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SIP Debug Asterisk WebRTC

navaismo
Aug 2nd, 2014
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  1. Asterisk 11.11.0, Copyright (C) 1999 - 2013 Digium, Inc. and others.
  2. Created by Mark Spencer <[email protected]>
  3. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  4. This is free software, with components licensed under the GNU General Public
  5. License version 2 and other licenses; you are welcome to redistribute it under
  6. certain conditions. Type 'core show license' for details.
  7. =========================================================================
  8. Connected to Asterisk 11.11.0 currently running on asterisk (pid = 2599)
  9. asterisk*CLI>
  10. <--- SIP read from WS:10.0.1.107:51114 --->
  11. INVITE sip:011525555555555@asterisk SIP/2.0
  12. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvq67CGtbT7RG8w2ZOTeyLTChBsM1aM5m;rport
  13. From: "WEBRTC"<sip:[email protected]>;tag=4D1TKAr7hDF8WhKcmaoZ
  14. To: <sip:011525555555555@asterisk>
  15. Contact: "WEBRTC"<sip:[email protected];rtcweb-breaker=no;click2call=no;transport=ws>
  16. Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
  17. CSeq: 35986 INVITE
  18. Content-Type: application/sdp
  19. Content-Length: 1835
  20. Route: <sip:10.0.1.108:5060;lr;sipml5-outbound;transport=udp>
  21. ax-Forwards: 70
  22. User-Agent: DM_SIPWEB-UA
  23. Organization: Digital-Merge
  24.  
  25. v=0
  26. o=- 6498442466790256000 2 IN IP4 127.0.0.1
  27. s=Doubango Telecom - chrome
  28. t=0 0
  29. a=group:BUNDLE audio
  30. a=msid-semantic: WMS t7qGTnlmAnKdVBPKvjVvRUEXAL4mUeASMy0B
  31. m=audio 60478 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
  32. c=IN IP4 192.168.56.1
  33. a=rtcp:60478 IN IP4 192.168.56.1
  34. a=candidate:2999745851 1 udp 2122260223 192.168.56.1 60478 typ host generation 0
  35. a=candidate:2999745851 2 udp 2122260223 192.168.56.1 60478 typ host generation 0
  36. a=candidate:2322994768 1 udp 2122194687 10.0.1.107 60479 typ host generation 0
  37. a=candidate:2322994768 2 udp 2122194687 10.0.1.107 60479 typ host generation 0
  38. a=candidate:4233069003 1 tcp 1518280447 192.168.56.1 0 typ host generation 0
  39. a=candidate:4233069003 2 tcp 1518280447 192.168.56.1 0 typ host generation 0
  40. a=candidate:3304450720 1 tcp 1518214911 10.0.1.107 0 typ host generation 0
  41. a=candidate:3304450720 2 tcp 1518214911 10.0.1.107 0 typ host generation 0
  42. a=ice-ufrag:Z2alDX+HHwpejZTd
  43. a=ice-pwd:V6i3HRU3ygTpQ4ETnNHEJ8xS
  44. a=ice-options:google-ice
  45. a=fingerprint:sha-256 D5:C5:EB:62:8F:E8:C4:D5:6A:C6:EF:2F:65:0C:6B:6B:01:31:FA:42:CD:24:CF:8C:30:98:90:61:E4:59:6B:DD
  46. a=setup:actpass
  47. a=mid:audio
  48. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  49. a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
  50. a=sendrecv
  51. a=rtcp-mux
  52. a=rtpmap:111 opus/48000/2
  53. a=fmtp:111 minptime=10
  54. a=rtpmap:103 ISAC/16000
  55. a=rtpmap:104 ISAC/32000
  56. a=rtpmap:0 PCMU/8000
  57. a=rtpmap:8 PCMA/8000
  58. a=rtpmap:106 CN/32000
  59. a=rtpmap:105 CN/16000
  60. a=rtpmap:13 CN/8000
  61. a=rtpmap:126 telephone-event/8000
  62. a=maxptime:60
  63. a=ssrc:573627479 cname:clMeURHWmYKPuarq
  64. a=ssrc:573627479 msid:t7qGTnlmAnKdVBPKvjVvRUEXAL4mUeASMy0B 3950e683-b0aa-41ed-869f-b3df7dc1066c
  65. a=ssrc:573627479 mslabel:t7qGTnlmAnKdVBPKvjVvRUEXAL4mUeASMy0B
  66. a=ssrc:573627479 label:3950e683-b0aa-41ed-869f-b3df7dc1066c
  67. <------------->
  68. asterisk*CLI>
  69. --- (13 headers 42 lines) ---
  70. asterisk*CLI>
  71. Using INVITE request as basis request - 36bb3d83-bb35-d00f-829f-4180a329a7fa
  72. asterisk*CLI>
  73. Found peer '5005' for '5005' from 10.0.1.107:51114
  74. asterisk*CLI>
  75. <--- Reliably Transmitting (NAT) to 10.0.1.107:51114 --->
  76. SIP/2.0 401 Unauthorized
  77. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvq67CGtbT7RG8w2ZOTeyLTChBsM1aM5m;received=10.0.1.107;rport=51114
  78. From: "WEBRTC"<sip:[email protected]>;tag=4D1TKAr7hDF8WhKcmaoZ
  79. To: <sip:011525555555555@asterisk>;tag=as52ef68de
  80. Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
  81. CSeq: 35986 INVITE
  82. Server: Digital-Merge_UA
  83. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  84. Supported: replaces, timer
  85. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="774b7649"
  86. Content-Length: 0
  87.  
  88.  
  89. <------------>
  90. asterisk*CLI>
  91. Scheduling destruction of SIP dialog '36bb3d83-bb35-d00f-829f-4180a329a7fa' in 8000 ms (Method: INVITE)
  92. asterisk*CLI>
  93. <--- SIP read from WS:10.0.1.107:51114 --->
  94. ACK sip:011525555555555@asterisk SIP/2.0
  95. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvq67CGtbT7RG8w2ZOTeyLTChBsM1aM5m;rport
  96. From: "WEBRTC"<sip:[email protected]>;tag=4D1TKAr7hDF8WhKcmaoZ
  97. To: <sip:011525555555555@asterisk>;tag=as52ef68de
  98. Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
  99. CSeq: 35986 ACK
  100. Content-Length: 0
  101. Route: <sip:10.0.1.108:5060;lr;sipml5-outbound;transport=udp>
  102. ax-Forwards: 70
  103.  
