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SIP DEBUGGING CALLN 200 EXT

powerponch
Oct 2nd, 2015
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  1. ***** http.conf *****
  2. [general]
  3. enabled=yes
  4. bindaddr=0.0.0.0
  5. bindport=8088
  6.  
  7.  
  8. **** rtp.conf *****
  9. [general]
  10. rtpstart=19000
  11. rtpend=21000
  12. icesupport=true
  13. stunaddr=stun.l.google.com:19302
  14.  
  15. ***** sip.conf *****
  16. [general]
  17. udpbindaddr=0.0.0.0:5060
  18. rtcachefriends=yes
  19. limitonpeers=yes
  20. callcounter=yes
  21. allowoverlap=no ; Disable overlap dialing support. (Default is yes)
  22. localnet=127.0.0.1/255.255.255.0
  23. externip=192.168.0.9
  24.  
  25. [1000] ;zoiper UA
  26. type=friend
  27. secret=rendezvous080193
  28. qualify=yes
  29. host=dynamic
  30. dtmfmode=rfc2833
  31. context=miContexto
  32. directmedia=no
  33. callerid=Fulano <1000>
  34. disallow=all
  35. allow=gsm
  36. allow=ulaw ; Allow codecs in order of preference
  37. allow=alaw
  38.  
  39. [8000] ;mozilla UA
  40. type=friend
  41. secret=rendezvous080193
  42. context=miContexto
  43. host=dynamic
  44. qualify=yes
  45. callerid=WebRTC <8000>
  46. transport=udp,ws
  47. avpf=yes
  48. force_avp=yes
  49. encryption=yes
  50. icesupport=yes
  51. directmedia=no
  52. dtlsenable=yes
  53. dtlsverify=no
  54. disallow=all
  55. allow=all
  56. dtmf=auto
  57. ;nat=force_rport,comedia
  58. nat=no
  59. dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
  60. dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
  61. dtlscertfile=/etc/asterisk/keys/asterisk.pem
  62. dtlsprivatekey=/etc/asterisk/keys/asterisk.pem
  63. dtlssetup=actpass
  64.  
  65.  
  66. ***** extensions.conf *****
  67.  
  68. [miContexto]
  69. exten=> 200,1,Answer()
  70. same=> n,Playback(hello-world)
  71. same => n,Hangup()
  72.  
  73. exten=> 1000,1,Dial(SIP/1000)
  74. exten=> 8000,1,Dial(SIP/8000)
  75.  
  76. exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
  77. exten => 600,2,Echo ; Do the echo test
  78. exten => 600,3,Playback(demo-echodone) ; Let them know it's over
  79. *******************************************************************************************************
  80. ASTERISK SAYS THIS (having RTP debug enabled):
  81. == WebSocket connection from '127.0.0.1:35106' for protocol 'sip' accepted using version '13'
  82. == WebSocket connection from '127.0.0.1:35096' closed
  83. -- Registered SIP '8000' at 127.0.0.1:35106
  84. == Using SIP RTP CoS mark 5
  85. -- Executing [200@miContexto:1] Answer("SIP/8000-00000000", "") in new stack
  86. == Spawn extension (miContexto, 200, 1) exited non-zero on 'SIP/8000-00000000'
  87.  
  88. *****************************************************************************************************
  89.  
  90.  
