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  1. Incoming call log:
  2.  
  3. <--- SIP read from UDP:127.0.0.1:21676 --->
  4.  
  5.  
  6. <------------->
  7. Reliably Transmitting (NAT) to 127.0.0.1:27651:
  8. OPTIONS sip:[email protected];rtcweb-breaker=yes;transport=ws SIP/2.0
  9. Via: SIP/2.0/WS 192.168.1.31:5060;branch=z9hG4bK07c196d1;rport
  10. Max-Forwards: 70
  11. From: "Unknown" <sip:[email protected]>;tag=as3dfd1271
  12. To: <sip:[email protected];rtcweb-breaker=yes;transport=ws>
  13. Contact: <sip:[email protected]:5060;transport=WS>
  14. Call-ID: [email protected]:5060
  15. CSeq: 102 OPTIONS
  16. User-Agent: FPBX-2.11.0(11.9.0)
  17. Date: Fri, 09 May 2014 11:33:37 GMT
  18. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  19. Supported: replaces, timer
  20. Content-Length: 0
  21.  
  22.  
  23. ---
  24.  
  25. <--- SIP read from WS:127.0.0.1:27651 --->
  26. SIP/2.0 405 Method Not Allowed
  27. Via: SIP/2.0/WS 192.168.1.31:5060;rport=5060;branch=z9hG4bK07c196d1
  28. From: "Unknown"<sip:[email protected]>;tag=as3dfd1271
  29. To: <sip:[email protected];rtcweb-breaker=yes;transport=ws>
  30. Call-ID: [email protected]:5060
  31. CSeq: 102 OPTIONS
  32. Content-Length: 0
  33.  
  34. <------------->
  35. --- (7 headers 0 lines) ---
  36. Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
  37.  
  38. <--- SIP read from WS:127.0.0.1:27651 --->
  39. INVITE sip:[email protected] SIP/2.0
  40. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKQTlZ37WqtV60kINfo5kHdZL910haQrCQ;rport
  41. From: "8003"<sip:[email protected]>;tag=W8Z2awaPx3roAhyDsCMM
  42. To: <sip:[email protected]>
  43. Contact: "8003"<sip:[email protected];rtcweb-breaker=yes;click2call=no;transport=ws>;impi=8003;ha1=8cdf0cadb0d151b62cc3228649489eea;+g.oma.sip-im;+sip.ice;language="en,fr"
  44. Call-ID: 41a34fad-bd29-d73c-b2dd-98c311f27718
  45. CSeq: 39199 INVITE
  46. Content-Type: application/sdp
  47. Content-Length: 1848
  48. Max-Forwards: 70
  49. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.03.26
  50. Organization: Doubango Telecom
  51.  
  52. v=0
  53. o=- 8787614679018738000 2 IN IP4 127.0.0.1
  54. s=Doubango Telecom - chrome
  55. t=0 0
  56. a=group:BUNDLE audio
  57. a=msid-semantic: WMS rS2gAvkonr7vsOl3BhhTbP5vIrSRhiraoDlX
  58. m=audio 28346 RTP/SAVPF 111 103 104 0 8 106 105 13 126
  59. c=IN IP4 127.0.0.1
  60. a=rtcp:28346 IN IP4 127.0.0.1
  61. a=candidate:1002984303 1 udp 2122260223 192.168.1.24 53996 typ host generation 0
  62. a=candidate:1002984303 2 udp 2122260223 192.168.1.24 53996 typ host generation 0
  63. a=candidate:1967894431 1 tcp 1518280447 192.168.1.24 0 typ host generation 0
  64. a=candidate:1967894431 2 tcp 1518280447 192.168.1.24 0 typ host generation 0
  65. a=candidate:3453495995 1 udp 1686052607 127.0.0.1 28346 typ srflx raddr 192.168.1.24 rport 53996 generation 0
  66. a=candidate:3453495995 2 udp 1686052607 127.0.0.1 28346 typ srflx raddr 192.168.1.24 rport 53996 generation 0
  67. a=ice-ufrag:cvZVd48YygDnmdyO
  68. a=ice-pwd:XSxdmNfryl15IbVmP49r9kiV
  69. a=ice-options:google-ice
  70. a=fingerprint:sha-256 CF:B5:45:74:E1:39:75:92:57:C1:11:DD:1F:5C:94:29:7A:82:90:7B:73:A3:5A:2E:FC:E7:8E:E0:B8:33:EF:CE
  71. a=setup:actpass
  72. a=mid:audio