  104. <------------->
  105. asterisk*CLI>
  106. --- (9 headers 0 lines) ---
  107. asterisk*CLI>
  108. <--- SIP read from WS:10.0.1.107:51114 --->
  109. INVITE sip:011525555555555@asterisk SIP/2.0
  110. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKGSGhfHFVUWyo1OsHmyQgdcxL3NFhI9lz;rport
  111. From: "WEBRTC"<sip:[email protected]>;tag=4D1TKAr7hDF8WhKcmaoZ
  112. To: <sip:011525555555555@asterisk>
  113. Contact: "WEBRTC"<sip:[email protected];rtcweb-breaker=no;click2call=no;transport=ws>
  114. Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
  115. CSeq: 35987 INVITE
  116. Content-Type: application/sdp
  117. Content-Length: 1835
  118. Route: <sip:10.0.1.108:5060;lr;sipml5-outbound;transport=udp>
  119. ax-Forwards: 70
  120. Authorization: Digest username="5005",realm="asterisk",nonce="774b7649",uri="sip:011525555555555@asterisk",response="0cf7ef57b65ad0a96f0c2dc0855074ac",algorithm=MD5
  121. User-Agent: DM_SIPWEB-UA
  122. Organization: Digital-Merge
  123.  
  124. v=0
  125. o=- 6498442466790256000 2 IN IP4 127.0.0.1
  126. s=Doubango Telecom - chrome
  127. t=0 0
  128. a=group:BUNDLE audio
  129. a=msid-semantic: WMS t7qGTnlmAnKdVBPKvjVvRUEXAL4mUeASMy0B
  130. m=audio 60478 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
  131. c=IN IP4 192.168.56.1
  132. a=rtcp:60478 IN IP4 192.168.56.1
  133. a=candidate:2999745851 1 udp 2122260223 192.168.56.1 60478 typ host generation 0
  134. a=candidate:2999745851 2 udp 2122260223 192.168.56.1 60478 typ host generation 0
  135. a=candidate:2322994768 1 udp 2122194687 10.0.1.107 60479 typ host generation 0
  136. a=candidate:2322994768 2 udp 2122194687 10.0.1.107 60479 typ host generation 0
  137. a=candidate:4233069003 1 tcp 1518280447 192.168.56.1 0 typ host generation 0
  138. a=candidate:4233069003 2 tcp 1518280447 192.168.56.1 0 typ host generation 0
  139. a=candidate:3304450720 1 tcp 1518214911 10.0.1.107 0 typ host generation 0
  140. a=candidate:3304450720 2 tcp 1518214911 10.0.1.107 0 typ host generation 0
  141. a=ice-ufrag:Z2alDX+HHwpejZTd
  142. a=ice-pwd:V6i3HRU3ygTpQ4ETnNHEJ8xS
  143. a=ice-options:google-ice
  144. a=fingerprint:sha-256 D5:C5:EB:62:8F:E8:C4:D5:6A:C6:EF:2F:65:0C:6B:6B:01:31:FA:42:CD:24:CF:8C:30:98:90:61:E4:59:6B:DD
  145. a=setup:actpass
  146. a=mid:audio
  147. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  148. a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
  149. a=sendrecv
  150. a=rtcp-mux
  151. a=rtpmap:111 opus/48000/2
  152. a=fmtp:111 minptime=10
  153. a=rtpmap:103 ISAC/16000
  154. a=rtpmap:104 ISAC/32000
  155. a=rtpmap:0 PCMU/8000
  156. a=rtpmap:8 PCMA/8000
  157. a=rtpmap:106 CN/32000
  158. a=rtpmap:105 CN/16000
  159. a=rtpmap:13 CN/8000
  160. a=rtpmap:126 telephone-event/8000
  161. a=maxptime:60
  162. a=ssrc:573627479 cname:clMeURHWmYKPuarq
  163. a=ssrc:573627479 msid:t7qGTnlmAnKdVBPKvjVvRUEXAL4mUeASMy0B 3950e683-b0aa-41ed-869f-b3df7dc1066c
  164. a=ssrc:573627479 mslabel:t7qGTnlmAnKdVBPKvjVvRUEXAL4mUeASMy0B
  165. a=ssrc:573627479 label:3950e683-b0aa-41ed-869f-b3df7dc1066c
  166.  
  167. <------------->
  168. --- (14 headers 42 lines) ---
  169. Using INVITE request as basis request - 36bb3d83-bb35-d00f-829f-4180a329a7fa
  170. Found peer '5005' for '5005' from 10.0.1.107:51114
  171. asterisk*CLI>
  172. == Using SIP RTP CoS mark 5
  173. asterisk*CLI>
  174. Found RTP audio format 111
  175. asterisk*CLI>
  176. Found RTP audio format 103
  177. asterisk*CLI>
  178. Found RTP audio format 104
  179. asterisk*CLI>
  180. Found RTP audio format 0
  181. asterisk*CLI>
  182. Found RTP audio format 8
  183. asterisk*CLI>
  184. Found RTP audio format 106
  185. asterisk*CLI>
  186. Found RTP audio format 105
  187. asterisk*CLI>
  188. Found RTP audio format 13
  189. asterisk*CLI>
  190. Found RTP audio format 126
  191. asterisk*CLI>
  192. Found unknown media description format opus for ID 111
  193. asterisk*CLI>
  194. Found unknown media description format ISAC for ID 103
  195. asterisk*CLI>
  196. Found unknown media description format ISAC for ID 104
  197. asterisk*CLI>
  198. Found audio description format PCMU for ID 0
  199. Found audio description format PCMA for ID 8
  200. Found unknown media description format CN for ID 106
  201. asterisk*CLI>
  202. Found unknown media description format CN for ID 105
  203. Found audio description format CN for ID 13
  204. Found audio description format telephone-event for ID 126
  205. asterisk*CLI>
  206. Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  207. asterisk*CLI>
  208. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
  209. asterisk*CLI>
  210. Peer audio RTP is at port 192.168.56.1:60478
  211. asterisk*CLI>
  212. Looking for 011525555555555 in wrtc (domain asterisk)
  213. asterisk*CLI>
  214. list_route: hop: <sip:[email protected];rtcweb-breaker=no;click2call=no;transport=ws>
  215. asterisk*CLI>
  216. <--- Transmitting (NAT) to 10.0.1.107:51114 --->
  217. SIP/2.0 100 Trying
  218. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKGSGhfHFVUWyo1OsHmyQgdcxL3NFhI9lz;received=10.0.1.107;rport=51114
  219. From: "WEBRTC"<sip:[email protected]>;tag=4D1TKAr7hDF8WhKcmaoZ
  220. To: <sip:011525555555555@asterisk>
  221. Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
  222. CSeq: 35987 INVITE
  223. Server: Digital-Merge_UA
  224. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  225. Supported: replaces, timer
  226. Contact: <sip:[email protected]:5060;transport=WS>
  227. Content-Length: 0
  228.  
  229.  
