
<------------>
Scheduling destruction of SIP dialog 'tneuhruf1l6596af282bdr' in 32000 ms (Method: REGISTER)

<--- SIP read from WS:186.XX.XX.XX:60753 --->
INVITE sip:101@59.XX.XX.XX SIP/2.0
Via: SIP/2.0/WS ksdmllpbgsdf.invalid;branch=z9hG4bK4083279
Max-Forwards: 69
To: <sip:101@59.XX.XX.XX>
From: "102" <sip:102@59.XX.XX.XX>;tag=sa54urm37e
Call-ID: hqaqv9bdm2s5u92le2rn
CSeq: 9778 INVITE
X-Can-Renegotiate: false
Contact: <sip:81iaqkr7@ksdmllpbgsdf.invalid;transport=ws;ob>
Content-Type: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: ice,outbound
User-Agent: JsSIP 0.6.12
Content-Length: 982

v=0
o=Mozilla-SIPUA-34.0.1 26605 0 IN IP4 0.0.0.0
s=SIP Call
t=0 0
a=ice-ufrag:97fd037e
a=ice-pwd:a6822dc433d2cee279c9ea59b1945cb0
a=fingerprint:sha-256 D2:B2:93:C3:22:55:17:DB:FD:F6:62:39:73:B9:59:29:01:5C:62:8C:7C:76:49:92:A0:FD:3C:11:DA:6B:3A:56
m=audio 9 RTP/SAVPF 109 9 0 8 101
c=IN IP4 0.0.0.0
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=rtcp-mux
a=candidate:0 1 UDP 2129133823 192.168.2.103 58090 typ host
a=candidate:0 2 UDP 2129133822 192.168.2.103 58631 typ host
m=video 9 RTP/SAVPF 120
c=IN IP4 0.0.0.0
a=rtpmap:120 VP8/90000
a=sendrecv
a=rtcp-fb:120 nack
a=rtcp-fb:120 nack pli
a=rtcp-fb:120 ccm fir
a=setup:actpass
a=rtcp-mux
a=candidate:0 1 UDP 2129133823 192.168.2.103 42932 typ host
a=candidate:0 2 UDP 2129133822 192.168.2.103 39849 typ host
<------------->
--- (14 headers 33 lines) ---
Using INVITE request as basis request - hqaqv9bdm2s5u92le2rn
Found peer '102' for '102' from 186.XX.XX.XX:60753

<--- Reliably Transmitting (NAT) to 186.XX.XX.XX:60753 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS ksdmllpbgsdf.invalid;branch=z9hG4bK4083279;received=186.XX.XX.XX;rport=60753
From: "102" <sip:102@59.XX.XX.XX>;tag=sa54urm37e
To: <sip:101@59.XX.XX.XX>;tag=as2f2f52a5
Call-ID: hqaqv9bdm2s5u92le2rn
CSeq: 9778 INVITE
Server: Asterisk PBX 13.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="59.XX.XX.XX", nonce="7eb08087"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'hqaqv9bdm2s5u92le2rn' in 32000 ms (Method: INVITE)

<--- SIP read from WS:186.XX.XX.XX:60753 --->
ACK sip:101@59.XX.XX.XX SIP/2.0
Via: SIP/2.0/WS ksdmllpbgsdf.invalid;branch=z9hG4bK4083279
To: <sip:101@59.XX.XX.XX>;tag=as2f2f52a5
From: "102" <sip:102@59.XX.XX.XX>;tag=sa54urm37e
Call-ID: hqaqv9bdm2s5u92le2rn
CSeq: 9778 ACK
Content-Length: 0

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

<--- SIP read from WS:186.XX.XX.XX:60753 --->
INVITE sip:101@59.XX.XX.XX SIP/2.0
Via: SIP/2.0/WS ksdmllpbgsdf.invalid;branch=z9hG4bK3762205
Max-Forwards: 69
To: <sip:101@59.XX.XX.XX>
From: "102" <sip:102@59.XX.XX.XX>;tag=sa54urm37e
Call-ID: hqaqv9bdm2s5u92le2rn
CSeq: 9779 INVITE
Authorization: Digest algorithm=MD5, username="102", realm="59.XX.XX.XX", nonce="7eb08087", uri="sip:101@59.XX.XX.XX", response="5f1068553c36adfcb42bd3527c0ee97c"
X-Can-Renegotiate: false
Contact: <sip:81iaqkr7@ksdmllpbgsdf.invalid;transport=ws;ob>
Content-Type: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: ice,outbound
User-Agent: JsSIP 0.6.12
Content-Length: 982