  73. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  74. a=sendrecv
  75. a=rtcp-mux
  76. a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:3cwayv4FIfgPQlaaxkncTypCdQy+TpQbzIxZ49H/
  77. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:QZnalBUkSq8mXKMf7yyJQukSJPB4lWpUm9+b3smW
  78. a=rtpmap:111 opus/48000/2
  79. a=fmtp:111 minptime=10
  80. a=rtpmap:103 ISAC/16000
  81. a=rtpmap:104 ISAC/32000
  82. a=rtpmap:0 PCMU/8000
  83. a=rtpmap:8 PCMA/8000
  84. a=rtpmap:106 CN/32000
  85. a=rtpmap:105 CN/16000
  86. a=rtpmap:13 CN/8000
  87. a=rtpmap:126 telephone-event/8000
  88. a=maxptime:60
  89. a=ssrc:2083165809 cname:DznWlZqxQpEnXiEi
  90. a=ssrc:2083165809 msid:rS2gAvkonr7vsOl3BhhTbP5vIrSRhiraoDlX e9603b5b-44c7-46f3-b6ec-e5f0676b617c
  91. a=ssrc:2083165809 mslabel:rS2gAvkonr7vsOl3BhhTbP5vIrSRhiraoDlX
  92. a=ssrc:2083165809 label:e9603b5b-44c7-46f3-b6ec-e5f0676b617c
  93. <------------->
  94. --- (12 headers 41 lines) ---
  95. Using INVITE request as basis request - 41a34fad-bd29-d73c-b2dd-98c311f27718
  96. Found peer '8003' for '8003' from 127.0.0.1:27651
  97.  
  98. <--- Reliably Transmitting (NAT) to 127.0.0.1:27651 --->
  99. SIP/2.0 401 Unauthorized
  100. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKQTlZ37WqtV60kINfo5kHdZL910haQrCQ;received=127.0.0.1;rport=27651
  101. From: "8003"<sip:[email protected]>;tag=W8Z2awaPx3roAhyDsCMM
  102. To: <sip:[email protected]>;tag=as766d8b89
  103. Call-ID: 41a34fad-bd29-d73c-b2dd-98c311f27718
  104. CSeq: 39199 INVITE
  105. Server: FPBX-2.11.0(11.9.0)
  106. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  107. Supported: replaces, timer
  108. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="166fee3d"
  109. Content-Length: 0
  110.  
  111.  
  112. <------------>
  113. Scheduling destruction of SIP dialog '41a34fad-bd29-d73c-b2dd-98c311f27718' in 54464 ms (Method: INVITE)
  114.  
  115. <--- SIP read from WS:127.0.0.1:27651 --->
  116. ACK sip:[email protected] SIP/2.0
  117. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKQTlZ37WqtV60kINfo5kHdZL910haQrCQ;rport
  118. From: "8003"<sip:[email protected]>;tag=W8Z2awaPx3roAhyDsCMM
  119. To: <sip:[email protected]>;tag=as766d8b89
  120. Call-ID: 41a34fad-bd29-d73c-b2dd-98c311f27718
  121. CSeq: 39199 ACK
  122. Content-Length: 0
  123. Max-Forwards: 70
  124.  
  125. <------------->
  126. --- (8 headers 0 lines) ---
  127.  
  128. <--- SIP read from WS:127.0.0.1:27651 --->
  129. INVITE sip:[email protected] SIP/2.0
  130. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK1b6Z7RHWemf4eF92dxqvsJUPW6qVKvm6;rport
  131. From: "8003"<sip:[email protected]>;tag=W8Z2awaPx3roAhyDsCMM
  132. To: <sip:[email protected]>
  133. Contact: "8003"<sip:[email protected];rtcweb-breaker=yes;click2call=no;transport=ws>;impi=8003;ha1=8cdf0cadb0d151b62cc3228649489eea;+g.oma.sip-im;+sip.ice;language="en,fr"
  134. Call-ID: 41a34fad-bd29-d73c-b2dd-98c311f27718
  135. CSeq: 39200 INVITE
  136. Content-Type: application/sdp
  137. Content-Length: 1848
  138. Max-Forwards: 70
  139. Authorization: Digest username="8003",realm="asterisk",nonce="166fee3d",uri="sip:[email protected]",response="3b0542bae8d2e3437dfbbe2075ffcfdc",algorithm=MD5
  140. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.03.26
  141. Organization: Doubango Telecom
  142.  