  230. <------------>
  231. -- Executing [011525555555555@wrtc:1] Dial("SIP/5005-0000000d", "SIP/MySuperSIPProvider/011525555555555,40,TtWw") in new stack
  232. asterisk*CLI>
  233. == Using SIP RTP CoS mark 5
  234. asterisk*CLI>
  235. Audio is at 20974
  236. asterisk*CLI>
  237. Adding codec 100003 (ulaw) to SDP
  238. asterisk*CLI>
  239. Adding codec 100002 (gsm) to SDP
  240. asterisk*CLI>
  241. Adding codec 100004 (alaw) to SDP
  242. asterisk*CLI>
  243. Adding non-codec 0x1 (telephone-event) to SDP
  244. asterisk*CLI>
  245. Reliably Transmitting (NAT) to 74.54.XXX.XXX:5060:
  246. INVITE sip:[email protected] SIP/2.0
  247. Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK53edb567;rport
  248. ax-Forwards: 70
  249. From: "WebRTC" <sip:[email protected]>;tag=as613a93d5
  250. To: <sip:[email protected]>
  251. Contact: <sip:[email protected]:5060>
  252. Call-ID: [email protected]:5060
  253. CSeq: 102 INVITE
  254. User-Agent: Digital-Merge_UA
  255. Date: Sat, 02 Aug 2014 17:58:19 GMT
  256. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  257. Supported: replaces, timer
  258. Remote-Party-ID: "WebRTC" <sip:[email protected]>;party=calling;privacy=off;screen=no
  259. Content-Type: application/sdp
  260. Content-Length: 279
  261.  
  262. v=0
  263. o=root 1175191988 1175191988 IN IP4 10.0.1.108
  264. s=Asterisk PBX 11.11.0
  265. c=IN IP4 10.0.1.108
  266. t=0 0
  267. m=audio 20974 RTP/AVP 0 3 8 101
  268. a=rtpmap:0 PCMU/8000
  269. a=rtpmap:3 GSM/8000
  270. a=rtpmap:8 PCMA/8000
  271. a=rtpmap:101 telephone-event/8000
  272. a=fmtp:101 0-16
  273. a=ptime:20
  274. a=sendrecv
  275.  
  276. ---
  277. -- Called SIP/MySuperSIPProvider/011525555555555
  278. asterisk*CLI>
  279. Retransmitting #1 (NAT) to 74.54.XXX.XXX:5060:
  280. INVITE sip:[email protected] SIP/2.0
  281. Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK53edb567;rport
  282. ax-Forwards: 70
  283. From: "WebRTC" <sip:[email protected]>;tag=as613a93d5
  284. To: <sip:[email protected]>
  285. Contact: <sip:[email protected]:5060>
  286. Call-ID: [email protected]:5060
  287. CSeq: 102 INVITE
  288. User-Agent: Digital-Merge_UA
  289. Date: Sat, 02 Aug 2014 17:58:19 GMT
  290. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  291. Supported: replaces, timer
  292. Remote-Party-ID: "WebRTC" <sip:[email protected]>;party=calling;privacy=off;screen=no
  293. Content-Type: application/sdp
  294. Content-Length: 279
  295.  
  296. v=0
  297. o=root 1175191988 1175191988 IN IP4 10.0.1.108
  298. s=Asterisk PBX 11.11.0
  299. c=IN IP4 10.0.1.108
  300. t=0 0
  301. m=audio 20974 RTP/AVP 0 3 8 101
  302. a=rtpmap:0 PCMU/8000
  303. a=rtpmap:3 GSM/8000
  304. a=rtpmap:8 PCMA/8000
  305. a=rtpmap:101 telephone-event/8000
  306. a=fmtp:101 0-16
  307. a=ptime:20
  308. a=sendrecv
  309.  
  310. ---
  311. asterisk*CLI>
  312. Retransmitting #2 (NAT) to 74.54.XXX.XXX:5060:
  313. INVITE sip:[email protected] SIP/2.0
  314. Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK53edb567;rport
  315. ax-Forwards: 70
  316. From: "WebRTC" <sip:[email protected]>;tag=as613a93d5
  317. To: <sip:[email protected]>
  318. Contact: <sip:[email protected]:5060>
  319. Call-ID: [email protected]:5060
  320. CSeq: 102 INVITE
  321. User-Agent: Digital-Merge_UA
  322. Date: Sat, 02 Aug 2014 17:58:19 GMT
  323. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  324. Supported: replaces, timer
  325. Remote-Party-ID: "WebRTC" <sip:[email protected]>;party=calling;privacy=off;screen=no
  326. Content-Type: application/sdp
  327. Content-Length: 279
  328.  
  329. v=0
  330. o=root 1175191988 1175191988 IN IP4 10.0.1.108
  331. s=Asterisk PBX 11.11.0
  332. c=IN IP4 10.0.1.108
  333. t=0 0
  334. m=audio 20974 RTP/AVP 0 3 8 101
  335. a=rtpmap:0 PCMU/8000
  336. a=rtpmap:3 GSM/8000
  337. a=rtpmap:8 PCMA/8000
  338. a=rtpmap:101 telephone-event/8000
  339. a=fmtp:101 0-16
  340. a=ptime:20
  341. a=sendrecv
  342.  
  343. ---
  344. asterisk*CLI>
  345. <--- SIP read from UDP:74.54.XXX.XXX:5060 --->
  346. SIP/2.0 407 Proxy Authentication Required
  347. Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK53edb567;received=189.241.6.199;rport=13027
  348. From: "WebRTC" <sip:[email protected]>;tag=as613a93d5
  349. To: <sip:[email protected]>;tag=as52d6980b
  350. Call-ID: [email protected]:5060
  351. CSeq: 102 INVITE
  352. User-Agent: voip.ms
  353. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  354. Supported: replaces
  355. Proxy-Authenticate: Digest algorithm=MD5, realm="dallas.voip.ms", nonce="795f186d"
  356. Content-Length: 0
  357.  
  358. <------------->
  359. asterisk*CLI>
  360. --- (11 headers 0 lines) ---
  361. asterisk*CLI>
  362. Transmitting (NAT) to 74.54.XXX.XXX:5060:
  363. ACK sip:[email protected] SIP/2.0
  364. Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK53edb567;rport
  365. ax-Forwards: 70
  366. From: "WebRTC" <sip:[email protected]>;tag=as613a93d5
  367. To: <sip:[email protected]>;tag=as52d6980b
  368. Contact: <sip:[email protected]:5060>
  369. Call-ID: [email protected]:5060
  370. CSeq: 102 ACK
  371. User-Agent: Digital-Merge_UA
  372. Content-Length: 0
  373.  
  374.  
