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Incoming call log: 
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<--- 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>
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Contact: <sip:[email protected]:5060;transport=WS>
14
Call-ID: [email protected]:5060
15
CSeq: 102 OPTIONS
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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
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38
<--- SIP read from WS:127.0.0.1:27651 --->
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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]>
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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
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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
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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 webrtc (domain 192.168.1.31)
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@webrtc:1] Dial("SIP/8003-00000013", "SIP/8002") in new stack
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
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Supported: replaces, timer
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Content-Length: 0
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---
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<--- SIP read from WS:127.0.0.1:26698 --->
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SIP/2.0 405 Method Not Allowed
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Via: SIP/2.0/WS 192.168.1.31:5060;rport=5060;branch=z9hG4bK4be7f694
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From: "Unknown"<sip:[email protected]>;tag=as622e065d
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To: <sip:[email protected];rtcweb-breaker=yes;transport=ws>
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Call-ID: [email protected]:5060
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CSeq: 102 OPTIONS
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Content-Length: 0
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<------------->
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--- (7 headers 0 lines) ---
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Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS