11/09/2015: SIP REGISTER ASTERISK WEBRTC

      == WebSocket connection from '148.X.X.X:46841' for protocol 'sip' accepted using version '13'

    <--- SIP read from WS:148.X.X.X:46841 --->
    REGISTER sip:148.X.X.X SIP/2.0
    Via: SIP/2.0/WS 192.0.2.36;branch=z9hG4bK51938
    Max-Forwards: 69
    To: <sip:1001@148.X.X.X>
    From: "UA WebRTC" <sip:1001@148.X.X.X>;tag=v6n4ogr1hb
    Call-ID: 8rfvjcjhat0kfjs50oihga
    CSeq: 1 REGISTER
    Contact: <sip:m4bvnghf@192.0.2.36;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:2a64f5ad-3915-4158-9fd0-c2d570199112>";expires=600
    Expires: 600
    Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER
    Supported: path,gruu,outbound
    User-Agent: JsSIP 0.7.4
    Content-Length: 0

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

    <--- Transmitting (no NAT) to 148.X.X.X:5060 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/WS 192.0.2.36;branch=z9hG4bK51938;received=148.X.X.X
    From: "UA WebRTC" <sip:1001@148.X.X.X>;tag=v6n4ogr1hb
    To: <sip:1001@148.X.X.X>;tag=as1b57d07a
    Call-ID: 8rfvjcjhat0kfjs50oihga
    CSeq: 1 REGISTER
    Server: Asterisk PBX 12.8.2
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1f8ce5ba"
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog '8rfvjcjhat0kfjs50oihga' in 32000 ms (Method: REGISTER)

    <--- SIP read from WS:148.X.X.X:46841 --->
    REGISTER sip:148.X.X.X SIP/2.0
    Via: SIP/2.0/WS 192.0.2.36;branch=z9hG4bK4391054
    Max-Forwards: 69
    To: <sip:1001@148.X.X.X>
    From: "UA WebRTC" <sip:1001@148.X.X.X>;tag=v6n4ogr1hb
    Call-ID: 8rfvjcjhat0kfjs50oihga
    CSeq: 2 REGISTER
    Authorization: Digest algorithm=MD5, username="1001", realm="asterisk", nonce="1f8ce5ba", uri="sip:148.X.X.X", response="9e9880037daa27f172203d4562e60571"
    Contact: <sip:m4bvnghf@192.0.2.36;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:2a64f5ad-3915-4158-9fd0-c2d570199112>";expires=600
    Expires: 600
    Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER
    Supported: path,gruu,outbound
    User-Agent: JsSIP 0.7.4
    Content-Length: 0

    <------------->
    --- (14 headers 0 lines) ---
      == WebSocket connection from '148.X.X.X:46809' closed
        -- Registered SIP '1001' at 148.X.X.X:46841

    <--- Transmitting (no NAT) to 148.X.X.X:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/WS 192.0.2.36;branch=z9hG4bK4391054;received=148.X.X.X
    From: "UA WebRTC" <sip:1001@148.X.X.X>;tag=v6n4ogr1hb
    To: <sip:1001@148.X.X.X>;tag=as1b57d07a
    Call-ID: 8rfvjcjhat0kfjs50oihga
    CSeq: 2 REGISTER
    Server: Asterisk PBX 12.8.2
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Expires: 600
    Contact: <sip:m4bvnghf@192.0.2.36;transport=ws>;expires=600
    Date: Fri, 11 Sep 2015 20:13:37 GMT
    Content-Length: 0


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