  143. v=0
  144. o=- 8787614679018738000 2 IN IP4 127.0.0.1
  145. s=Doubango Telecom - chrome
  146. t=0 0
  147. a=group:BUNDLE audio
  148. a=msid-semantic: WMS rS2gAvkonr7vsOl3BhhTbP5vIrSRhiraoDlX
  149. m=audio 28346 RTP/SAVPF 111 103 104 0 8 106 105 13 126
  150. c=IN IP4 127.0.0.1
  151. a=rtcp:28346 IN IP4 127.0.0.1
  152. a=candidate:1002984303 1 udp 2122260223 192.168.1.24 53996 typ host generation 0
  153. a=candidate:1002984303 2 udp 2122260223 192.168.1.24 53996 typ host generation 0
  154. a=candidate:1967894431 1 tcp 1518280447 192.168.1.24 0 typ host generation 0
  155. a=candidate:1967894431 2 tcp 1518280447 192.168.1.24 0 typ host generation 0
  156. a=candidate:3453495995 1 udp 1686052607 127.0.0.1 28346 typ srflx raddr 192.168.1.24 rport 53996 generation 0
  157. a=candidate:3453495995 2 udp 1686052607 127.0.0.1 28346 typ srflx raddr 192.168.1.24 rport 53996 generation 0
  158. a=ice-ufrag:cvZVd48YygDnmdyO
  159. a=ice-pwd:XSxdmNfryl15IbVmP49r9kiV
  160. a=ice-options:google-ice
  161. a=fingerprint:sha-256 CF:B5:45:74:E1:39:75:92:57:C1:11:DD:1F:5C:94:29:7A:82:90:7B:73:A3:5A:2E:FC:E7:8E:E0:B8:33:EF:CE
  162. a=setup:actpass
  163. a=mid:audio
  164. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  165. a=sendrecv
  166. a=rtcp-mux
  167. a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:3cwayv4FIfgPQlaaxkncTypCdQy+TpQbzIxZ49H/
  168. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:QZnalBUkSq8mXKMf7yyJQukSJPB4lWpUm9+b3smW
  169. a=rtpmap:111 opus/48000/2
  170. a=fmtp:111 minptime=10
  171. a=rtpmap:103 ISAC/16000
  172. a=rtpmap:104 ISAC/32000
  173. a=rtpmap:0 PCMU/8000
  174. a=rtpmap:8 PCMA/8000
  175. a=rtpmap:106 CN/32000
  176. a=rtpmap:105 CN/16000
  177. a=rtpmap:13 CN/8000
  178. a=rtpmap:126 telephone-event/8000
  179. a=maxptime:60
  180. a=ssrc:2083165809 cname:DznWlZqxQpEnXiEi
  181. a=ssrc:2083165809 msid:rS2gAvkonr7vsOl3BhhTbP5vIrSRhiraoDlX e9603b5b-44c7-46f3-b6ec-e5f0676b617c
  182. a=ssrc:2083165809 mslabel:rS2gAvkonr7vsOl3BhhTbP5vIrSRhiraoDlX
  183. a=ssrc:2083165809 label:e9603b5b-44c7-46f3-b6ec-e5f0676b617c
  184. <------------->
  185. --- (13 headers 41 lines) ---
  186. Using INVITE request as basis request - 41a34fad-bd29-d73c-b2dd-98c311f27718
  187. Found peer '8003' for '8003' from 127.0.0.1:27651
  188. == Using SIP RTP TOS bits 184
  189. == Using SIP RTP CoS mark 5
  190. Found RTP audio format 111
  191. Found RTP audio format 103
  192. Found RTP audio format 104
  193. Found RTP audio format 0
  194. Found RTP audio format 8
  195. Found RTP audio format 106
  196. Found RTP audio format 105
  197. Found RTP audio format 13
  198. Found RTP audio format 126
  199. Found unknown media description format opus for ID 111
  200. Found unknown media description format ISAC for ID 103
  201. Found unknown media description format ISAC for ID 104
  202. Found audio description format PCMU for ID 0
  203. Found audio description format PCMA for ID 8
  204. Found unknown media description format CN for ID 106
  205. Found unknown media description format CN for ID 105
  206. Found audio description format CN for ID 13
  207. Found audio description format telephone-event for ID 126
  208. Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  209. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
  210. Peer audio RTP is at port 127.0.0.1:28346
  211. Looking for 8002 in sipml5 (domain 192.168.1.31)
  212. list_route: hop: <sip:[email protected];rtcweb-breaker=yes;click2call=no;transport=ws>
  213.  
