INVITE sip:6001@domain.com SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKwBZaRIGi5yRoZqSlUzK9Ak8HM62PG2sr;rport
From: "Web"<sip:6000@domain.com>;tag=nTDEGTooi0OLRl3vlVGd
To: <sip:6001@domain.com>
Contact: "Web"<sips:6000@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
Call-ID: f553434d-85ad-fe14-5c68-74ac32ea92a3
CSeq: 16258 INVITE
Content-Type: application/sdp
Content-Length: 2326
Route: <sip:domain.com:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="6000",realm="52.247.186.5",nonce="74afd34b",uri="sip:6001@domain.com",response="b7e019457d904e61f783a5e840e59f0d",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom

v=0
o=- 1649533246144286700 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS Z2udPeP3ySbyliTTsvRGmH8fJOzZt8TzIbqA
m=audio 51346 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 42.109.165.150
a=rtcp:33289 IN IP4 42.109.165.150
a=candidate:2894319779 1 udp 2122260223 192.168.1.52 60553 typ host generation 0 network-id 2
a=candidate:183819031 1 udp 2122194687 192.168.0.52 51346 typ host generation 0 network-id 1
a=candidate:2894319779 2 udp 2122260222 192.168.1.52 55600 typ host generation 0 network-id 2
a=candidate:183819031 2 udp 2122194686 192.168.0.52 33289 typ host generation 0 network-id 1
a=candidate:4242842307 1 udp 1685987071 42.109.165.150 51346 typ srflx raddr 192.168.0.52 rport 51346 generation 0 network-id 1
a=candidate:3791662163 1 tcp 1518280447 192.168.1.52 9 typ host tcptype active generation 0 network-id 2
a=candidate:1148659687 1 tcp 1518214911 192.168.0.52 9 typ host tcptype active generation 0 network-id 1
a=candidate:3791662163 2 tcp 1518280446 192.168.1.52 9 typ host tcptype active generation 0 network-id 2
a=candidate:1148659687 2 tcp 1518214910 192.168.0.52 9 typ host tcptype active generation 0 network-id 1
a=candidate:4242842307 2 udp 1685987070 42.109.165.150 33289 typ srflx raddr 192.168.0.52 rport 33289 generation 0 network-id 1
a=ice-ufrag:AFUO
a=ice-pwd:wAZ+z9G3R4H56HZbf/nRuv8Z
a=fingerprint:sha-256 14:38:1D:B5:0D:B7:93:C9:6B:55:71:44:F2:F1:91:42:80:02:5B:09:B1:C2:FA:5C:B2:78:7E:5B:43:63:F1:FA
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:3715592214 cname:PEYeldeIUHYrdyeR
a=ssrc:3715592214 msid:Z2udPeP3ySbyliTTsvRGmH8fJOzZt8TzIbqA f4086808-636d-4b36-b509-9356a7bd4774
a=ssrc:3715592214 mslabel:Z2udPeP3ySbyliTTsvRGmH8fJOzZt8TzIbqA
a=ssrc:3715592214 label:f4086808-636d-4b36-b509-9356a7bd4774
<------------->
--- (14 headers 46 lines) ---
Using INVITE request as basis request - f553434d-85ad-fe14-5c68-74ac32ea92a3
Found peer '6000' for '6000' from 42.109.165.150:53831
  == DTLS ECDH initialized (secp256r1), faster PFS enabled
  == Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 110
Found RTP audio format 112
Found RTP audio format 113
Found RTP audio format 126
Found audio description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found unknown media description format telephone-event for ID 110
Found unknown media description format telephone-event for ID 112
Found unknown media description format telephone-event for ID 113
Found audio description format telephone-event for ID 126
Capabilities: us - (ulaw), peer - audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 42.109.165.150:51346
Looking for 6001 in wrtc (domain domain.com)
sip_route_dump: route/path hop: <sips:6000@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>

<--- Transmitting (NAT) to 42.109.165.150:53831 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKwBZaRIGi5yRoZqSlUzK9Ak8HM62PG2sr;received=42.109.165.150;rport=53831
From: "Web"<sip:6000@domain.com>;tag=nTDEGTooi0OLRl3vlVGd
To: <sip:6001@domain.com>
Call-ID: f553434d-85ad-fe14-5c68-74ac32ea92a3
CSeq: 16258 INVITE
Server: Asterisk PBX 13.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:6001@52.247.186.5:0;transport=ws>
Content-Length: 0