  375. ---
  376. asterisk*CLI>
  377. Audio is at 20974
  378. asterisk*CLI>
  379. Adding codec 100003 (ulaw) to SDP
  380. Adding codec 100002 (gsm) to SDP
  381. Adding codec 100004 (alaw) to SDP
  382. Adding non-codec 0x1 (telephone-event) to SDP
  383. asterisk*CLI>
  384. Reliably Transmitting (NAT) to 74.54.XXX.XXX:5060:
  385. INVITE sip:[email protected] SIP/2.0
  386. Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK404698f6;rport
  387. ax-Forwards: 70
  388. From: "WebRTC" <sip:[email protected]>;tag=as613a93d5
  389. To: <sip:[email protected]>
  390. Contact: <sip:[email protected]:5060>
  391. Call-ID: [email protected]:5060
  392. CSeq: 103 INVITE
  393. User-Agent: Digital-Merge_UA
  394. Proxy-Authorization: Digest username="143987", realm="dallas.voip.ms", algorithm=MD5, uri="sip:[email protected]", nonce="795f186d", response="d8bf9497de2ffae33a1e09729300a777"
  395. Date: Sat, 02 Aug 2014 17:58:20 GMT
  396. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  397. Supported: replaces, timer
  398. Remote-Party-ID: "WebRTC" <sip:[email protected]>;party=calling;privacy=off;screen=no
  399. Content-Type: application/sdp
  400. Content-Length: 279
  401.  
  402. v=0
  403. o=root 1175191988 1175191989 IN IP4 10.0.1.108
  404. s=Asterisk PBX 11.11.0
  405. c=IN IP4 10.0.1.108
  406. t=0 0
  407. m=audio 20974 RTP/AVP 0 3 8 101
  408. a=rtpmap:0 PCMU/8000
  409. a=rtpmap:3 GSM/8000
  410. a=rtpmap:8 PCMA/8000
  411. a=rtpmap:101 telephone-event/8000
  412. a=fmtp:101 0-16
  413. a=ptime:20
  414. a=sendrecv
  415.  
  416. ---
  417.  
  418. <--- SIP read from UDP:74.54.XXX.XXX:5060 --->
  419. SIP/2.0 100 Trying
  420. Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK404698f6;received=189.241.6.199;rport=13027
  421. From: "WebRTC" <sip:[email protected]>;tag=as613a93d5
  422. To: <sip:[email protected]>
  423. Call-ID: [email protected]:5060
  424. CSeq: 103 INVITE
  425. User-Agent: voip.ms
  426. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  427. Supported: replaces
  428. Contact: <sip:[email protected]>
  429. Content-Length: 0
  430.  
  431. <------------->
  432. --- (11 headers 0 lines) ---
  433. asterisk*CLI>
  434. <--- SIP read from UDP:74.54.XXX.XXX:5060 --->
  435. SIP/2.0 183 Session Progress
  436. Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK404698f6;received=189.241.6.199;rport=13027
  437. From: "WebRTC" <sip:[email protected]>;tag=as613a93d5
  438. To: <sip:[email protected]>;tag=as15352569
  439. Call-ID: [email protected]:5060
  440. CSeq: 103 INVITE
  441. User-Agent: voip.ms
  442. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  443. Supported: replaces
  444. Contact: <sip:[email protected]>
  445. Content-Type: application/sdp
  446. Content-Length: 263
  447.  
  448. v=0
  449. o=root 18731 18731 IN IP4 74.54.XXX.XXX
  450. s=session
  451. c=IN IP4 74.54.XXX.XXX
  452. t=0 0
  453. m=audio 18042 RTP/AVP 0 3 101
  454. a=rtpmap:0 PCMU/8000
  455. a=rtpmap:3 GSM/8000
  456. a=rtpmap:101 telephone-event/8000
  457. a=fmtp:101 0-16
  458. a=silenceSupp:off - - - -
  459. a=ptime:20
  460. a=sendrecv
  461. <------------->
  462. asterisk*CLI>
  463. --- (12 headers 13 lines) ---
  464. asterisk*CLI>
  465. list_route: hop: <sip:[email protected]>
  466. Found RTP audio format 0
  467. Found RTP audio format 3
  468. Found RTP audio format 101
  469. Found audio description format PCMU for ID 0
  470. Found audio description format GSM for ID 3
  471. Found audio description format telephone-event for ID 101
  472. Capabilities: us - (gsm|ulaw|alaw), peer - audio=(gsm|ulaw)/video=(nothing)/text=(nothing), combined - (gsm|ulaw)
  473. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  474. Peer audio RTP is at port 74.54.XXX.XXX:18042
  475. -- SIP/MySuperSIPProvider-0000000e is making progress passing it to SIP/5005-0000000d
  476. asterisk*CLI>
  477. Audio is at 19880
  478. asterisk*CLI>
  479. Adding codec 100003 (ulaw) to SDP
  480. asterisk*CLI>
  481. Adding codec 100004 (alaw) to SDP
  482. asterisk*CLI>
  483. Adding non-codec 0x1 (telephone-event) to SDP
  484. asterisk*CLI>
  485. <--- Transmitting (NAT) to 10.0.1.107:51114 --->
  486. SIP/2.0 183 Session Progress
  487. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKGSGhfHFVUWyo1OsHmyQgdcxL3NFhI9lz;received=10.0.1.107;rport=51114
  488. From: "WEBRTC"<sip:[email protected]>;tag=4D1TKAr7hDF8WhKcmaoZ
  489. To: <sip:011525555555555@asterisk>;tag=as654c7257
  490. Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
  491. CSeq: 35987 INVITE
  492. Server: Digital-Merge_UA
  493. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  494. Supported: replaces, timer
  495. Contact: <sip:[email protected]:5060;transport=WS>
  496. Content-Type: application/sdp
  497. Content-Length: 777
  498.  
  499. v=0
  500. o=root 462328627 462328627 IN IP4 10.0.1.108
  501. s=Asterisk PBX 11.11.0
  502. c=IN IP4 10.0.1.108
  503. t=0 0
  504. m=audio 19880 UDP/TLS/RTP/SAVPF 0 8 126
  505. a=rtpmap:0 PCMU/8000
  506. a=rtpmap:8 PCMA/8000
  507. a=rtpmap:126 telephone-event/8000
  508. a=fmtp:126 0-16
  509. a=ptime:20
  510. a=ice-ufrag:4693399c4251f1db40f8054a159138d7
  511. a=ice-pwd:0d4f0a8861da5d277fd5f466337a088d
  512. a=candidate:Ha00016c 1 UDP 2130706431 10.0.1.108 19880 typ host
  513. a=candidate:Sbdf106c7 1 UDP 1694498815 189.241.6.199 13082 typ srflx
  514. a=candidate:Ha00016c 2 UDP 2130706430 10.0.1.108 19881 typ host
  515. a=candidate:Sbdf106c7 2 UDP 1694498814 189.241.6.199 13084 typ srflx
  516. a=connection:new
  517. a=setup:active
  518. a=fingerprint:SHA-256 49:BF:9C:44:4B:E8:63:28:31:3C:36:7D:7C:F9:DC:6A:C8:AF:71:C0:E3:3D:36:E0:87:C0:27:00:9E:FC:FC:6A
  519. a=sendrecv
  520.  
