Asterisk + WebRTC subscribing UA jsSIP

REGISTER sip:192.168.0.4 SIP/2.0
Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK9352120
Max-Forwards: 69
To: <sip:1001@192.168.0.4>
From: "UA WebRTC" <sip:1001@192.168.0.4>;tag=f8m06houfb
Call-ID: liamoji649fhds9jtdajfi
CSeq: 1 REGISTER
Contact: <sip:grd0bbtt@192.0.2.177;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:b1a09033-7533-47a8-a039-1a6760b2af84>";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

+15ms
jssip-0.7.4.js (line 22459)
JsSIP:Transport
received WebSocket text message:

SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK9352120;received=192.168.0.4
From: "UA WebRTC" <sip:1001@192.168.0.4>;tag=f8m06houfb
To: <sip:1001@192.168.0.4>;tag=as2235a73d
Call-ID: liamoji649fhds9jtdajfi
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="0f6c6c26"
Content-Length: 0


+30ms
jssip-0.7.4.js (line 22459)
JsSIP:Transport
sending WebSocket message:

REGISTER sip:192.168.0.4 SIP/2.0
Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK1900716
Max-Forwards: 69
To: <sip:1001@192.168.0.4>
From: "UA WebRTC" <sip:1001@192.168.0.4>;tag=f8m06houfb
Call-ID: liamoji649fhds9jtdajfi
CSeq: 2 REGISTER
Authorization: Digest algorithm=MD5, username="1001", realm="asterisk", nonce="0f6c6c26", uri="sip:192.168.0.4", response="bed4984f5fc3c21f3854dccdb30a74b2"
Contact: <sip:grd0bbtt@192.0.2.177;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:b1a09033-7533-47a8-a039-1a6760b2af84>";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

+32ms
jssip-0.7.4.js (line 22459)
JsSIP:Transport
received WebSocket text message:

SIP/2.0 200 OK
Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK1900716;received=192.168.0.4
From: "UA WebRTC" <sip:1001@192.168.0.4>;tag=f8m06houfb
To: <sip:1001@192.168.0.4>;tag=as2235a73d
Call-ID: liamoji649fhds9jtdajfi
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:grd0bbtt@192.0.2.177;transport=ws>;expires=600
Date: Mon, 24 Aug 2015 14:15:52 GMT
Content-Length: 0