<------------>
    -- Executing [6001@wrtc:1] Dial("SIP/6000-000000b8", "SIP/6001") in new stack
  == DTLS ECDH initialized (secp256r1), faster PFS enabled
  == Using SIP RTP CoS mark 5
Audio is at 17470
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 42.109.165.150:50859:
INVITE sips:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss SIP/2.0
Via: SIP/2.0/WS 52.247.186.5:0;branch=z9hG4bK457721d5;rport
Max-Forwards: 70
From: "Web" <sip:6000@52.247.186.5:0>;tag=as71b1a88c
To: <sips:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss>
Contact: <sip:6000@52.247.186.5:0;transport=ws>
Call-ID: 184014a70f307bdd6d0df8b47969b55a@52.247.186.5:0
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.15.0
Date: Fri, 28 Apr 2017 06:33:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 836

v=0
o=root 740926397 740926397 IN IP4 52.247.186.5
s=Asterisk PBX 13.15.0
c=IN IP4 52.247.186.5
t=0 0
m=audio 17470 RTP/SAVPF 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=ice-ufrag:5fa824db420b8f152ebd312573e43287
a=ice-pwd:544965980aa7b45e6a7fe8a936804c63
a=candidate:Hac1f225d 1 UDP 2130706431 172.31.34.93 17470 typ host
a=candidate:S34219b9a 1 UDP 1694498815 52.247.186.5 17470 typ srflx raddr 172.31.34.93 rport 17470
a=candidate:Hac1f225d 2 UDP 2130706430 172.31.34.93 17471 typ host
a=candidate:S34219b9a 2 UDP 1694498814 52.247.186.5 17471 typ srflx raddr 172.31.34.93 rport 17471
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 27:1D:7D:89:95:7D:33:82:36:23:29:3B:13:3C:94:21:19:D4:B8:B8:C0:CF:F6:49:41:45:4F:0E:B6:46:8B:5C
a=rtcp-mux
a=sendrecv

---
    -- Called SIP/6001

<--- SIP read from WS:42.109.165.150:50859 --->
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS 52.247.186.5;rport;branch=z9hG4bK457721d5
From: "Web"<sip:6000@52.247.186.5>;tag=as71b1a88c
To: <sips:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss>
Call-ID: 184014a70f307bdd6d0df8b47969b55a@52.247.186.5:0
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from WS:42.109.165.150:50859 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 52.247.186.5;rport;branch=z9hG4bK457721d5
From: "Web"<sip:6000@52.247.186.5>;tag=as71b1a88c
To: <sips:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss>;tag=2baeFQ2f9iO9rdRiURNr
Contact: <sips:6001@df7jal23ls0d.invalid;transport=wss>
Call-ID: 184014a70f307bdd6d0df8b47969b55a@52.247.186.5:0
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

<------------->
--- (9 headers 0 lines) ---
sip_route_dump: route/path hop: <sips:6001@df7jal23ls0d.invalid;transport=wss>
    -- SIP/6001-000000b9 is ringing

<--- Transmitting (NAT) to 42.109.165.150:53831 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKwBZaRIGi5yRoZqSlUzK9Ak8HM62PG2sr;received=42.109.165.150;rport=53831
From: "Web"<sip:6000@domain.com>;tag=nTDEGTooi0OLRl3vlVGd
To: <sip:6001@domain.com>;tag=as55400d63
Call-ID: f553434d-85ad-fe14-5c68-74ac32ea92a3
CSeq: 16258 INVITE
Server: Asterisk PBX 13.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:6001@52.247.186.5:0;transport=ws>
Content-Length: 0


<------------>


<------------->
Really destroying SIP dialog 'TdbzOTKjpoC3elYQ4v1qnA..' Method: REGISTER
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK5dwJ1BbXPQyZ78Cva4X6yu047b2NoGSM;rport
From: "WebRtc"<sip:6001@domain.com>;tag=v0x23MZqbrlcrPOZWc3l
To: <sip:6000@domain.com>;tag=as0713802c
Contact: "WebRtc"<sips:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
Call-ID: 6cfc28f3-bbc4-b2dd-0834-e0ce2310dc92
CSeq: 640 INVITE
Content-Type: application/sdp
Content-Length: 2484
Route: <sip:domain.com:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="6001",realm="52.247.186.5",nonce="462ca2f5",uri="sip:6000@52.247.186.5;transport=ws",response="8ebe77a044553e0500747a148c773649",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom

v=0
o=- 1843232178561686000 4 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS jzp7S9q0xhcSFJfLLGtlSzPMcrrGnkBzBcPL
m=audio 53935 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 42.109.165.150
a=rtcp:53937 IN IP4 42.109.165.150
a=candidate:4043973916 1 udp 2122255103 2001::9d38:6abd:20e2:d76:3f57:ff81 53934 typ host generation 0 network-id 2 network-cost 50
a=candidate:547636346 1 udp 2122194687 192.168.0.126 53935 typ host generation 0 network-id 1
a=candidate:4043973916 2 udp 2122255102 2001::9d38:6abd:20e2:d76:3f57:ff81 53936 typ host generation 0 network-id 2 network-cost 50
a=candidate:547636346 2 udp 2122194686 192.168.0.126 53937 typ host generation 0 network-id 1
a=candidate:3213482476 1 tcp 1518275327 2001::9d38:6abd:20e2:d76:3f57:ff81 9 typ host tcptype active generation 0 network-id 2 network-cost 50
a=candidate:1848096906 1 tcp 1518214911 192.168.0.126 9 typ host tcptype active generation 0 network-id 1
a=candidate:3213482476 2 tcp 1518275326 2001::9d38:6abd:20e2:d76:3f57:ff81 9 typ host tcptype active generation 0 network-id 2 network-cost 50
a=candidate:1848096906 2 tcp 1518214910 192.168.0.126 9 typ host tcptype active generation 0 network-id 1
a=candidate:2716615374 1 udp 1685987071 42.109.165.150 53935 typ srflx raddr 192.168.0.126 rport 53935 generation 0 network-id 1
a=candidate:2716615374 2 udp 1685987070 42.109.165.150 53937 typ srflx raddr 192.168.0.126 rport 53937 generation 0 network-id 1
a=ice-ufrag:gRVs
a=ice-pwd:/yWozvWryI/XztMf8F+4m7cF
a=fingerprint:sha-256 6C:06:28:AF:A9:A9:08:20:3E:35:EB:A0:4B:AA:1D:FC:3C:5B:FC:0C:99:EC:C3:8D:A7:60:8A:E2:2B:38:C8:91
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:3286410753 cname:LKT7L9tJzXyBcoQI
a=ssrc:3286410753 msid:jzp7S9q0xhcSFJfLLGtlSzPMcrrGnkBzBcPL 1959510b-c0fc-4250-a1dc-1561481d7910
a=ssrc:3286410753 mslabel:jzp7S9q0xhcSFJfLLGtlSzPMcrrGnkBzBcPL
a=ssrc:3286410753 label:1959510b-c0fc-4250-a1dc-1561481d7910
<------------->
--- (14 headers 46 lines) ---
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 110
Found RTP audio format 112
Found RTP audio format 113
Found RTP audio format 126
Found audio description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found unknown media description format telephone-event for ID 110
Found unknown media description format telephone-event for ID 112
Found unknown media description format telephone-event for ID 113
Found audio description format telephone-event for ID 126
Capabilities: us - (ulaw), peer - audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 42.109.165.150:53935

<--- Transmitting (NAT) to 42.109.165.150:50859 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK5dwJ1BbXPQyZ78Cva4X6yu047b2NoGSM;received=42.109.165.150;rport=50859
From: "WebRtc"<sip:6001@domain.com>;tag=v0x23MZqbrlcrPOZWc3l
To: <sip:6000@domain.com>;tag=as0713802c
Call-ID: 6cfc28f3-bbc4-b2dd-0834-e0ce2310dc92
CSeq: 640 INVITE
Server: Asterisk PBX 13.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:6000@52.247.186.5:0;transport=ws>
Content-Length: 0


<------------>
Audio is at 19550
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 42.109.165.150:50859 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK5dwJ1BbXPQyZ78Cva4X6yu047b2NoGSM;received=42.109.165.150;rport=50859
From: "WebRtc"<sip:6001@domain.com>;tag=v0x23MZqbrlcrPOZWc3l
To: <sip:6000@domain.com>;tag=as0713802c
Call-ID: 6cfc28f3-bbc4-b2dd-0834-e0ce2310dc92
CSeq: 640 INVITE
Server: Asterisk PBX 13.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:6000@52.247.186.5:0;transport=ws>
Content-Type: application/sdp
Content-Length: 837

v=0
o=root 1586083636 1586083638 IN IP4 52.247.186.5
s=Asterisk PBX 13.15.0
c=IN IP4 52.247.186.5
t=0 0
m=audio 19550 RTP/SAVPF 0 126
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=maxptime:150
a=ice-ufrag:076184ac48c5291b3ea7628908ac051c
a=ice-pwd:60a9efbe43cdfdd8063b31201652892e
a=candidate:Hac1f225d 1 UDP 2130706431 172.31.34.93 19550 typ host
a=candidate:S34219b9a 1 UDP 1694498815 52.247.186.5 19550 typ srflx raddr 172.31.34.93 rport 19550
a=candidate:Hac1f225d 2 UDP 2130706430 172.31.34.93 19551 typ host
a=candidate:S34219b9a 2 UDP 1694498814 52.247.186.5 19551 typ srflx raddr 172.31.34.93 rport 19551
a=connection:new
a=setup:active
a=fingerprint:SHA-256 27:1D:7D:89:95:7D:33:82:36:23:29:3B:13:3C:94:21:19:D4:B8:B8:C0:CF:F6:49:41:45:4F:0E:B6:46:8B:5C
a=rtcp-mux
a=sendrecv