  521. <------------>
  522. asterisk*CLI>
  523. > 0xb7511138 -- Probation passed - setting RTP source address to 10.0.1.107:60479
  524. asterisk*CLI>
  525. > 0xb7511138 -- Probation passed - setting RTP source address to 10.0.1.107:60479
  526. asterisk*CLI>
  527. > 0xb7382f18 -- Probation passed - setting RTP source address to 74.54.XXX.XXX:18042
  528. asterisk*CLI>
  529. <--- SIP read from UDP:74.54.XXX.XXX:5060 --->
  530. SIP/2.0 200 OK
  531. Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK404698f6;received=189.241.6.199;rport=13027
  532. From: "WebRTC" <sip:[email protected]>;tag=as613a93d5
  533. To: <sip:[email protected]>;tag=as15352569
  534. Call-ID: [email protected]:5060
  535. CSeq: 103 INVITE
  536. User-Agent: voip.ms
  537. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  538. Supported: replaces
  539. Contact: <sip:[email protected]>
  540. Content-Type: application/sdp
  541. Content-Length: 263
  542.  
  543. v=0
  544. o=root 18731 18732 IN IP4 74.54.XXX.XXX
  545. s=session
  546. c=IN IP4 74.54.XXX.XXX
  547. t=0 0
  548. m=audio 18042 RTP/AVP 0 3 101
  549. a=rtpmap:0 PCMU/8000
  550. a=rtpmap:3 GSM/8000
  551. a=rtpmap:101 telephone-event/8000
  552. a=fmtp:101 0-16
  553. a=silenceSupp:off - - - -
  554. a=ptime:20
  555. a=sendrecv
  556. <------------->
  557. asterisk*CLI>
  558. --- (12 headers 13 lines) ---
  559. asterisk*CLI>
  560. Found RTP audio format 0
  561. asterisk*CLI>
  562. Found RTP audio format 3
  563. Found RTP audio format 101
  564. asterisk*CLI>
  565. Found audio description format PCMU for ID 0
  566. Found audio description format GSM for ID 3
  567. Found audio description format telephone-event for ID 101
  568. Capabilities: us - (gsm|ulaw|alaw), peer - audio=(gsm|ulaw)/video=(nothing)/text=(nothing), combined - (gsm|ulaw)
  569. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  570. asterisk*CLI>
  571. Peer audio RTP is at port 74.54.XXX.XXX:18042
  572. list_route: hop: <sip:[email protected]>
  573. set_destination: Parsing <sip:[email protected]> for address/port to send to
  574. set_destination: set destination to 74.54.XXX.XXX:5060
  575. Transmitting (NAT) to 74.54.XXX.XXX:5060:
  576. ACK sip:[email protected] SIP/2.0
  577. Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK0152723a;rport
  578. ax-Forwards: 70
  579. From: "WebRTC" <sip:[email protected]>;tag=as613a93d5
  580. To: <sip:[email protected]>;tag=as15352569
  581. Contact: <sip:[email protected]:5060>
  582. Call-ID: [email protected]:5060
  583. CSeq: 103 ACK
  584. User-Agent: Digital-Merge_UA
  585. Content-Length: 0
  586.  
  587.  
  588. ---
  589. -- SIP/MySuperSIPProvider-0000000e answered SIP/5005-0000000d
  590. Audio is at 19880
  591. Adding codec 100003 (ulaw) to SDP
  592. Adding codec 100004 (alaw) to SDP
  593. Adding non-codec 0x1 (telephone-event) to SDP
  594.  
  595. <--- Reliably Transmitting (NAT) to 10.0.1.107:51114 --->
  596. SIP/2.0 200 OK
  597. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKGSGhfHFVUWyo1OsHmyQgdcxL3NFhI9lz;received=10.0.1.107;rport=51114
  598. From: "WEBRTC"<sip:[email protected]>;tag=4D1TKAr7hDF8WhKcmaoZ
  599. To: <sip:011525555555555@asterisk>;tag=as654c7257
  600. Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
  601. CSeq: 35987 INVITE
  602. Server: Digital-Merge_UA
  603. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  604. Supported: replaces, timer
  605. Contact: <sip:[email protected]:5060;transport=WS>
  606. Content-Type: application/sdp
  607. Content-Length: 782
  608.  
  609. v=0
  610. o=root 462328627 462328627 IN IP4 10.0.1.108
  611. s=Asterisk PBX 11.11.0
  612. c=IN IP4 10.0.1.108
  613. t=0 0
  614. m=audio 19880 UDP/TLS/RTP/SAVPF 0 8 126
  615. a=rtpmap:0 PCMU/8000
  616. a=rtpmap:8 PCMA/8000
  617. a=rtpmap:126 telephone-event/8000
  618. a=fmtp:126 0-16
  619. a=ptime:20
  620. a=ice-ufrag:4693399c4251f1db40f8054a159138d7
  621. a=ice-pwd:0d4f0a8861da5d277fd5f466337a088d
  622. a=candidate:Ha00016c 1 UDP 2130706431 10.0.1.108 19880 typ host
  623. a=candidate:Sbdf106c7 1 UDP 1694498815 189.241.6.199 13082 typ srflx
  624. a=candidate:Ha00016c 2 UDP 2130706430 10.0.1.108 19881 typ host
  625. a=candidate:Sbdf106c7 2 UDP 1694498814 189.241.6.199 13084 typ srflx
  626. a=connection:existing
  627. a=setup:active
  628. a=fingerprint:SHA-256 49:BF:9C:44:4B:E8:63:28:31:3C:36:7D:7C:F9:DC:6A:C8:AF:71:C0:E3:3D:36:E0:87:C0:27:00:9E:FC:FC:6A
  629. a=sendrecv
  630.  
  631. <------------>
  632. asterisk*CLI>
  633. > 0xb7382f18 -- Probation passed - setting RTP source address to 74.54.XXX.XXX:18042
  634. asterisk*CLI>
  635. <--- SIP read from WS:10.0.1.107:51114 --->
  636. ACK sip:[email protected]:5060;transport=WS SIP/2.0
  637. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKMrQxdOwCdr9pIGWDZPg4;rport
  638. From: "WEBRTC"<sip:[email protected]>;tag=4D1TKAr7hDF8WhKcmaoZ
  639. To: <sip:011525555555555@asterisk>;tag=as654c7257
  640. Contact: "WEBRTC"<sip:[email protected];rtcweb-breaker=no;click2call=no;transport=ws>
  641. Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
  642. CSeq: 35987 ACK
  643. Content-Length: 0
  644. Route: <sip:10.0.1.108:5060;lr;sipml5-outbound;transport=udp>
  645. ax-Forwards: 70
  646. Authorization: Digest username="5005",realm="asterisk",nonce="774b7649",uri="sip:[email protected]:5060;transport=WS",response="ddb5a7829e26a98edbe09483987c894d",algorithm=MD5
  647. User-Agent: DM_SIPWEB-UA
  648. Organization: Digital-Merge
  649.  
