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  1. jssip-3.3.11.js:26433 JsSIP:WebSocketInterface received WebSocket message +13s
  2. jssip-3.3.11.js:26433 JsSIP:Transport received text message:INVITE sip:3002@192.168.74.111:34318;transport=ws SIP/2.0
  3. Via: SIP/2.0/WS 192.168.73.13:8088;rport;branch=z9hG4bKPjb0a8cd6d-6d38-4461-b79a-3e13c09d9065;alias
  4. From: <sip:3001@asterisk>;tag=b99f9a54-2b0f-4977-b5bc-458cbc625b8c
  5. To: <sip:3002@192.168.74.111>
  6. Contact: <sip:asterisk@asterisk:5060;transport=ws>
  7. Call-ID: 2428cf77-ea2d-4d16-9174-f389cb9d5390
  8. CSeq: 22716 INVITE
  9. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
  10. Supported: 100rel, timer, replaces, norefersub
  11. Session-Expires: 1800
  12. Min-SE: 90
  13. Max-Forwards: 70
  14. User-Agent: Asterisk PBX 17.1.0
  15. Content-Type: application/sdp
  16. Content-Length: 983
  17.  
  18. v=0
  19. o=- 847222214 847222214 IN IP4 192.168.73.13
  20. s=Asterisk
  21. c=IN IP4 192.168.73.13
  22. t=0 0
  23. a=msid-semantic:WMS *
  24. a=group:BUNDLE audio-0
  25. m=audio 16264 UDP/TLS/RTP/SAVPF 0 101
  26. a=connection:new
  27. a=setup:actpass
  28. a=fingerprint:SHA-256 2E:BB:78:71:89:6A:73:8D:23:7E:64:CD:F6:32:D5:7F:4D:AA:EF:8D:97:9B:7F:A5:C8:A8:6B:89:5B:A7:18:CD
  29. a=ice-ufrag:45120ba25b6f52914df93d4822440540
  30. a=ice-pwd:437b38fe6f2a08fe05b2be2f2689a534
  31. a=candidate:H68929ee2 1 UDP 2130706431 fe80::a00:27ff:feb1:2abb 16264 typ host
  32. a=candidate:Hc0a8490d 1 UDP 2130706431 192.168.73.13 16264 typ host
  33. a=candidate:S5b52dbcd 1 UDP 1694498815 91.82.219.205 16264 typ srflx raddr 192.168.73.13 rport 16264
  34. a=rtpmap:0 PCMU/8000
  35. a=rtpmap:101 telephone-event/8000
  36. a=fmtp:101 0-16
  37. a=ptime:20
  38. a=maxptime:150
  39. a=sendrecv
  40. a=rtcp-mux
  41. a=ssrc:6337074 cname:b05a910a-cda7-47a9-b4ab-97a0b3f41f64
  42. a=msid:a7b90526-28cb-4c54-98ed-26636e2635ed 4a5403a7-59bf-47db-9e96-d445107bba41
  43. a=rtcp-fb:* transport-cc
  44. a=mid:audio-0
  45. +13s
  46. jssip-3.3.11.js:26433 JsSIP:Transport send() +7ms
  47. jssip-3.3.11.js:26433 JsSIP:Transport sending message:SIP/2.0 100 Trying
  48. Via: SIP/2.0/WS 192.168.73.13:8088;rport;branch=z9hG4bKPjb0a8cd6d-6d38-4461-b79a-3e13c09d9065;alias
  49. To: <sip:3002@192.168.74.111>
  50. From: <sip:3001@asterisk>;tag=b99f9a54-2b0f-4977-b5bc-458cbc625b8c
  51. Call-ID: 2428cf77-ea2d-4d16-9174-f389cb9d5390
  52. CSeq: 22716 INVITE
  53. Supported: timer,ice,replaces,outbound
  54. Content-Length: 0
  55.  
  56. +2ms
  57. jssip-3.3.11.js:26433 JsSIP:WebSocketInterface send() +10ms
  58. jssip-3.3.11.js:26433 JsSIP:RTCSession new +19s
  59. jssip-3.3.11.js:26433 JsSIP:RTCSession init_incoming() +1ms
  60. jssip-3.3.11.js:26433 JsSIP:Dialog new UAS dialog created with status EARLY +19s
  61. jssip-3.3.11.js:26433 JsSIP:RTCSession newRTCSession() +2ms
  62. cyber_mega_phone.js:121 session incoming null
  63. jssip-3.3.11.js:26433 JsSIP:RTCSession answer() +1ms
  64. jssip-3.3.11.js:26433 JsSIP:Dialog dialog 2428cf77-ea2d-4d16-9174-f389cb9d5390gqt53q8tmfb99f9a54-2b0f-4977-b5bc-458cbc625b8c changed to CONFIRMED state +3ms
  65. jssip-3.3.11.js:26433 JsSIP:RTCSession emit "peerconnection" +4ms
  66. cyber_mega_phone.js:197 answer undefined
  67. jssip-3.3.11.js:26433 JsSIP:Transport send() +12ms
  68. jssip-3.3.11.js:26433 JsSIP:Transport sending message:SIP/2.0 180 Ringing
  69. Via: SIP/2.0/WS 192.168.73.13:8088;rport;branch=z9hG4bKPjb0a8cd6d-6d38-4461-b79a-3e13c09d9065;alias
  70. To: <sip:3002@192.168.74.111>;tag=gqt53q8tmf
  71. From: <sip:3001@asterisk>;tag=b99f9a54-2b0f-4977-b5bc-458cbc625b8c
  72. Call-ID: 2428cf77-ea2d-4d16-9174-f389cb9d5390
  73. CSeq: 22716 INVITE
  74. Contact: <sip:3002@192.168.73.13>
  75. Supported: timer,ice,replaces,outbound
  76. Content-Length: 0
  77.  
