Invite:

Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKmUkFPRoiZpCT9H2z225tdyAZnDJNcvW9;received=xxx.xxx.xxx.xxx;rport=2092
From: "1060"<sip:1060@sip3.website.com>;tag=krzZncRmZCIX25ILV3wu
To: <sip:1061@sip3.website.com>;tag=as455e418c
...
Server: Asterisk PBX 11.8.0-rc1
...
Contact: <sip:1061@xxx.xxx.xxx.xxx:5060;transport=WS>


v=0
o=root 1926966382 1926966382 IN IP4 xxx.xxx.xxx.xxx
s=Asterisk PBX 11.8.0-rc1
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 19888 RTP/SAVPF 0 0 8 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:7a13ee9f5dd85b135f8649af108f0d71
a=ice-pwd:6436aeba68e1514635ab1e6021edc44b
...
a=connection:new
a=setup:active
a=fingerprint:SHA-256 4B:D0:F1:39:36:B7:B0:91:57:87:9F:1F:6E:CF:C4:37:D2:BB:C7:72:F9:DC:A4:CB:50:64:9E:0D:FF:D0:40:39
a=sendrecv


Invite from Asterisk:


INVITE sips:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss SIP/2.0
Via: SIP/2.0/WS xxx.xxx.xxx.xxx:5060;branch=z9hG4bK2af64b4a;rport
Max-Forwards: 70
From: "New User" <sip:1060@xxx.xxx.xxx.xxx>;tag=as28ac4957
To: <sips:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>
Contact: <sip:1060@xxx.xxx.xxx.xxx:5060;transport=WS>
...
User-Agent: Asterisk PBX 11.8.0-rc1
...
Content-Type: application/sdp
Content-Length: 2075

v=0
o=root 1826944884 1826944884 IN IP4 xxx.xxx.xxx.xxx
s=Asterisk PBX 11.8.0-rc1
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 10964 RTP/SAVPF 0 4 3 3 8 8 112 5 7 18 110 97 111 9 102 115 116 117 96 100 107 108 10 118 119 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
...
a=ice-ufrag:2eeea59a1a8fdf60118d4b370f1a4cba
a=ice-pwd:59cab47971dc46593474a11d4d287cda
...
a=connection:new
a=setup:active
a=sendrecv

As you can see, the WSS INVITE is still going through WS from Asterisk