v=0
o=Mozilla-SIPUA-34.0.1 26605 0 IN IP4 0.0.0.0
s=SIP Call
t=0 0
a=ice-ufrag:97fd037e
a=ice-pwd:a6822dc433d2cee279c9ea59b1945cb0
a=fingerprint:sha-256 D2:B2:93:C3:22:55:17:DB:FD:F6:62:39:73:B9:59:29:01:5C:62:8C:7C:76:49:92:A0:FD:3C:11:DA:6B:3A:56
m=audio 9 RTP/SAVPF 109 9 0 8 101
c=IN IP4 0.0.0.0
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=rtcp-mux
a=candidate:0 1 UDP 2129133823 192.168.2.103 58090 typ host
a=candidate:0 2 UDP 2129133822 192.168.2.103 58631 typ host
m=video 9 RTP/SAVPF 120
c=IN IP4 0.0.0.0
a=rtpmap:120 VP8/90000
a=sendrecv
a=rtcp-fb:120 nack
a=rtcp-fb:120 nack pli
a=rtcp-fb:120 ccm fir
a=setup:actpass
a=rtcp-mux
a=candidate:0 1 UDP 2129133823 192.168.2.103 42932 typ host
a=candidate:0 2 UDP 2129133822 192.168.2.103 39849 typ host
<------------->
--- (15 headers 33 lines) ---
Using INVITE request as basis request - hqaqv9bdm2s5u92le2rn
Found peer '102' for '102' from 186.XX.XX.XX:60753
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Found RTP audio format 109
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format opus for ID 109
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 audio description format telephone-event for ID 101
Found RTP video format 120
Found video description format VP8 for ID 120
Capabilities: us - (ulaw|vp8), peer - audio=(ulaw|alaw|g722|opus)/video=(vp8)/text=(nothing), combined - (ulaw|vp8)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 0.0.0.0:9
Peer video RTP is at port 0.0.0.0:9
Looking for 101 in video-test (domain 59.XX.XX.XX)
sip_route_dump: route/path hop: <sip:81iaqkr7@ksdmllpbgsdf.invalid;transport=ws;ob>

<--- Transmitting (NAT) to 186.XX.XX.XX:60753 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS ksdmllpbgsdf.invalid;branch=z9hG4bK3762205;received=186.XX.XX.XX;rport=60753
From: "102" <sip:102@59.XX.XX.XX>;tag=sa54urm37e
To: <sip:101@59.XX.XX.XX>
Call-ID: hqaqv9bdm2s5u92le2rn
CSeq: 9779 INVITE
Server: Asterisk PBX 13.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:101@10.37.172.89:5060;transport=WS>
Content-Length: 0


<------------>
    -- Executing [101@video-test:1] Answer("SIP/102-00000006", "") in new stack
Audio is at 12956
Video is at 10.37.172.89:14270
Adding codec ulaw to SDP
Adding video codec vp8 to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 186.XX.XX.XX:60753 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS ksdmllpbgsdf.invalid;branch=z9hG4bK3762205;received=186.XX.XX.XX;rport=60753
From: "102" <sip:102@59.XX.XX.XX>;tag=sa54urm37e
To: <sip:101@59.XX.XX.XX>;tag=as0df222f9
Call-ID: hqaqv9bdm2s5u92le2rn
CSeq: 9779 INVITE
Server: Asterisk PBX 13.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:101@10.37.172.89:5060;transport=WS>
Content-Type: application/sdp
Content-Length: 1499

v=0
o=root 1055886195 1055886195 IN IP4 10.37.172.89
s=Asterisk PBX 13.1.0
c=IN IP4 10.37.172.89
b=CT:384
t=0 0
m=audio 12956 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:6cc40d140ce4c91742bbef110ab14dc1
a=ice-pwd:2da0f5fd2d62479a5356720b55c04580
a=candidate:Ha25ac59 1 UDP 2130706431 10.37.172.89 12956 typ host
a=candidate:S369fc99e 1 UDP 1694498815 59.XX.XX.XX 12956 typ srflx raddr 10.37.172.89 rport 12956
a=candidate:Ha25ac59 2 UDP 2130706430 10.37.172.89 12957 typ host
a=candidate:S369fc99e 2 UDP 1694498814 59.XX.XX.XX 12957 typ srflx raddr 10.37.172.89 rport 12957
a=connection:new
a=setup:active
a=fingerprint:SHA-256 04:9A:2F:12:1F:73:73:20:F5:B3:D9:08:FB:10:6F:86:1D:5F:60:2B:98:D4:71:26:B9:D2:87:36:41:64:FA:A7
a=sendrecv
m=video 14270 RTP/SAVPF 120
a=ice-ufrag:30e69b685abdf0de5caa63f31d7a72f9
a=ice-pwd:5dbab6bb24b2bc737bb50c9a2804a623
a=candidate:Ha25ac59 1 UDP 2130706431 10.37.172.89 14270 typ host
a=candidate:S369fc99e 1 UDP 1694498815 59.XX.XX.XX 14270 typ srflx raddr 10.37.172.89 rport 14270
a=candidate:Ha25ac59 2 UDP 2130706430 10.37.172.89 14271 typ host
a=candidate:S369fc99e 2 UDP 1694498814 59.XX.XX.XX 14271 typ srflx raddr 10.37.172.89 rport 14271
a=connection:new
a=setup:active
a=fingerprint:SHA-256 04:9A:2F:12:1F:73:73:20:F5:B3:D9:08:FB:10:6F:86:1D:5F:60:2B:98:D4:71:26:B9:D2:87:36:41:64:FA:A7
a=rtpmap:120 VP8/90000
a=rtcp-fb:* ccm fir
a=sendrecv