  214. <--- Transmitting (NAT) to 127.0.0.1:27651 --->
  215. SIP/2.0 100 Trying
  216. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK1b6Z7RHWemf4eF92dxqvsJUPW6qVKvm6;received=127.0.0.1;rport=27651
  217. From: "8003"<sip:[email protected]>;tag=W8Z2awaPx3roAhyDsCMM
  218. To: <sip:[email protected]>
  219. Call-ID: 41a34fad-bd29-d73c-b2dd-98c311f27718
  220. CSeq: 39200 INVITE
  221. Server: FPBX-2.11.0(11.9.0)
  222. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  223. Supported: replaces, timer
  224. Contact: <sip:[email protected]:5060;transport=WS>
  225. Content-Length: 0
  226.  
  227.  
  228. <------------>
  229. -- Executing [8002@sipml5:1] Dial("SIP/8003-00000013", "SIP/8002") in new stack
  230. == Using SIP RTP TOS bits 184
  231. == Using SIP RTP CoS mark 5
  232. We think we can do text
  233. Audio is at 13616
  234. Adding codec 100003 (ulaw) to SDP
  235. Adding codec 100004 (alaw) to SDP
  236. Adding codec 100002 (gsm) to SDP
  237. Adding codec 100001 (g723) to SDP
  238. Adding codec 100005 (g726aal2) to SDP
  239. Adding codec 100006 (adpcm) to SDP
  240. Adding codec 100007 (lpc10) to SDP
  241. Adding codec 100008 (g729) to SDP
  242. Adding codec 100009 (speex) to SDP
  243. Adding codec 100010 (ilbc) to SDP
  244. Adding codec 100011 (g726) to SDP
  245. Adding codec 100012 (g722) to SDP
  246. Adding codec 100013 (siren7) to SDP
  247. Adding codec 100014 (siren14) to SDP
  248. Adding codec 100015 (g719) to SDP
  249. Adding codec 100016 (speex16) to SDP
  250. Adding codec 100017 (testlaw) to SDP
  251. Adding codec 100019 (slin) to SDP
  252. Adding codec 100020 (slin12) to SDP
  253. Adding codec 100021 (slin16) to SDP
  254. Adding codec 100022 (slin24) to SDP
  255. Adding codec 100023 (slin32) to SDP
  256. Adding codec 100024 (slin44) to SDP
  257. Adding codec 100025 (slin48) to SDP
  258. Adding codec 100026 (slin96) to SDP
  259. Adding codec 100027 (slin192) to SDP
  260. Adding codec 100028 (speex32) to SDP
  261. Adding non-codec 0x1 (telephone-event) to SDP
  262. Reliably Transmitting (NAT) to 127.0.0.1:27817:
  263. INVITE sip:[email protected];rtcweb-breaker=yes;transport=ws SIP/2.0
  264. Via: SIP/2.0/WS 192.168.1.31:5060;branch=z9hG4bK3fadc9d0;rport
  265. Max-Forwards: 70
  266. From: "8003" <sip:[email protected]>;tag=as2a13b85f
  267. To: <sip:[email protected];rtcweb-breaker=yes;transport=ws>
  268. Contact: <sip:[email protected]:5060;transport=WS>
  269. Call-ID: [email protected]:5060
  270. CSeq: 102 INVITE
  271. User-Agent: FPBX-2.11.0(11.9.0)
  272. Date: Fri, 09 May 2014 11:33:41 GMT
  273. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  274. Supported: replaces, timer
  275. Content-Type: application/sdp
  276. Content-Length: 954
  277.  
  278. v=0
  279. o=root 816536178 816536178 IN IP4 192.168.1.31
  280. s=Asterisk PBX 11.9.0
  281. c=IN IP4 192.168.1.31
  282. t=0 0
  283. m=audio 13616 RTP/SAVP 0 8 3 4 112 5 7 18 110 97 111 9 102 115 116 117 10 118 119 101
  284. a=rtpmap:0 PCMU/8000
  285. a=rtpmap:8 PCMA/8000
  286. a=rtpmap:3 GSM/8000
  287. a=rtpmap:4 G723/8000
  288. a=fmtp:4 annexa=no
  289. a=rtpmap:112 AAL2-G726-32/8000
  290. a=rtpmap:5 DVI4/8000
  291. a=rtpmap:7 LPC/8000
  292. a=rtpmap:18 G729/8000
  293. a=fmtp:18 annexb=no
  294. a=rtpmap:110 speex/8000
  295. a=rtpmap:97 iLBC/8000
  296. a=fmtp:97 mode=30
  297. a=rtpmap:111 G726-32/8000
  298. a=rtpmap:9 G722/8000
  299. a=rtpmap:102 G7221/16000
  300. a=fmtp:102 bitrate=32000
  301. a=rtpmap:115 G7221/32000
  302. a=fmtp:115 bitrate=48000
  303. a=rtpmap:116 G719/48000
  304. a=fmtp:116 bitrate=64000
  305. a=rtpmap:117 speex/16000
  306. a=rtpmap:10 L16/8000
  307. a=rtpmap:118 L16/16000
  308. a=rtpmap:119 speex/32000
  309. a=rtpmap:101 telephone-event/8000
  310. a=fmtp:101 0-16
  311. a=ptime:20
  312. a=sendrecv
  313. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:keyjtt34IDaP/2whKDYzjz7Mck4ryU6KzLgwiwD7
  314.  