  650. <------------->
  651. --- (13 headers 0 lines) ---
  652. asterisk*CLI> rtp set debug on
  653. asterisk*CLI> RTP Debugging Enabled
  654. asterisk*CLI>
  655. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039707, ts 3166686328, len 000033)
  656. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033589, ts 3166686328, len 000170)
  657. asterisk*CLI>
  658. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018367, ts 1324339245, len 000160)
  659. asterisk*CLI>
  660. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041587, ts 1324339240, len 000033)
  661. asterisk*CLI>
  662. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039708, ts 3166686488, len 000033)
  663. asterisk*CLI>
  664. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033590, ts 3166686488, len 000170)
  665. asterisk*CLI>
  666. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018368, ts 1324339405, len 000160)
  667. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041588, ts 1324339400, len 000033)
  668. asterisk*CLI>
  669. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039709, ts 3166686648, len 000033)
  670. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033591, ts 3166686648, len 000170)
  671. asterisk*CLI>
  672. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018369, ts 1324339565, len 000160)
  673. asterisk*CLI>
  674. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041589, ts 1324339560, len 000033)
  675. asterisk*CLI>
  676. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039710, ts 3166686808, len 000033)
  677. asterisk*CLI>
  678. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033592, ts 3166686808, len 000170)
  679. asterisk*CLI>
  680. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018370, ts 1324339725, len 000160)
  681. asterisk*CLI>
  682. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041590, ts 1324339720, len 000033)
  683. asterisk*CLI>
  684. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039711, ts 3166686968, len 000033)
  685. asterisk*CLI>
  686. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033593, ts 3166686968, len 000170)
  687. asterisk*CLI>
  688. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018371, ts 1324339885, len 000160)
  689. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041591, ts 1324339880, len 000033)
  690. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039712, ts 3166687128, len 000033)
  691. asterisk*CLI>
  692. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033594, ts 3166687128, len 000170)
  693. asterisk*CLI>
  694. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039713, ts 3166687288, len 000033)
  695. asterisk*CLI>
  696. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033595, ts 3166687288, len 000170)
  697. asterisk*CLI>
  698. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018372, ts 1324340045, len 000160)
  699. asterisk*CLI>
  700. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041592, ts 1324340040, len 000033)
  701. asterisk*CLI>
  702. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039714, ts 3166687448, len 000033)
  703. asterisk*CLI>
  704. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033596, ts 3166687448, len 000170)
  705. asterisk*CLI>
  706. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018373, ts 1324340205, len 000160)
  707. asterisk*CLI>
  708. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041593, ts 1324340200, len 000033)
  709. asterisk*CLI>
  710. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039715, ts 3166687608, len 000033)
  711. asterisk*CLI>
  712. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033597, ts 3166687608, len 000170)
  713. asterisk*CLI>
  714. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018374, ts 1324340365, len 000160)
  715. asterisk*CLI>
  716. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041594, ts 1324340360, len 000033)
  717. asterisk*CLI>
  718. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039716, ts 3166687768, len 000033)
  719. asterisk*CLI>
  720. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033598, ts 3166687768, len 000170)
  721. asterisk*CLI>
  722. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018375, ts 1324340525, len 000160)
  723. asterisk*CLI>
  724. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041595, ts 1324340520, len 000033)
  725. asterisk*CLI>
  726. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039717, ts 3166687928, len 000033)
  727. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033599, ts 3166687928, len 000170)
  728. asterisk*CLI>
  729. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018376, ts 1324340685, len 000160)
  730. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041596, ts 1324340680, len 000033)
  731. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039718, ts 3166688088, len 000033)
  732. asterisk*CLI>
  733. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033600, ts 3166688088, len 000170)
  734. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018377, ts 1324340845, len 000160)
  735. asterisk*CLI>
  736. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041597, ts 1324340840, len 000033)
  737. asterisk*CLI>
  738. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039719, ts 3166688248, len 000033)
  739. asterisk*CLI>
  740. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033601, ts 3166688248, len 000170)
  741. asterisk*CLI>
  742. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018378, ts 1324341005, len 000160)
  743. asterisk*CLI> rtp set debug on
  744. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041598, ts 1324341000, len 000033)
  745. asterisk*CLI> rtp set debug on
  746. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039720, ts 3166688408, len 000033)
  747. asterisk*CLI> rtp set debug on
  748. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033602, ts 3166688408, len 000170)
  749. asterisk*CLI> rtp set debug on
  750. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018379, ts 1324341165, len 000160)
  751. asterisk*CLI> rtp set debug on
  752. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041599, ts 1324341160, len 000033)
  753. asterisk*CLI> rtp set debug on
  754. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039721, ts 3166688568, len 000033)
  755. asterisk*CLI> rtp set debug on
  756. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033603, ts 3166688568, len 000170)
  757. asterisk*CLI> rtp set debug on
  758. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039722, ts 3166688728, len 000033)
  759. asterisk*CLI> rtp set debug on
  760. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033604, ts 3166688728, len 000170)
  761. asterisk*CLI> rtp set debug on
  762. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018380, ts 1324341325, len 000160)
  763. asterisk*CLI> rtp set debug on
  764. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041600, ts 1324341320, len 000033)
  765. asterisk*CLI> rtp set debug on
  766. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039723, ts 3166688888, len 000033)
  767. asterisk*CLI> rtp set debug on
  768. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033605, ts 3166688888, len 000170)
  769. asterisk*CLI> rtp set debug on
  770. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018381, ts 1324341485, len 000160)
  771. asterisk*CLI> rtp set debug on
  772. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041601, ts 1324341480, len 000033)
  773. asterisk*CLI> rtp set debug on
  774. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039724, ts 3166689048, len 000033)
  775. asterisk*CLI> rtp set debug on
  776. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033606, ts 3166689048, len 000170)
  777. asterisk*CLI> rtp set debug on
  778. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039725, ts 3166689208, len 000033)
  779. asterisk*CLI> rtp set debug on
  780. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033607, ts 3166689208, len 000170)
  781. asterisk*CLI> rtp set debug on
  782. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018382, ts 1324341645, len 000160)
  783. asterisk*CLI> rtp set debug on
  784. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041602, ts 1324341640, len 000033)
  785. asterisk*CLI> rtp set debug on
  786. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039726, ts 3166689368, len 000033)
  787. asterisk*CLI> rtp set debug on
  788. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033608, ts 3166689368, len 000170)
  789. asterisk*CLI> rtp set debug on
  790. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018383, ts 1324341805, len 000160)
  791. asterisk*CLI> rtp set debug on
  792. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041603, ts 1324341800, len 000033)
  793. asterisk*CLI> rtp set debug on
  794. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039727, ts 3166689528, len 000033)
  795. asterisk*CLI> rtp set debug on
  796. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033609, ts 3166689528, len 000170)
  797. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018384, ts 1324341965, len 000160)
  798. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041604, ts 1324341960, len 000033)
  799. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039728, ts 3166689688, len 000033)
  800. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033610, ts 3166689688, len 000170)
  801. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018385, ts 1324342125, len 000160)
  802. asterisk*CLI> rtp set debug on
  803. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041605, ts 1324342120, len 000033)