  91. "Iniciando creación del UA SIP...." miScript.js:19:4
  92. "JsSIP:UA " "configuration parameters after validation:" " +0ms" jssip-0.7.4.js:22459:7
  93. "JsSIP:UA " "- via_host: "192.0.2.31"" " +2ms" jssip-0.7.4.js:22459:7
  94. "JsSIP:UA " "- password: NOT SHOWN" " +2ms" jssip-0.7.4.js:22459:7
  95. "JsSIP:UA " "- register_expires: 600" " +1ms" jssip-0.7.4.js:22459:7
  96. "JsSIP:UA " "- register: true" " +2ms" jssip-0.7.4.js:22459:7
  97. "JsSIP:UA " "- registrar_server: sip:127.0.0.1" " +1ms" jssip-0.7.4.js:22459:7
  98. "JsSIP:UA " "- ws_server_max_reconnection: 3" " +1ms" jssip-0.7.4.js:22459:7
  99. "JsSIP:UA " "- ws_server_reconnection_timeout: 4" " +2ms" jssip-0.7.4.js:22459:7
  100. "JsSIP:UA " "- connection_recovery_min_interval: 2" " +2ms" jssip-0.7.4.js:22459:7
  101. "JsSIP:UA " "- connection_recovery_max_interval: 30" " +1ms" jssip-0.7.4.js:22459:7
  102. "JsSIP:UA " "- use_preloaded_route: false" " +2ms" jssip-0.7.4.js:22459:7
  103. "JsSIP:UA " "- no_answer_timeout: 60000" " +2ms" jssip-0.7.4.js:22459:7
  104. "JsSIP:UA " "- session_timers: true" " +1ms" jssip-0.7.4.js:22459:7
  105. "JsSIP:UA " "- hack_via_tcp: false" " +1ms" jssip-0.7.4.js:22459:7
  106. "JsSIP:UA " "- hack_via_ws: false" " +1ms" jssip-0.7.4.js:22459:7
  107. "JsSIP:UA " "- hack_ip_in_contact: true" " +1ms" jssip-0.7.4.js:22459:7
  108. "JsSIP:UA " "- node_websocket_options: {}" " +1ms" jssip-0.7.4.js:22459:7
  109. "JsSIP:UA " "- uri: sip:8000@127.0.0.1" " +2ms" jssip-0.7.4.js:22459:7
  110. "JsSIP:UA " "- ws_servers: [{"ws_uri":"ws://127.0.0.1:8088/ws","sip_uri":"<sip:127.0.0.1:8088;transport=ws;lr>","weight":0,"status":0,"scheme":"WS"}]" " +1ms" jssip-0.7.4.js:22459:7
  111. "JsSIP:UA " "- display_name: "UA WebRTC"" " +1ms" jssip-0.7.4.js:22459:7
  112. "JsSIP:UA " "- instance_id: "47e7b0b8-ab7e-49f1-9189-3ff5379132c9"" " +1ms" jssip-0.7.4.js:22459:7
  113. "JsSIP:UA " "- jssip_id: "3n0vm"" " +1ms" jssip-0.7.4.js:22459:7
  114. "JsSIP:UA " "- hostport_params: "127.0.0.1"" " +1ms" jssip-0.7.4.js:22459:7
  115. "JsSIP:UA " "- authorization_user: "8000"" " +1ms" jssip-0.7.4.js:22459:7
  116. "JsSIP:UA " "start()" " +2ms" jssip-0.7.4.js:22459:7
  117. "JsSIP:Transport " "connecting to WebSocket ws://127.0.0.1:8088/ws" " +2ms" jssip-0.7.4.js:22459:7
  118. "JsSIP:Transport " "WebSocket ws://127.0.0.1:8088/ws connected" " +250ms" jssip-0.7.4.js:22459:7
  119. "JsSIP:Transport " "sending WebSocket message:
  120.  
  121. REGISTER sip:127.0.0.1 SIP/2.0
  122.  
  123. Via: SIP/2.0/WS 192.0.2.31;branch=z9hG4bK9265330
  124.  
  125. Max-Forwards: 69
  126.  
  127. To: <sip:8000@127.0.0.1>
  128.  
  129. From: "UA WebRTC" <sip:8000@127.0.0.1>;tag=ro13hp8deg
  130.  
  131. Call-ID: 55b6vusi0pu0jcpvbp124u
  132.  
  133. CSeq: 1 REGISTER
  134.  
  135. Contact: <sip:tkiipe6h@192.0.2.31;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:47e7b0b8-ab7e-49f1-9189-3ff5379132c9>";expires=600
  136.  
  137. Expires: 600
  138.  
  139. Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER
  140.  
  141. Supported: path,gruu,outbound
  142.  
  143. User-Agent: JsSIP 0.7.4
  144.  
  145. Content-Length: 0
  146.  
  147.  
  148.  
  149.  
  150. " " +9ms" jssip-0.7.4.js:22459:7
  151. "JsSIP:Transport " "received WebSocket text message:
  152.  
  153. SIP/2.0 401 Unauthorized
  154.  
  155. Via: SIP/2.0/WS 192.0.2.31;branch=z9hG4bK9265330;received=127.0.0.1
  156.  
  157. From: "UA WebRTC" <sip:8000@127.0.0.1>;tag=ro13hp8deg
  158.  
  159. To: <sip:8000@127.0.0.1>;tag=as2783d8f8
  160.  
  161. Call-ID: 55b6vusi0pu0jcpvbp124u
  162.  