<------------>
    -- Executing [101@video-test:2] Wait("SIP/102-00000006", "3") in new stack

<--- SIP read from WS:186.XX.XX.XX:60753 --->
ACK sip:101@10.37.172.89:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS ksdmllpbgsdf.invalid;branch=z9hG4bK9533906
Max-Forwards: 69
To: <sip:101@59.XX.XX.XX>;tag=as0df222f9
From: "102" <sip:102@59.XX.XX.XX>;tag=sa54urm37e
Call-ID: hqaqv9bdm2s5u92le2rn
CSeq: 9779 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: outbound
User-Agent: JsSIP 0.6.12
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
       > 0x7fc3880387a0 -- Probation passed - setting RTP source address to 186.XX.XX.XX:42932
       > 0x7fc38800baf0 -- Probation passed - setting RTP source address to 186.XX.XX.XX:58090
    -- Executing [101@video-test:3] Playback("SIP/102-00000006", "letters/asterisk") in new stack
    -- <SIP/102-00000006> Playing 'letters/asterisk.ulaw' (language 'en')
    -- Executing [101@video-test:4] Dial("SIP/102-00000006", "SIP/101") in new stack
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Audio is at 14800
Video is at 10.37.172.89:10976
Adding codec ulaw to SDP
Adding video codec vp8 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 186.XX.XX.XX:37687:
INVITE sip:7dhldjib@jvp15e71m9jt.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.37.172.89:5060;branch=z9hG4bK2d65cf48;rport
Max-Forwards: 70
From: "102" <sip:102@10.37.172.89>;tag=as5abd6230
To: <sip:7dhldjib@jvp15e71m9jt.invalid;transport=ws>
Contact: <sip:102@10.37.172.89:5060;transport=WS>
Call-ID: 4ead8d547980693835eed3c5623cac67@10.37.172.89:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.1.0
Date: Wed, 28 Jan 2015 19:31:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1501

v=0
o=root 1867163197 1867163197 IN IP4 10.37.172.89
s=Asterisk PBX 13.1.0
c=IN IP4 10.37.172.89
b=CT:384
t=0 0
m=audio 14800 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:182c7c4f74e73d0624009e24222e9c85
a=ice-pwd:757b0db73bfe1a003b2796660539eb81
a=candidate:Ha25ac59 1 UDP 2130706431 10.37.172.89 14800 typ host
a=candidate:S369fc99e 1 UDP 1694498815 59.XX.XX.XX 14800 typ srflx raddr 10.37.172.89 rport 14800
a=candidate:Ha25ac59 2 UDP 2130706430 10.37.172.89 14801 typ host
a=candidate:S369fc99e 2 UDP 1694498814 59.XX.XX.XX 14801 typ srflx raddr 10.37.172.89 rport 14801
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 04:9A:2F:12:1F:73:73:20:F5:B3:D9:08:FB:10:6F:86:1D:5F:60:2B:98:D4:71:26:B9:D2:87:36:41:64:FA:A7
a=sendrecv
m=video 10976 RTP/SAVPF 100
a=ice-ufrag:3010cd2c58ca0d3d4caf0f0905271818
a=ice-pwd:19cff47b4c20adaf42408fb2074ea8a3
a=candidate:Ha25ac59 1 UDP 2130706431 10.37.172.89 10976 typ host
a=candidate:S369fc99e 1 UDP 1694498815 59.XX.XX.XX 10976 typ srflx raddr 10.37.172.89 rport 10976
a=candidate:Ha25ac59 2 UDP 2130706430 10.37.172.89 10977 typ host
a=candidate:S369fc99e 2 UDP 1694498814 59.XX.XX.XX 10977 typ srflx raddr 10.37.172.89 rport 10977
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 04:9A:2F:12:1F:73:73:20:F5:B3:D9:08:FB:10:6F:86:1D:5F:60:2B:98:D4:71:26:B9:D2:87:36:41:64:FA:A7
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=sendrecv

