== WebSocket connection from '192.168.0.105:53643' for protocol 'sip' accepted using version '13'
<--- Received SIP request (568 bytes) from WS:192.168.0.105:53643 --->
REGISTER sip:192.168.0.106 SIP/2.0
Via: SIP/2.0/WS k5v7qdhee21i.invalid;branch=z9hG4bK211693
Max-Forwards: 69
To: <sip:[email protected]>
From: "User WebRTC 6001" <sip:[email protected]>;tag=88kogfq6mh
Call-ID: ro5f0ohad43lkhevfara26
CSeq: 1 REGISTER
Contact: <sip:[email protected];transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:a384eef3-4ff1-4e4d-b5bb-14aa5620b252>";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: path,gruu,outbound
User-Agent: JsSIP 2.0.2
Content-Length: 0


<--- Transmitting SIP response (484 bytes) to WS:192.168.0.105:53643 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS k5v7qdhee21i.invalid;rport=53643;received=192.168.0.105;branch=z9hG4bK211693
Call-ID: ro5f0ohad43lkhevfara26
From: "User WebRTC 6001" <sip:[email protected]>;tag=88kogfq6mh
To: <sip:[email protected]>;tag=z9hG4bK211693
CSeq: 1 REGISTER
WWW-Authenticate: Digest  realm="asterisk",nonce="1469177118/e77523413fd8316816034a71dd26332f",opaque="07e9ed120ad79990",algorithm=md5,qop="auth"
Server: Asterisk PBX 13.10.0
Content-Length:  0


<--- Received SIP request (835 bytes) from WS:192.168.0.105:53643 --->
REGISTER sip:192.168.0.106 SIP/2.0
Via: SIP/2.0/WS k5v7qdhee21i.invalid;branch=z9hG4bK52092
Max-Forwards: 69
To: <sip:[email protected]>
From: "User WebRTC 6001" <sip:[email protected]>;tag=88kogfq6mh
Call-ID: ro5f0ohad43lkhevfara26
CSeq: 2 REGISTER
Authorization: Digest algorithm=MD5, username="6001", realm="asterisk", nonce="1469177118/e77523413fd8316816034a71dd26332f", uri="sip:192.168.0.106", response="a11e368c3688910978e00a3638352dd9", opaque="07e9ed120ad79990", qop=auth, cnonce="nl0vurqeqp95", nc=00000001
Contact: <sip:[email protected];transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:a384eef3-4ff1-4e4d-b5bb-14aa5620b252>";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: path,gruu,outbound
User-Agent: JsSIP 2.0.2
Content-Length: 0


    -- Added contact 'sip:[email protected]:53643;transport=ws' to AOR '6001' with expiration of 600 seconds
<--- Transmitting SIP response (446 bytes) to WS:192.168.0.105:53643 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS k5v7qdhee21i.invalid;rport=53643;received=192.168.0.105;branch=z9hG4bK52092
Call-ID: ro5f0ohad43lkhevfara26
From: "User WebRTC 6001" <sip:[email protected]>;tag=88kogfq6mh
To: <sip:[email protected]>;tag=z9hG4bK52092
CSeq: 2 REGISTER
Date: Fri, 22 Jul 2016 08:45:18 GMT
Contact: <sip:[email protected]:53643;transport=ws>;expires=599
Expires: 600
Server: Asterisk PBX 13.10.0
Content-Length:  0


  == Contact 6001/sip:[email protected]:53643;transport=ws has been created
  == Contact 6001/sip:[email protected]:53622;transport=ws has been deleted
    -- Contact 6001/sip:[email protected]:53643;transport=ws is now Unknown.  RTT: 0.000 msec