  163. CSeq: 1 REGISTER
  164.  
  165. Server: Asterisk PBX 11.19.0
  166.  
  167. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  168.  
  169. Supported: replaces, timer
  170.  
  171. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5a08ff30"
  172.  
  173. Content-Length: 0
  174.  
  175.  
  176.  
  177.  
  178. " " +11ms" jssip-0.7.4.js:22459:7
  179. "JsSIP:Transport " "sending WebSocket message:
  180.  
  181. REGISTER sip:127.0.0.1 SIP/2.0
  182.  
  183. Via: SIP/2.0/WS 192.0.2.31;branch=z9hG4bK7539545
  184.  
  185. Max-Forwards: 69
  186.  
  187. To: <sip:8000@127.0.0.1>
  188.  
  189. From: "UA WebRTC" <sip:8000@127.0.0.1>;tag=ro13hp8deg
  190.  
  191. Call-ID: 55b6vusi0pu0jcpvbp124u
  192.  
  193. CSeq: 2 REGISTER
  194.  
  195. Authorization: Digest algorithm=MD5, username="8000", realm="asterisk", nonce="5a08ff30", uri="sip:127.0.0.1", response="9cadd75a83cbb02e36b45998ad1eb409"
  196.  
  197. Contact: <sip:tkiipe6h@192.0.2.31;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:47e7b0b8-ab7e-49f1-9189-3ff5379132c9>";expires=600
  198.  
  199. Expires: 600
  200.  
  201. Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER
  202.  
  203. Supported: path,gruu,outbound
  204.  
  205. User-Agent: JsSIP 0.7.4
  206.  
  207. Content-Length: 0
  208.  
  209.  
  210.  
  211.  
  212. " " +26ms" jssip-0.7.4.js:22459:7
  213. "JsSIP:Transport " "received WebSocket text message:
  214.  
  215. OPTIONS sip:tkiipe6h@192.0.2.31;transport=ws SIP/2.0
  216.  
  217. Via: SIP/2.0/WS 127.0.0.1:5060;branch=z9hG4bK6963fc33
  218.  
  219. Max-Forwards: 70
  220.  
  221. From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as6ba0e1ac
  222.  
  223. To: <sip:tkiipe6h@192.0.2.31;transport=ws>
  224.  
  225. Contact: <sip:asterisk@127.0.0.1:5060;transport=WS>
  226.  
  227. Call-ID: 2025288564340a066f4969894ceceb99@127.0.0.1:5060
  228.  
  229. CSeq: 102 OPTIONS
  230.  
  231. User-Agent: Asterisk PBX 11.19.0
  232.  
  233. Date: Fri, 02 Oct 2015 14:12:00 GMT
  234.  
  235. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  236.  
  237. Supported: replaces, timer
  238.  
  239. Content-Length: 0
  240.  
  241.  
  242.  
  243.  
  244. " " +70ms" jssip-0.7.4.js:22459:7
  245. "JsSIP:Transport " "sending WebSocket message:
  246.  
  247. SIP/2.0 200 OK
  248.  
  249. Via: SIP/2.0/WS 127.0.0.1:5060;branch=z9hG4bK6963fc33
  250.  
  251. To: <sip:tkiipe6h@192.0.2.31;transport=ws>;tag=foaelvfqod
  252.  
  253. From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as6ba0e1ac
  254.  
  255. Call-ID: 2025288564340a066f4969894ceceb99@127.0.0.1:5060
  256.  
  257. CSeq: 102 OPTIONS
  258.  
  259. Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER
  260.  
  261. Accept: application/sdp, application/dtmf-relay
  262.  
  263. Supported: outbound
  264.  
  265. Content-Length: 0
  266.  
  267.  
  268.  
  269.  
  270. " " +13ms" jssip-0.7.4.js:22459:7
  271. "JsSIP:Transport " "received WebSocket text message:
  272.  
  273. SIP/2.0 200 OK
  274.  
  275. Via: SIP/2.0/WS 192.0.2.31;branch=z9hG4bK7539545;received=127.0.0.1
  276.  
  277. From: "UA WebRTC" <sip:8000@127.0.0.1>;tag=ro13hp8deg
  278.  
  279. To: <sip:8000@127.0.0.1>;tag=as2783d8f8
  280.  
  281. Call-ID: 55b6vusi0pu0jcpvbp124u
  282.  