  804. asterisk*CLI> rtp set debug on
  805. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039729, ts 3166689848, len 000033)
  806. asterisk*CLI> rtp set debug on
  807. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033611, ts 3166689848, len 000170)
  808. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018386, ts 1324342285, len 000160)
  809. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041606, ts 1324342280, len 000033)
  810. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039730, ts 3166690008, len 000033)
  811. asterisk*CLI> rtp set debug on
  812. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033612, ts 3166690008, len 000170)
  813. asterisk*CLI> rtp set debug on
  814. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018387, ts 1324342445, len 000160)
  815. asterisk*CLI> rtp set debug on
  816. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041607, ts 1324342440, len 000033)
  817. asterisk*CLI> rtp set debug on
  818. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039731, ts 3166690168, len 000033)
  819. asterisk*CLI> rtp set debug on
  820. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033613, ts 3166690168, len 000170)
  821. asterisk*CLI> rtp set debug on
  822. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018388, ts 1324342605, len 000160)
  823. asterisk*CLI> rtp set debug on
  824. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041608, ts 1324342600, len 000033)
  825. asterisk*CLI> rtp set debug on
  826. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039732, ts 3166690328, len 000033)
  827. asterisk*CLI> rtp set debug on
  828. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033614, ts 3166690328, len 000170)
  829. asterisk*CLI> rtp set debug on
  830. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039733, ts 3166690488, len 000033)
  831. asterisk*CLI> rtp set debug on
  832. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033615, ts 3166690488, len 000170)
  833. asterisk*CLI> rtp set debug o
  834. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018389, ts 1324342765, len 000160)
  835. asterisk*CLI> rtp set debug o
  836. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041609, ts 1324342760, len 000033)
  837. asterisk*CLI> rtp set debug o
  838. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039734, ts 3166690648, len 000033)
  839. asterisk*CLI> rtp set debug o
  840. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033616, ts 3166690648, len 000170)
  841. asterisk*CLI> rtp set debug o
  842. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018390, ts 1324342925, len 000160)
  843. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041610, ts 1324342920, len 000033)
  844. asterisk*CLI> rtp set debug o
  845. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039735, ts 3166690808, len 000033)
  846. asterisk*CLI> rtp set debug o
  847. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033617, ts 3166690808, len 000170)
  848. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018391, ts 1324343085, len 000160)
  849. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041611, ts 1324343080, len 000033)
  850. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039736, ts 3166690968, len 000033)
  851. asterisk*CLI> rtp set debug o
  852. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033618, ts 3166690968, len 000170)
  853. asterisk*CLI> rtp set debug o
  854. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018392, ts 1324343245, len 000160)
  855. asterisk*CLI> rtp set debug o
  856. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041612, ts 1324343240, len 000033)
  857. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039737, ts 3166691128, len 000033)
  858. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033619, ts 3166691128, len 000170)
  859. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018393, ts 1324343405, len 000160)
  860. asterisk*CLI> rtp set debug o
  861. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041613, ts 1324343400, len 000033)
  862. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039738, ts 3166691288, len 000033)
  863. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033620, ts 3166691288, len 000170)
  864. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018394, ts 1324343565, len 000160)
  865. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041614, ts 1324343560, len 000033)
  866. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039739, ts 3166691448, len 000033)
  867. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033621, ts 3166691448, len 000170)
  868. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018395, ts 1324343725, len 000160)
  869. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041615, ts 1324343720, len 000033)
  870. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039740, ts 3166691608, len 000033)
  871. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033622, ts 3166691608, len 000170)
  872. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018396, ts 1324343885, len 000160)
  873. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041616, ts 1324343880, len 000033)
  874. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039741, ts 3166691768, len 000033)
  875. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033623, ts 3166691768, len 000170)
  876. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018397, ts 1324344045, len 000160)
  877. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041617, ts 1324344040, len 000033)
  878. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039742, ts 3166691928, len 000033)
  879. asterisk*CLI> rtp set debug o
  880. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033624, ts 3166691928, len 000170)
  881. asterisk*CLI> rtp set debug o
  882. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018398, ts 1324344205, len 000160)
  883. asterisk*CLI> rtp set debug o
  884. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041618, ts 1324344200, len 000033)
  885. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039743, ts 3166692088, len 000033)
  886. asterisk*CLI> rtp set debug o
  887. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033625, ts 3166692088, len 000170)
  888. asterisk*CLI> rtp set debug o
  889. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039744, ts 3166692248, len 000033)
  890. asterisk*CLI> rtp set debug o
  891. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033626, ts 3166692248, len 000170)
  892. asterisk*CLI> rtp set debug o
  893. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018399, ts 1324344365, len 000160)
  894. asterisk*CLI> rtp set debug o
  895. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041619, ts 1324344360, len 000033)
  896. asterisk*CLI> rtp set debug of
  897. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039745, ts 3166692408, len 000033)
  898. asterisk*CLI> rtp set debug of
  899. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033627, ts 3166692408, len 000170)
  900. asterisk*CLI> rtp set debug of
  901. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018400, ts 1324344525, len 000160)
  902. asterisk*CLI> rtp set debug of
  903. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041620, ts 1324344520, len 000033)
  904. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039746, ts 3166692568, len 000033)
  905. asterisk*CLI> rtp set debug off
  906. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033628, ts 3166692568, len 000170)
  907. asterisk*CLI> rtp set debug off
  908. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018401, ts 1324344685, len 000160)
  909. asterisk*CLI> rtp set debug off
  910. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041621, ts 1324344680, len 000033)
  911. asterisk*CLI> rtp set debug off
  912. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039747, ts 3166692728, len 000033)
  913. asterisk*CLI> rtp set debug off
  914. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033629, ts 3166692728, len 000170)
  915. asterisk*CLI> rtp set debug off
  916. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018402, ts 1324344845, len 000160)
  917. asterisk*CLI> rtp set debug off
  918. asterisk*CLI> Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041622, ts 1324344840, len 000033)
  919. asterisk*CLI>
  920. RTP Debugging Disabled
  921. asterisk*CLI>
  922. Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039748, ts 3166692888, len 000033)
  923. Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033630, ts 3166692888, len 000170)
  924. Got RTP packet from 10.0.1.107:60479 (type 00, seq 018403, ts 1324345005, len 000160)
  925. Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041623, ts 1324345000, len 000033)
  926. asterisk*CLI>
  927. <--- SIP read from WS:10.0.1.107:51114 --->
  928. BYE sip:[email protected]:5060;transport=WS SIP/2.0
  929. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKOfenJK1FwIZ55oAdVTQgUV0XUiugfgmJ;rport
  930. From: "WEBRTC"<sip:[email protected]>;tag=4D1TKAr7hDF8WhKcmaoZ
  931. To: <sip:011525555555555@asterisk>;tag=as654c7257
  932. Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
  933. CSeq: 35988 BYE
  934. Content-Length: 0
  935. Route: <sip:10.0.1.108:5060;lr;sipml5-outbound;transport=udp>
  936. ax-Forwards: 70
  937. Authorization: Digest username="5005",realm="asterisk",nonce="774b7649",uri="sip:[email protected]:5060;transport=WS",response="db4ddf767da6689802886efbdcbdc01b",algorithm=MD5
  938. User-Agent: DM_SIPWEB-UA
  939. Organization: Digital-Merge
  940.  