---
    -- Called SIP/101

<--- SIP read from WS:186.XX.XX.XX:37687 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS 10.37.172.89:5060;branch=z9hG4bK2d65cf48;rport
To: <sip:7dhldjib@jvp15e71m9jt.invalid;transport=ws>
From: "102" <sip:102@10.37.172.89>;tag=as5abd6230
Call-ID: 4ead8d547980693835eed3c5623cac67@10.37.172.89:5060
CSeq: 102 INVITE
Supported: ice,outbound
Content-Length: 0

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

<--- SIP read from WS:186.XX.XX.XX:37687 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 10.37.172.89:5060;branch=z9hG4bK2d65cf48;rport
To: <sip:7dhldjib@jvp15e71m9jt.invalid;transport=ws>;tag=a6tct1dgu3
From: "102" <sip:102@10.37.172.89>;tag=as5abd6230
Call-ID: 4ead8d547980693835eed3c5623cac67@10.37.172.89:5060
CSeq: 102 INVITE
Contact: <sip:7dhldjib@jvp15e71m9jt.invalid;transport=ws>
Supported: ice,outbound
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:7dhldjib@jvp15e71m9jt.invalid;transport=ws>
    -- SIP/101-00000007 is ringing

<--- SIP read from WS:186.XX.XX.XX:37687 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 10.37.172.89:5060;branch=z9hG4bK2d65cf48;rport
To: <sip:7dhldjib@jvp15e71m9jt.invalid;transport=ws>;tag=a6tct1dgu3
From: "102" <sip:102@10.37.172.89>;tag=as5abd6230
Call-ID: 4ead8d547980693835eed3c5623cac67@10.37.172.89:5060
CSeq: 102 INVITE
Contact: <sip:7dhldjib@jvp15e71m9jt.invalid;transport=ws>
X-Can-Renegotiate: false
Supported: ice,outbound
Content-Type: application/sdp
Content-Length: 762

v=0
o=Mozilla-SIPUA-34.0 6957 0 IN IP4 0.0.0.0
s=SIP Call
t=0 0
a=ice-ufrag:c5484af7
a=ice-pwd:695f916d590ab08a8771ffe5146e503b
a=fingerprint:sha-256 66:45:3F:AA:E3:92:F9:ED:0F:00:51:E0:07:12:E6:3A:B8:16:51:FC:4A:30:68:F2:36:A9:3C:DA:F0:C9:3A:C7
m=audio 9 RTP/SAVPF 0 101
c=IN IP4 0.0.0.0
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=setup:active
a=candidate:0 1 UDP 2122252543 192.168.2.106 59186 typ host
a=candidate:0 2 UDP 2122252542 192.168.2.106 41177 typ host
m=video 9 RTP/SAVPF 100
c=IN IP4 0.0.0.0
a=rtpmap:100 VP8/90000
a=sendrecv
a=rtcp-fb:100 ccm fir
a=setup:active
a=candidate:0 1 UDP 2122252543 192.168.2.106 43997 typ host
a=candidate:0 2 UDP 2122252542 192.168.2.106 57885 typ host
<------------->
--- (11 headers 24 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Found RTP video format 100
Found video description format VP8 for ID 100
Capabilities: us - (ulaw|vp8), peer - audio=(ulaw)/video=(vp8)/text=(nothing), combined - (ulaw|vp8)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 0.0.0.0:9
Peer video RTP is at port 0.0.0.0:9
sip_route_dump: route/path hop: <sip:7dhldjib@jvp15e71m9jt.invalid;transport=ws>
Transmitting (NAT) to 186.XX.XX.XX:37687:
ACK sip:7dhldjib@jvp15e71m9jt.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.37.172.89:5060;branch=z9hG4bK5fee6a77;rport
Max-Forwards: 70
From: "102" <sip:102@10.37.172.89>;tag=as5abd6230
To: <sip:7dhldjib@jvp15e71m9jt.invalid;transport=ws>;tag=a6tct1dgu3
Contact: <sip:102@10.37.172.89:5060;transport=WS>
Call-ID: 4ead8d547980693835eed3c5623cac67@10.37.172.89:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.1.0
Content-Length: 0