SIP/2.0 488 Not Acceptable Here on webRTC welcome call login

[samadsaeed]

Hello,
Vicidial scratch install on:

Intel® Core™ i7-8700 Hexa-Core Coffee Lake
64 GB DDR4 RAM
2 x 1 TB NVMe SSD (Software-RAID 1)
1 Gbit/s bandwidth

OS: Centos 7.7 64bits
Asterisk: 13.21.0-vici
VERSION: 2.14-750a
BUILD: 200409-1719
manual followed: https://vicigeek.com/vicidial-scratch-i ... terisk-13/

I have configured webRTC on this as per requirement. I'm able to do it successfully on vicibox 9.0.1.
On this scratch installation I have managed to install SRTP, OPUS Codec and SSL certificates are also working fine (SSL certificates are acquired from LetsEncrypt using certbot). and manually placed them in /etc/asterisk/http.conf and httpd ssl.conf etc. Webphone is registering OK. but when I try to login on phone 8001 it doesnt get the welcome call. when i set the SIP debug ON it shows the following.


v=0
o=root 81223449 81223449 IN IP4 95.217.111.112
s=Asterisk PBX 13.21.0-vici
c=IN IP4 95.217.111.112
t=0 0
m=audio 17400 UDP/TLS/RTP/SAVPF 0 8 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 1A:90:17:BD:26:FA:16:CD:40:29:29:5F:B4:25:0F:06:E1:B3:50:87:28:A8:5A:4F:8E:16:74:DC:9F:74:C0:77
a=rtcp-mux
a=sendrecv

---
-- Called 8001

<--- SIP read from WS:45.116.232.59:22459 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS 95.217.111.112:5060;branch=z9hG4bK390a83c0;rport
To: <sip:k0vmdrov@192.0.2.183;transport=wss>
From: "ACagcW15873828026666666666666666" <sip:0000000000@95.217.111.112>;tag=as1974ec42
Call-ID: 2f5d524b293192da0b6e15b0650f1fd9@95.217.111.112:5060
CSeq: 102 INVITE
Supported: outbound
User-Agent: VICIphone 1.0-rc1
Content-Length: 0

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

<--- SIP read from WS:45.116.232.59:22459 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WS 95.217.111.112:5060;branch=z9hG4bK390a83c0;rport
To: <sip:k0vmdrov@192.0.2.183;transport=wss>;tag=bs709pgq95
From: "ACagcW15873828026666666666666666" <sip:0000000000@95.217.111.112>;tag=as1974ec42
Call-ID: 2f5d524b293192da0b6e15b0650f1fd9@95.217.111.112:5060
CSeq: 102 INVITE
Supported: outbound
User-Agent: VICIphone 1.0-rc1
Content-Length: 0
---------------------------------->

below is my webRTC phone template:

type=friend
host=dynamic
trustrpid=yes
sendrpid=no
qualify=yes
qualifyfreq=600
transport=ws,wss,udp
encryption=yes
avpf=yes
icesupport=yes
rtcp_mux=yes
directmedia=no
disallow=all
allow=ulaw
allow=alaw
allow=opus
nat=yes
directmedia=no
dtlsenable=yes
dtlsverify=no
dtlscertfile=/etc/certbot/live/samad1.kalsoninc.com/cert.pem
dtlsprivatekey=/etc/certbot/live/samad1.kalsoninc.com/privkey.pem
dtlssetup=actpass

I have tried different codecs but i think there is something still missing. Can anyone please help me here? I would really be grateful to you, just help me point out the fix for not getting the welcome call. i'm here to provide you any further information required for visibility. Thanks
PS: VICIBOX ISO is not used because the hosting provider doesn't support ISO upload or IPMI access.

2 days have passed still no response from anyone! can anyone please tag some experts here who can help me sort out this. I think ICE isn't working i've missed something. please help me!