  315. ---
  316. -- Called SIP/8002
  317.  
  318. <--- SIP read from WS:127.0.0.1:27817 --->
  319. SIP/2.0 100 Trying (sent from the Transaction Layer)
  320. Via: SIP/2.0/WS 192.168.1.31:5060;rport=5060;branch=z9hG4bK3fadc9d0
  321. From: "8003"<sip:[email protected]>;tag=as2a13b85f
  322. To: <sip:[email protected];rtcweb-breaker=yes;transport=ws>
  323. Call-ID: [email protected]:5060
  324. CSeq: 102 INVITE
  325. Content-Length: 0
  326.  
  327. <------------->
  328. --- (7 headers 0 lines) ---
  329.  
  330. <--- SIP read from WS:127.0.0.1:27817 --->
  331. SIP/2.0 180 Ringing
  332. Via: SIP/2.0/WS 192.168.1.31:5060;rport=5060;branch=z9hG4bK3fadc9d0
  333. From: "8003"<sip:[email protected]>;tag=as2a13b85f
  334. To: <sip:[email protected];rtcweb-breaker=yes;transport=ws>;tag=OfSWDTO99tlHmwEo1S8u
  335. Contact: <sip:[email protected];transport=ws>
  336. Call-ID: [email protected]:5060
  337. CSeq: 102 INVITE
  338. Content-Length: 0
  339. Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
  340.  
  341. <------------->
  342. --- (9 headers 0 lines) ---
  343. list_route: hop: <sip:[email protected];transport=ws>
  344. -- SIP/8002-00000014 is ringing
  345.  
  346. <--- Transmitting (NAT) to 127.0.0.1:27651 --->
  347. SIP/2.0 180 Ringing
  348. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK1b6Z7RHWemf4eF92dxqvsJUPW6qVKvm6;received=127.0.0.1;rport=27651
  349. From: "8003"<sip:[email protected]>;tag=W8Z2awaPx3roAhyDsCMM
  350. To: <sip:[email protected]>;tag=as05816afb
  351. Call-ID: 41a34fad-bd29-d73c-b2dd-98c311f27718
  352. CSeq: 39200 INVITE
  353. Server: FPBX-2.11.0(11.9.0)
  354. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  355. Supported: replaces, timer
  356. Contact: <sip:[email protected]:5060;transport=WS>
  357. Content-Length: 0
  358.  
  359.  
  360. <------------>
  361. Reliably Transmitting (NAT) to 127.0.0.1:26698:
  362. OPTIONS sip:[email protected];rtcweb-breaker=yes;transport=ws SIP/2.0
  363. Via: SIP/2.0/WS 192.168.1.31:5060;branch=z9hG4bK4be7f694;rport
  364. Max-Forwards: 70
  365. From: "Unknown" <sip:[email protected]>;tag=as622e065d
  366. To: <sip:[email protected];rtcweb-breaker=yes;transport=ws>
  367. Contact: <sip:[email protected]:5060;transport=WS>
  368. Call-ID: [email protected]:5060
  369. CSeq: 102 OPTIONS
  370. User-Agent: FPBX-2.11.0(11.9.0)
  371. Date: Fri, 09 May 2014 11:33:42 GMT
  372. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  373. Supported: replaces, timer
  374. Content-Length: 0
  375.  
  376.  
  377. ---
  378.  
  379. <--- SIP read from WS:127.0.0.1:26698 --->
  380. SIP/2.0 405 Method Not Allowed
  381. Via: SIP/2.0/WS 192.168.1.31:5060;rport=5060;branch=z9hG4bK4be7f694
  382. From: "Unknown"<sip:[email protected]>;tag=as622e065d
  383. To: <sip:[email protected];rtcweb-breaker=yes;transport=ws>
  384. Call-ID: [email protected]:5060
  385. CSeq: 102 OPTIONS
  386. Content-Length: 0
  387.  
  388. <------------->
  389. --- (7 headers 0 lines) ---
  390. Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
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