  283. CSeq: 2 REGISTER
  284.  
  285. Server: Asterisk PBX 11.19.0
  286.  
  287. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  288.  
  289. Supported: replaces, timer
  290.  
  291. Expires: 600
  292.  
  293. Contact: <sip:tkiipe6h@192.0.2.31;transport=ws>;expires=600
  294.  
  295. Date: Fri, 02 Oct 2015 14:12:00 GMT
  296.  
  297. Content-Length: 0
  298.  
  299.  
  300.  
  301.  
  302. " " +2ms" jssip-0.7.4.js:22459:7
  303. "JsSIP:NonInviteServerTransaction " "Timer J expired for transaction z9hG4bK6963fc33" " +53ms" jssip-0.7.4.js:22459:7
  304. "Intentando hacer una llamada ...." miScript.js:50:4
  305. "200" miScript.js:53:4
  306. "JsSIP:UA " "call()" " +2s" jssip-0.7.4.js:22459:7
  307. "JsSIP:RTCSession " "new" " +1ms" jssip-0.7.4.js:22459:7
  308. "JsSIP:RTCSession " "connect()" " +2ms" jssip-0.7.4.js:22459:7
  309. "JsSIP:RTCSession " "newRTCSession" " +481ms" jssip-0.7.4.js:22459:7
  310. "Marcando ...." miScript.js:101:4
  311. "JsSIP:RTCSession " "session connecting" " +2s" jssip-0.7.4.js:22459:7
  312. "JsSIP:RTCSession " "createLocalDescription()" " +2ms" jssip-0.7.4.js:22459:7
  313. "JsSIP:Transport " "sending WebSocket message:
  314.  
  315.  
  316.  
  317. ************************************************** trying to call extension 200 (hello world) ****************
  318. INVITE sip:200@127.0.0.1 SIP/2.0
  319.  
  320. Via: SIP/2.0/WS 192.0.2.31;branch=z9hG4bK145074
  321.  
  322. Max-Forwards: 69
  323.  
  324. To: <sip:200@127.0.0.1>
  325.  
  326. From: "UA WebRTC" <sip:8000@127.0.0.1>;tag=75qkalul6h
  327.  
  328. Call-ID: 3n0vmgthqm00ie7fckd7
  329.  
  330. CSeq: 9400 INVITE
  331.  
  332. Contact: <sip:tkiipe6h@192.0.2.31;transport=ws;ob>
  333.  
  334. Content-Type: application/sdp
  335.  
  336. Session-Expires: 90
  337.  
  338. Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER
  339.  
  340. Supported: timer,ice,replaces,outbound
  341.  
  342. User-Agent: JsSIP 0.7.4
  343.  
  344. Content-Length: 905
  345.  
  346.  
  347.  
  348. v=0
  349.  
  350. o=mozilla...THIS_IS_SDPARTA-38.2.1 4294967295 0 IN IP4 0.0.0.0
  351.  
  352. s=-
  353.  
  354. t=0 0
  355.  
  356. a=sendrecv
  357.  
  358. a=fingerprint:sha-256 7B:32:CA:D4:51:63:31:1F:85:46:9E:B0:41:EA:E3:2E:05:F5:0D:35:88:7A:59:AC:93:66:AB:CA:37:0E:BA:11
  359.  
  360. a=group:BUNDLE sdparta_0
  361.  
  362. a=ice-options:trickle
  363.  
  364. a=msid-semantic:WMS *
  365.  
  366. m=audio 49533 RTP/SAVPF 109 9 0 8
  367.  
  368. c=IN IP4 192.168.0.9
  369.  
  370. a=candidate:0 1 UDP 2122252543 192.168.0.9 49533 typ host
  371.  
  372. a=candidate:0 2 UDP 2122252542 192.168.0.9 54679 typ host
  373.  
  374. a=sendrecv
  375.  
  376. a=end-of-candidates
  377.  
  378. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  379.  
  380. a=ice-pwd:375317dda2bcb894ad2577a4cd9ddbee
  381.  
  382. a=ice-ufrag:9edebb48
  383.  
  384. a=mid:sdparta_0
  385.  
  386. a=msid:{adec7611-1e43-4955-b4c9-38d4f4f1ced7} {378e126d-aedb-4d91-a385-813c2c073e5e}
  387.  
  388. a=rtcp-mux
  389.  
  390. a=rtpmap:109 opus/48000/2
  391.  