  78. +0ms
  79. jssip-3.3.11.js:26433 JsSIP:WebSocketInterface send() +12ms
  80. jssip-3.3.11.js:26433 JsSIP:RTCSession session progress +5ms
  81. jssip-3.3.11.js:26433 JsSIP:RTCSession emit "progress" +0ms
  82. cyber_mega_phone.js:144 progress {originator: "local", response: null}
  83. jssip-3.3.11.js:26433 JsSIP:RTCSession emit "sdp" +15s
  84. jssip-3.3.11.js:26433 JsSIP:RTCSession session connecting +3ms
  85. jssip-3.3.11.js:26433 JsSIP:RTCSession emit "connecting" +0ms
  86. cyber_mega_phone.js:126 connecting {request: IncomingRequest}request: IncomingRequest {data: "INVITE sip:3002@192.168.74.111:34318;transport=ws …7bba41
  87. ↵a=rtcp-fb:* transport-cc
  88. ↵a=mid:audio-0
  89. ↵", headers: {…}, method: "INVITE", via: {…}, via_branch: "z9hG4bKPjb0a8cd6d-6d38-4461-b79a-3e13c09d9065", …}__proto__: Object
  90. cyber_mega_phone.js:127 session incoming RTCSession {_events: {…}, _eventsCount: 9, _maxListeners: undefined, _id: "2428cf77-ea2d-4d16-9174-f389cb9d5390b99f9a54-2b0f-4977-b5bc-458cbc625b8c", _ua: UA, …}C: (...)causes: (...)id: (...)connection: (...)contact: (...)direction: (...)local_identity: (...)remote_identity: (...)start_time: (...)end_time: (...)data: (...)status: (...)_events: {connecting: ƒ, sending: ƒ, progress: ƒ, accepted: ƒ, confirmed: ƒ, …}_eventsCount: 9_maxListeners: undefined_id: "2428cf77-ea2d-4d16-9174-f389cb9d5390b99f9a54-2b0f-4977-b5bc-458cbc625b8c"_ua: UA {_events: {…}, _eventsCount: 7, _maxListeners: undefined, _cache: {…}, _configuration: {…}, …}_status: 9_dialog: Dialog {_owner: RTCSession, _ua: UA, _uac_pending_reply: false, _uas_pending_reply: false, _id: {…}, …}_earlyDialogs: {}_contact: "<sip:3002@192.168.73.13>"_from_tag: "b99f9a54-2b0f-4977-b5bc-458cbc625b8c"_to_tag: "gqt53q8tmf"_connection: RTCPeerConnection {localDescription: RTCSessionDescription, currentLocalDescription: RTCSessionDescription, pendingLocalDescription: null, remoteDescription: RTCSessionDescription, currentRemoteDescription: RTCSessionDescription, …}_connectionPromiseQueue: Promise {<resolved>: undefined}_request: IncomingRequest {data: "INVITE sip:3002@192.168.74.111:34318;transport=ws …7bba41
  91. ↵a=rtcp-fb:* transport-cc
  92. ↵a=mid:audio-0
  93. ↵", headers: {…}, method: "INVITE", via: {…}, via_branch: "z9hG4bKPjb0a8cd6d-6d38-4461-b79a-3e13c09d9065", …}_is_canceled: false_cancel_reason: ""_is_confirmed: true_late_sdp: false_rtcOfferConstraints: null_rtcAnswerConstraints: null_localMediaStream: MediaStream {id: "apr7HjfQVFBHW0R1fefb6VUXkYfWSzTixszl", active: true, onaddtrack: null, onremovetrack: null, onactive: null, …}_localMediaStreamLocallyGenerated: true_rtcReady: true_timers: {ackTimer: 47, expiresTimer: null, invite2xxTimer: 46, userNoAnswerTimer: 42}_direction: "incoming"_local_identity: NameAddrHeader {_uri: URI, _parameters: {…}, _display_name: undefined}_remote_identity: NameAddrHeader {_uri: URI, _parameters: {…}, _display_name: undefined}_start_time: Wed Jan 29 2020 17:46:54 GMT+0100 (közép-európai téli idő) {}_end_time: null_tones: null_audioMuted: false_videoMuted: false_localHold: false_remoteHold: false_sessionTimers: {enabled: true, refreshMethod: "UPDATE", defaultExpires: 90, currentExpires: 1800, running: true, …}_referSubscribers: {}_data: {}__proto__: EventEmitter
  94. jssip-3.3.11.js:26433 JsSIP:RTCSession createLocalDescription() +0ms
