ms) -->
  86. <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
  87. <!-- definition of the call length repartition table (unit is ms) -->
  88. <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
  89. </scenario>