  392. a=rtpmap:9 G722/8000/1
  393.  
  394. a=rtpmap:0 PCMU/8000
  395.  
  396. a=rtpmap:8 PCMA/8000
  397.  
  398. a=setup:actpass
  399.  
  400. a=ssrc:2458917309 cname:{74bc0d3f-61c6-4f18-aef5-2b8fccd90c04}
  401.  
  402.  
  403. " " +73ms" jssip-0.7.4.js:22459:7
  404. "JsSIP:Transport " "received WebSocket text message:
  405.  
  406. SIP/2.0 401 Unauthorized
  407.  
  408. Via: SIP/2.0/WS 192.0.2.31;branch=z9hG4bK145074;received=127.0.0.1
  409.  
  410. From: "UA WebRTC" <sip:8000@127.0.0.1>;tag=75qkalul6h
  411.  
  412. To: <sip:200@127.0.0.1>;tag=as6308f70c
  413.  
  414. Call-ID: 3n0vmgthqm00ie7fckd7
  415.  
  416. CSeq: 9400 INVITE
  417.  
  418. Server: Asterisk PBX 11.19.0
  419.  
  420. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  421.  
  422. Supported: replaces, timer
  423.  
  424. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="06d37bcf"
  425.  
  426. Content-Length: 0
  427.  
  428.  
  429.  
  430.  
  431. " " +69ms" jssip-0.7.4.js:22459:7
  432. "JsSIP:Transport " "sending WebSocket message:
  433.  
  434. ACK sip:200@127.0.0.1 SIP/2.0
  435.  
  436. Via: SIP/2.0/WS 192.0.2.31;branch=z9hG4bK145074
  437.  
  438. To: <sip:200@127.0.0.1>;tag=as6308f70c
  439.  
  440. From: "UA WebRTC" <sip:8000@127.0.0.1>;tag=75qkalul6h
  441.  
  442. Call-ID: 3n0vmgthqm00ie7fckd7
  443.  
  444. CSeq: 9400 ACK
  445.  
  446. Content-Length: 0
  447.  
  448.  
  449.  
  450.  
  451. " " +9ms" jssip-0.7.4.js:22459:7
  452. "JsSIP:Transport " "sending WebSocket message:
  453.  
  454. INVITE sip:200@127.0.0.1 SIP/2.0
  455.  
  456. Via: SIP/2.0/WS 192.0.2.31;branch=z9hG4bK9928882
  457.  
  458. Max-Forwards: 69
  459.  
  460. To: <sip:200@127.0.0.1>
  461.  
  462. From: "UA WebRTC" <sip:8000@127.0.0.1>;tag=75qkalul6h
  463.  
  464. Call-ID: 3n0vmgthqm00ie7fckd7
  465.  
  466. CSeq: 9401 INVITE
  467.  
  468. Authorization: Digest algorithm=MD5, username="8000", realm="asterisk", nonce="06d37bcf", uri="sip:200@127.0.0.1", response="68c5da6731853dc9a695a00f26fd3e49"
  469.  
  470. Contact: <sip:tkiipe6h@192.0.2.31;transport=ws;ob>
  471.  
  472. Content-Type: application/sdp
  473.  
  474. Session-Expires: 90
  475.  
  476. Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER
  477.  
  478. Supported: timer,ice,replaces,outbound
  479.  
  480. User-Agent: JsSIP 0.7.4
  481.  
  482. Content-Length: 905
  483.  
  484.  
  485.  
  486. v=0
  487.  
  488. o=mozilla...THIS_IS_SDPARTA-38.2.1 4294967295 0 IN IP4 0.0.0.0
  489.  
  490. s=-
  491.  
  492. t=0 0
  493.  
  494. a=sendrecv
  495.  
  496. a=fingerprint:sha-256 7B:32:CA:D4:51:63:31:1F:85:46:9E:B0:41:EA:E3:2E:05:F5:0D:35:88:7A:59:AC:93:66:AB:CA:37:0E:BA:11
  497.  
  498. a=group:BUNDLE sdparta_0
  499.  
  500. a=ice-options:trickle
  501.  
  502. a=msid-semantic:WMS *
  503.  
  504. m=audio 49533 RTP/SAVPF 109 9 0 8
  505.  
  506. c=IN IP4 192.168.0.9
  507.  
  508. a=candidate:0 1 UDP 2122252543 192.168.0.9 49533 typ host
  509.  