  95. jssip-3.3.11.js:18530 connection RTCPeerConnection {localDescription: null, currentLocalDescription: null, pendingLocalDescription: null, remoteDescription: RTCSessionDescription, currentRemoteDescription: null, …}
  96. jssip-3.3.11.js:18533 create-start answer
  97. jssip-3.3.11.js:18554 local-start RTCSessionDescription {type: "answer", sdp: "v=0
  98. ↵o=- 9129688551605474471 2 IN IP4 127.0.0.1
  99. ↵s…t/8000
  100. ↵a=ssrc:765810801 cname:/3J0I8md4rw3UdBh
  101. ↵"}
  102. jssip-3.3.11.js:18565 ice new
  103. jssip-3.3.11.js:18586 ice1
  104. jssip-3.3.11.js:26433 JsSIP:RTCSession emit "sdp" +106ms
  105. jssip-3.3.11.js:26433 JsSIP:Transport send() +15s
  106. jssip-3.3.11.js:26433 JsSIP:Transport sending message:SIP/2.0 200 OK
  107. Via: SIP/2.0/WS 192.168.73.13:8088;rport;branch=z9hG4bKPjb0a8cd6d-6d38-4461-b79a-3e13c09d9065;alias
  108. To: <sip:3002@192.168.74.111>;tag=gqt53q8tmf
  109. From: <sip:3001@asterisk>;tag=b99f9a54-2b0f-4977-b5bc-458cbc625b8c
  110. Call-ID: 2428cf77-ea2d-4d16-9174-f389cb9d5390
  111. CSeq: 22716 INVITE
  112. Contact: <sip:3002@192.168.73.13>
  113. Session-Expires: 1800;refresher=uas
  114. Supported: timer,ice,replaces,outbound
  115. Content-Type: application/sdp
  116. Content-Length: 781
  117.  
  118. v=0
  119. o=- 9129688551605474471 2 IN IP4 127.0.0.1
  120. s=-
  121. t=0 0
  122. a=group:BUNDLE audio-0
  123. a=msid-semantic: WMS apr7HjfQVFBHW0R1fefb6VUXkYfWSzTixszl
  124. m=audio 9 UDP/TLS/RTP/SAVPF 0 101
  125. c=IN IP4 0.0.0.0
  126. a=rtcp:9 IN IP4 0.0.0.0
  127. a=candidate:972255516 1 udp 2113937151 b195de90-2cfa-4670-ba7f-4ae9a7772ab2.local 59581 typ host generation 0 network-cost 999
  128. a=ice-ufrag:CGz9
  129. a=ice-pwd:6mRO2Lly6QPKhxbp3V4ewtEX
  130. a=ice-options:trickle
  131. a=fingerprint:sha-256 44:DD:1F:30:B7:48:74:21:D3:3C:AC:41:82:58:0B:02:5E:08:3E:DC:74:DA:4B:4A:61:91:0F:38:27:D0:88:D9
  132. a=setup:active
  133. a=mid:audio-0
  134. a=sendrecv
  135. a=msid:apr7HjfQVFBHW0R1fefb6VUXkYfWSzTixszl 47613259-2483-4f62-86c2-fd7e4cdafdd5
  136. a=rtcp-mux
  137. a=rtpmap:0 PCMU/8000
  138. a=rtpmap:101 telephone-event/8000
  139. a=ssrc:765810801 cname:/3J0I8md4rw3UdBh
  140. +0ms
  141. jssip-3.3.11.js:26433 JsSIP:WebSocketInterface send() +15s
  142. jssip-3.3.11.js:26433 JsSIP:RTCSession session accepted +1ms
  143. jssip-3.3.11.js:26433 JsSIP:RTCSession emit "accepted" +1ms
  144. cyber_mega_phone.js:147 the call has answered {originator: "local", response: null}
  145. jssip-3.3.11.js:26433 JsSIP:WebSocketInterface received WebSocket message +15ms
  146. jssip-3.3.11.js:26433 JsSIP:Transport received text message:ACK sip:3002@192.168.74.111:34318;transport=ws SIP/2.0
  147. Via: SIP/2.0/WS 192.168.73.13:8088;rport;branch=z9hG4bKPj92be4fe4-1fe0-40c7-b634-b211d0aa1602;alias
  148. From: <sip:3001@asterisk>;tag=b99f9a54-2b0f-4977-b5bc-458cbc625b8c
  149. To: <sip:3002@192.168.74.111>;tag=gqt53q8tmf
  150. Call-ID: 2428cf77-ea2d-4d16-9174-f389cb9d5390
  151. CSeq: 22716 ACK
  152. Max-Forwards: 70
  153. User-Agent: Asterisk PBX 17.1.0
  154. Content-Length: 0
  155.  
  156. +15ms
  157. jssip-3.3.11.js:26433 JsSIP:RTCSession receiveRequest() +15ms
  158. jssip-3.3.11.js:26433 JsSIP:RTCSession session confirmed +1ms
  159. jssip-3.3.11.js:26433 JsSIP:RTCSession emit "confirmed" +0ms
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