  941. <------------->
  942. --- (12 headers 0 lines) ---
  943. Scheduling destruction of SIP dialog '36bb3d83-bb35-d00f-829f-4180a329a7fa' in 8000 ms (Method: BYE)
  944. asterisk*CLI>
  945. <--- Transmitting (NAT) to 10.0.1.107:51114 --->
  946. SIP/2.0 200 OK
  947. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKOfenJK1FwIZ55oAdVTQgUV0XUiugfgmJ;received=10.0.1.107;rport=51114
  948. From: "WEBRTC"<sip:[email protected]>;tag=4D1TKAr7hDF8WhKcmaoZ
  949. To: <sip:011525555555555@asterisk>;tag=as654c7257
  950. Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
  951. CSeq: 35988 BYE
  952. Server: Digital-Merge_UA
  953. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  954. Supported: replaces, timer
  955. Content-Length: 0
  956.  
  957.  
  958. <------------>
  959. asterisk*CLI>
  960. Scheduling destruction of SIP dialog '[email protected]:5060' in 6400 ms (Method: INVITE)
  961. asterisk*CLI>
  962. set_destination: Parsing <sip:[email protected]> for address/port to send to
  963. asterisk*CLI>
  964. set_destination: set destination to 74.54.XXX.XXX:5060
  965. Reliably Transmitting (NAT) to 74.54.XXX.XXX:5060:
  966. BYE sip:[email protected] SIP/2.0
  967. Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK49190ac8;rport
  968. ax-Forwards: 70
  969. From: "WebRTC" <sip:[email protected]>;tag=as613a93d5
  970. To: <sip:[email protected]>;tag=as15352569
  971. Call-ID: [email protected]:5060
  972. CSeq: 104 BYE
  973. User-Agent: Digital-Merge_UA
  974. Proxy-Authorization: Digest username="143987", realm="dallas.voip.ms", algorithm=MD5, uri="sip:[email protected]", nonce="795f186d", response="74894d08f01c5106099e34bfbf63666a"
  975. X-Asterisk-HangupCause: Normal Clearing
  976. X-Asterisk-HangupCauseCode: 16
  977. Content-Length: 0
  978.  
  979.  
  980. ---
  981. == Spawn extension (wrtc, 011525555555555, 1) exited non-zero on 'SIP/5005-0000000d'
  982. asterisk*CLI>
  983. <--- SIP read from UDP:74.54.XXX.XXX:5060 --->
  984. SIP/2.0 200 OK
  985. Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK49190ac8;received=189.241.6.199;rport=13027
  986. From: "WebRTC" <sip:[email protected]:13027>;tag=as613a93d5
  987. To: <sip:[email protected]>;tag=as15352569
  988. Call-ID: [email protected]:5060
  989. CSeq: 104 BYE
  990. User-Agent: voip.ms
  991. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  992. Supported: replaces
  993. Content-Length: 0
  994.  
  995. <------------->
  996. --- (10 headers 0 lines) ---
  997. asterisk*CLI>
  998. Really destroying SIP dialog '[email protected]:5060' Method: INVITE
  999. asterisk*CLI>
  1000. Reliably Transmitting (NAT) to 74.54.XXX.XXX:5060:
  1001. OPTIONS sip:dallas.voip.ms SIP/2.0
  1002. Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK77f75df0;rport
  1003. ax-Forwards: 70
  1004. From: "asterisk" <sip:[email protected]>;tag=as05dccb4b
  1005. To: <sip:dallas.voip.ms>
  1006. Contact: <sip:[email protected]:5060>
  1007. Call-ID: [email protected]:5060
  1008. CSeq: 102 OPTIONS
  1009. User-Agent: Digital-Merge_UA
  1010. Date: Sat, 02 Aug 2014 17:59:04 GMT
  1011. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  1012. Supported: replaces, timer
  1013. Content-Length: 0
  1014.  
  1015.  
  1016. ---
  1017. asterisk*CLI>
  1018. <--- SIP read from UDP:74.54.XXX.XXX:5060 --->
  1019. SIP/2.0 200 OK
  1020. Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK77f75df0;received=189.241.6.199;rport=13027
  1021. From: "asterisk" <sip:[email protected]>;tag=as05dccb4b
  1022. To: <sip:dallas.voip.ms>;tag=as167f2ced
  1023. Call-ID: [email protected]:5060
  1024. CSeq: 102 OPTIONS
  1025. User-Agent: voip.ms
  1026. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  1027. Supported: replaces
  1028. Contact: <sip:74.54.XXX.XXX>
  1029. Accept: application/sdp
  1030. Content-Length: 0
  1031.  
  1032. <------------->
  1033. asterisk*CLI>
  1034. --- (12 headers 0 lines) ---
  1035. asterisk*CLI>
  1036. Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
  1037. asterisk*CLI>
  1038. Reliably Transmitting (NAT) to 10.0.1.107:51114:
  1039. OPTIONS sip:[email protected];rtcweb-breaker=no;transport=ws SIP/2.0
  1040. Via: SIP/2.0/WS 10.0.1.108:5060;branch=z9hG4bK09ea7b59;rport
  1041. ax-Forwards: 70
  1042. From: "asterisk" <sip:[email protected]>;tag=as7d80fe23
  1043. To: <sip:[email protected];rtcweb-breaker=no;transport=ws>
  1044. Contact: <sip:[email protected]:5060;transport=WS>
  1045. Call-ID: [email protected]:5060
  1046. CSeq: 102 OPTIONS
  1047. User-Agent: Digital-Merge_UA
  1048. Date: Sat, 02 Aug 2014 17:59:04 GMT
  1049. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  1050. Supported: replaces, timer
  1051. Content-Length: 0
  1052.  
  1053.  
  1054. ---
  1055. asterisk*CLI>
  1056. <--- SIP read from WS:10.0.1.107:51114 --->
  1057. SIP/2.0 405 Method Not Allowed
  1058. Via: SIP/2.0/WS 10.0.1.108:5060;rport=5060;branch=z9hG4bK09ea7b59
  1059. From: "asterisk"<sip:[email protected]>;tag=as7d80fe23
  1060. To: <sip:[email protected];rtcweb-breaker=no;transport=ws>
  1061. Call-ID: [email protected]:5060
  1062. CSeq: 102 OPTIONS
  1063. Content-Length: 0
  1064.  
  1065. <------------->
  1066. asterisk*CLI>
  1067. --- (7 headers 0 lines) ---
  1068. asterisk*CLI>
  1069. Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
  1070. asterisk*CLI>
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