  510. a=candidate:0 2 UDP 2122252542 192.168.0.9 54679 typ host
  511.  
  512. a=sendrecv
  513.  
  514. a=end-of-candidates
  515.  
  516. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  517.  
  518. a=ice-pwd:375317dda2bcb894ad2577a4cd9ddbee
  519.  
  520. a=ice-ufrag:9edebb48
  521.  
  522. a=mid:sdparta_0
  523.  
  524. a=msid:{adec7611-1e43-4955-b4c9-38d4f4f1ced7} {378e126d-aedb-4d91-a385-813c2c073e5e}
  525.  
  526. a=rtcp-mux
  527.  
  528. a=rtpmap:109 opus/48000/2
  529.  
  530. a=rtpmap:9 G722/8000/1
  531.  
  532. a=rtpmap:0 PCMU/8000
  533.  
  534. a=rtpmap:8 PCMA/8000
  535.  
  536. a=setup:actpass
  537.  
  538. a=ssrc:2458917309 cname:{74bc0d3f-61c6-4f18-aef5-2b8fccd90c04}
  539.  
  540.  
  541. " " +4ms" jssip-0.7.4.js:22459:7
  542. "JsSIP:InviteClientTransaction " "Timer D expired for transaction z9hG4bK145074" " +3ms" jssip-0.7.4.js:22459:7
  543. "JsSIP:Transport " "received WebSocket text message:
  544.  
  545. SIP/2.0 100 Trying
  546.  
  547. Via: SIP/2.0/WS 192.0.2.31;branch=z9hG4bK9928882;received=127.0.0.1
  548.  
  549. From: "UA WebRTC" <sip:8000@127.0.0.1>;tag=75qkalul6h
  550.  
  551. To: <sip:200@127.0.0.1>
  552.  
  553. Call-ID: 3n0vmgthqm00ie7fckd7
  554.  
  555. CSeq: 9401 INVITE
  556.  
  557. Server: Asterisk PBX 11.19.0
  558.  
  559. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  560.  
  561. Supported: replaces, timer
  562.  
  563. Session-Expires: 90;refresher=uas
  564.  
  565. Contact: <sip:200@127.0.0.1:5060;transport=WS>
  566.  
  567. Content-Length: 0
  568.  
  569.  
  570.  
  571.  
  572. " " +219ms" jssip-0.7.4.js:22459:7
  573. "JsSIP:RTCSession " "receiveInviteResponse()" " +9ms" jssip-0.7.4.js:22459:7
  574. "JsSIP:Transport " "received WebSocket text message:
  575.  
  576. SIP/2.0 200 OK
  577.  
  578. Via: SIP/2.0/WS 192.0.2.31;branch=z9hG4bK9928882;received=127.0.0.1
  579.  
  580. From: "UA WebRTC" <sip:8000@127.0.0.1>;tag=75qkalul6h
  581.  
  582. To: <sip:200@127.0.0.1>;tag=as6ea09826
  583.  
  584. Call-ID: 3n0vmgthqm00ie7fckd7
  585.  
  586. CSeq: 9401 INVITE
  587.  
  588. Server: Asterisk PBX 11.19.0
  589.  
  590. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  591.  
  592. Supported: replaces, timer
  593.  
  594. Session-Expires: 90;refresher=uas
  595.  
  596. Contact: <sip:200@127.0.0.1:5060;transport=WS>
  597.  
  598. Content-Type: application/sdp
  599.  
  600. Require: timer
  601.  
  602. Content-Length: 373
  603.  
  604.  
  605.  
  606. v=0
  607.  
  608. o=root 30902895 30902895 IN IP4 127.0.0.1
  609.  
  610. s=Asterisk PBX 11.19.0
  611.  
  612. c=IN IP4 127.0.0.1
  613.  
  614. t=0 0
  615.  
  616. m=audio 19148 RTP/SAVPF 0 8 9
  617.  
  618. a=rtpmap:0 PCMU/8000
  619.  
  620. a=rtpmap:8 PCMA/8000
  621.  
  622. a=rtpmap:9 G722/8000
  623.  
  624. a=ptime:20
  625.  
  626. a=connection:new
  627.  
  628. a=setup:active
  629.  
  630. a=fingerprint:SHA-256 33:29:CC:F2:79:9A:A1:26:31:DD:AF:BA:43:8C:D3:FB:C9:FC:EE:58:24:7C:B4:50:50:E6:7B:BA:93:3A:3E:58
  631.  