<------------>
    -- Stopped music on hold on SIP/6000-000000bb
Really destroying SIP dialog 'TdbzOTKjpoC3elYQ4v1qnA..' Method: REGISTER

<--- SIP read from WS:42.109.165.150:50859 --->
ACK sip:6000@52.247.186.5;transport=ws SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKVKfQNLsMh6FwsxJJ1oW1;rport
From: "WebRtc"<sip:6001@domain.com>;tag=v0x23MZqbrlcrPOZWc3l
To: <sip:6000@domain.com>;tag=as0713802c
Contact: "WebRtc"<sips:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
Call-ID: 6cfc28f3-bbc4-b2dd-0834-e0ce2310dc92
CSeq: 640 ACK
Content-Length: 0
Route: <sip:domain.com:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="6001",realm="52.247.186.5",nonce="462ca2f5",uri="sip:6000@52.247.186.5;transport=ws",response="c174d7fbcd32d767cbdcb4ae8b079e10",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog 'd9d179a4-1b4f-5ce1-7206-15b3158d7a00' Method: REGISTER
Really destroying SIP dialog 'dc93579-628c61c3-5902e2cd@52.247.186.5' Method: REGISTER

<------------->

Scheduling destruction of SIP dialog 'TdbzOTKjpoC3elYQ4v1qnA..' in 32000 ms (Method: REGISTER)

<--- SIP read from WS:42.109.165.150:53831 --->
BYE sip:6001@52.247.186.5;transport=ws SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKCI7FtM7NlXPASoDow1DImvTyq8dZfEkV;rport
From: <sips:6000@df7jal23ls0d.invalid>;tag=iTs9iT6QAqQmIMeZtPbp
To: "WebRtc"<sip:6001@52.247.186.5>;tag=as2d87a795
Call-ID: 345f17390353758829e5d4e0147b5ace@52.247.186.5:0
CSeq: 34871 BYE
Content-Length: 0
Route: <sip:domain.com:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language="en,fr"
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language="en,fr"
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom

<------------->
--- (15 headers 0 lines) ---
Scheduling destruction of SIP dialog '345f17390353758829e5d4e0147b5ace@52.247.186.5:0' in 32000 ms (Method: BYE)

<--- Transmitting (NAT) to 42.109.165.150:53831 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKCI7FtM7NlXPASoDow1DImvTyq8dZfEkV;received=42.109.165.150;rport=53831
From: <sips:6000@df7jal23ls0d.invalid>;tag=iTs9iT6QAqQmIMeZtPbp
To: "WebRtc"<sip:6001@52.247.186.5>;tag=as2d87a795
Call-ID: 345f17390353758829e5d4e0147b5ace@52.247.186.5:0
CSeq: 34871 BYE
Server: Asterisk PBX 13.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Channel SIP/6000-000000bb left 'simple_bridge' basic-bridge <cbe693fe-e145-4bcd-98be-1eca556d35e7>
    -- Channel SIP/6001-000000ba left 'simple_bridge' basic-bridge <cbe693fe-e145-4bcd-98be-1eca556d35e7>
  == Spawn extension (wrtc, 6000, 1) exited non-zero on 'SIP/6001-000000ba'
Scheduling destruction of SIP dialog '6cfc28f3-bbc4-b2dd-0834-e0ce2310dc92' in 32000 ms (Method: ACK)
Reliably Transmitting (NAT) to 42.109.165.150:50859:
BYE sips:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss SIP/2.0
Via: SIP/2.0/WS 52.247.186.5:0;branch=z9hG4bK36a9a61e;rport
Max-Forwards: 70
From: <sip:6000@domain.com>;tag=as0713802c
To: "WebRtc"<sip:6001@domain.com>;tag=v0x23MZqbrlcrPOZWc3l
Call-ID: 6cfc28f3-bbc4-b2dd-0834-e0ce2310dc92
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.15.0
Proxy-Authorization: Digest username="6001", realm="52.247.186.5", algorithm=MD5, uri="sip:domain.com", nonce="462ca2f5", response="6bfcb39df27dcd736c2d1fb0486c6029"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from WS:42.109.165.150:50859 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 52.247.186.5;rport;branch=z9hG4bK36a9a61e
From: <sip:6000@domain.com>;tag=as0713802c
To: "WebRtc"<sip:6001@domain.com>;tag=v0x23MZqbrlcrPOZWc3l
Contact: <sips:6001@df7jal23ls0d.invalid;transport=wss>
Call-ID: 6cfc28f3-bbc4-b2dd-0834-e0ce2310dc92
CSeq: 102 BYE
Content-Length: 0