  632. a=sendrecv
  633.  
  634.  
  635. " " +6ms" jssip-0.7.4.js:22459:7
  636. "JsSIP:RTCSession " "receiveInviteResponse()" " +9ms" jssip-0.7.4.js:22459:7
  637. "JsSIP:Dialog " "new UAC dialog created with status CONFIRMED" " +1ms" jssip-0.7.4.js:22459:7
  638. "JsSIP:RTCSession " "acceptAndTerminate()" " +5ms" jssip-0.7.4.js:22459:7
  639. "JsSIP:RTCSession " "sendRequest()" " +2ms" jssip-0.7.4.js:22459:7
  640. "JsSIP:RTCSession:Request " "new | ACK" " +1ms" jssip-0.7.4.js:22459:7
  641. "JsSIP:Transport " "sending WebSocket message:
  642.  
  643. ACK sip:200@127.0.0.1:5060;transport=ws SIP/2.0
  644.  
  645. Via: SIP/2.0/WS 192.0.2.31;branch=z9hG4bK7724467
  646.  
  647. Max-Forwards: 69
  648.  
  649. To: <sip:200@127.0.0.1>;tag=as6ea09826
  650.  
  651. From: "UA WebRTC" <sip:8000@127.0.0.1>;tag=75qkalul6h
  652.  
  653. Call-ID: 3n0vmgthqm00ie7fckd7
  654.  
  655. CSeq: 9401 ACK
  656.  
  657. Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER
  658.  
  659. Supported: outbound
  660.  
  661. User-Agent: JsSIP 0.7.4
  662.  
  663. Content-Length: 0
  664.  
  665.  
  666.  
  667.  
  668. " " +4ms" jssip-0.7.4.js:22459:7
  669. "JsSIP:RTCSession " "sendRequest()" " +2ms" jssip-0.7.4.js:22459:7
  670. "JsSIP:RTCSession:Request " "new | BYE" " +1ms" jssip-0.7.4.js:22459:7
  671. "JsSIP:Transport " "sending WebSocket message:
  672.  
  673. BYE sip:200@127.0.0.1:5060;transport=ws SIP/2.0
  674.  
  675. Via: SIP/2.0/WS 192.0.2.31;branch=z9hG4bK8929541
  676.  
  677. Max-Forwards: 69
  678.  
  679. To: <sip:200@127.0.0.1>;tag=as6ea09826
  680.  
  681. From: "UA WebRTC" <sip:8000@127.0.0.1>;tag=75qkalul6h
  682.  
  683. Call-ID: 3n0vmgthqm00ie7fckd7
  684.  
  685. CSeq: 9402 BYE
  686.  
  687. Reason: SIP ;cause=488; text="Not Acceptable Here"
  688.  
  689. Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER
  690.  
  691. Supported: outbound
  692.  
  693. User-Agent: JsSIP 0.7.4
  694.  
  695. Content-Length: 0
  696.  
  697.  
  698.  
  699.  
  700. " " +11ms" jssip-0.7.4.js:22459:7
  701. "JsSIP:RTCSession " "session failed" " +1ms" jssip-0.7.4.js:22459:7
  702. "JsSIP:RTCSession " "close()" " +3ms" jssip-0.7.4.js:22459:7
  703. "JsSIP:Transport " "received WebSocket text message:
  704.  
  705. SIP/2.0 200 OK
  706.  
  707. Via: SIP/2.0/WS 192.0.2.31;branch=z9hG4bK8929541;received=127.0.0.1
  708.  
  709. From: "UA WebRTC" <sip:8000@127.0.0.1>;tag=75qkalul6h
  710.  
  711. To: <sip:200@127.0.0.1>;tag=as6ea09826
  712.  
  713. Call-ID: 3n0vmgthqm00ie7fckd7
  714.  
  715. CSeq: 9402 BYE
  716.  
  717. Server: Asterisk PBX 11.19.0
  718.  
  719. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  720.  
  721. Supported: replaces, timer
  722.  
  723. Content-Length: 0
  724.  
  725.  
  726.  
  727.  
  728. " " +11ms" jssip-0.7.4.js:22459:7
  729. "JsSIP:RTCSession:Request " "onSuccessResponse" " +6ms" jssip-0.7.4.js:22459:7
  730. e.data is undefined
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