While calling “488 Not Acceptable here”

I have some problems when trying to connect to Asterisk and make calls using WebRTC.
I have installed Asterisk 18.
I am using SIP.js version 0.11.4. I was able to register successfully, but when I try to make call I receive 488 Not Acceptable Here.

please see this sip track log

<— Received SIP request (3394 bytes) from WSS:66.154.105.4:25357 —>
INVITE sip:102030@rus.za SIP/2.0
Via: SIP/2.0/WSS 192.0.2.20;branch=z9hG4bK1859767
To: sip:102030@rus.za
From: “2022207” sip:2022207@rus.za;tag=ltl3sm2l95
CSeq: 1 INVITE
Call-ID: ccdnovef64n75st5dg2g
Max-Forwards: 70
Contact: sip:v9qpk1a3@192.0.2.20;transport=wss;ob
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0)
Content-Type: application/sdp
Content-Length: 2896

v=0
o=- 8356290575410731161 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS c7bc33f4-06fb-40cb-910d-d92f67ddefcf
m=audio 56547 UDP/TLS/RTP/SAVPF 111 63 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 66.154.105.4
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:2078000290 1 udp 2122260223 198.18.21.196 56547 typ host generation 0 network-id 1 network-cost 50
a=candidate:2877176657 1 udp 2122197247 2408:8234:9c13:3d0:7846:3761:9e7d:78e7 56548 typ host generation 0 network-id 4
a=candidate:3123987641 1 udp 2122131711 2408:8234:9c13:3d0:fc57:4322:3eb5:d239 56549 typ host generation 0 network-id 5
a=candidate:3885784340 1 udp 2122063615 170.170.0.51 56550 typ host generation 0 network-id 2
a=candidate:2150709124 1 udp 2121998079 192.168.0.155 56551 typ host generation 0 network-id 3
a=candidate:894974034 1 tcp 1518280447 198.18.21.196 9 typ host tcptype active generation 0 network-id 1 network-cost 50
a=candidate:3858614177 1 tcp 1518217471 2408:8234:9c13:3d0:7846:3761:9e7d:78e7 9 typ host tcptype active generation 0 network-id 4
a=candidate:4105680969 1 tcp 1518151935 2408:8234:9c13:3d0:fc57:4322:3eb5:d239 9 typ host tcptype active generation 0 network-id 5
a=candidate:2837422564 1 tcp 1518083839 170.170.0.51 9 typ host tcptype active generation 0 network-id 2
a=candidate:3467823988 1 tcp 1518018303 192.168.0.155 9 typ host tcptype active generation 0 network-id 3
a=candidate:4203987478 1 udp 1686052607 66.154.105.4 56547 typ srflx raddr 198.18.21.196 rport 56547 generation 0 network-id 1 network-cost 50
a=candidate:294397120 1 udp 1685855999 58.245.96.42 56550 typ srflx raddr 170.170.0.51 rport 56550 generation 0 network-id 2
a=ice-ufrag:/m0T
a=ice-pwd:ySu5mM/C2MkAkxVWmRcBwCch
a=ice-options:trickle
a=fingerprint:sha-256 BB:1B:23:A2:28:65:18:FF:DC:3D:03:51:1A:16:81:86:58:C7:57:AE:71:4C:BF:F2:84:15:3A:C0:21:3B:31:28
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 docs/native-code/rtp-hdrext/abs-send-time - src - Git at Google
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:c7bc33f4-06fb-40cb-910d-d92f67ddefcf 296e671a-5c0c-4f66-9226-528f0a4a2ddf
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:1131245714 cname:ALKrG9zzJEC6d0yO
a=ssrc:1131245714 msid:c7bc33f4-06fb-40cb-910d-d92f67ddefcf 296e671a-5c0c-4f66-9226-528f0a4a2ddf

<— Transmitting SIP response (474 bytes) to WSS:66.154.105.4:25357 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.20;rport=25357;received=66.154.105.4;branch=z9hG4bK1859767
Call-ID: ccdnovef64n75st5dg2g
From: “2022207” sip:2022207@rus.za;tag=ltl3sm2l95
To: sip:102030@rus.za;tag=z9hG4bK1859767
CSeq: 1 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1658478113/04dd95cb6b8116d2fd779077187492ad”,opaque=“675bdef96fe43d90”,algorithm=md5,qop=“auth”
Server: FPBX-16.0.21.3(16.25.0)
Content-Length: 0

<— Received SIP request (289 bytes) from WSS:66.154.105.4:25357 —>
ACK sip:102030@rus.za SIP/2.0
Via: SIP/2.0/WSS 192.0.2.20;branch=z9hG4bK1859767
To: sip:102030@rus.za;tag=z9hG4bK1859767
From: “2022207” sip:2022207@rus.za;tag=ltl3sm2l95
Call-ID: ccdnovef64n75st5dg2g
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

<— Received SIP request (3674 bytes) from WSS:66.154.105.4:25357 —>
INVITE sip:102030@rus.za SIP/2.0
Via: SIP/2.0/WSS 192.0.2.20;branch=z9hG4bK6957869
To: sip:102030@rus.za
From: “2022207” sip:2022207@rus.za;tag=ltl3sm2l95
CSeq: 2 INVITE
Call-ID: ccdnovef64n75st5dg2g
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username=“2022207”, realm=“asterisk”, nonce=“1658478113/04dd95cb6b8116d2fd779077187492ad”, uri=“sip:102030@rus.za”, response=“597876c49b0c7de24a25d309e0fd92db”, opaque=“675bdef96fe43d90”, qop=auth, cnonce=“uhaok2qpsfc4”, nc=00000001
Contact: sip:v9qpk1a3@192.0.2.20;transport=wss;ob
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0)
Content-Type: application/sdp
Content-Length: 2896

v=0
o=- 8356290575410731161 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS c7bc33f4-06fb-40cb-910d-d92f67ddefcf
m=audio 56547 UDP/TLS/RTP/SAVPF 111 63 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 66.154.105.4
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:2078000290 1 udp 2122260223 198.18.21.196 56547 typ host generation 0 network-id 1 network-cost 50
a=candidate:2877176657 1 udp 2122197247 2408:8234:9c13:3d0:7846:3761:9e7d:78e7 56548 typ host generation 0 network-id 4
a=candidate:3123987641 1 udp 2122131711 2408:8234:9c13:3d0:fc57:4322:3eb5:d239 56549 typ host generation 0 network-id 5
a=candidate:3885784340 1 udp 2122063615 170.170.0.51 56550 typ host generation 0 network-id 2
a=candidate:2150709124 1 udp 2121998079 192.168.0.155 56551 typ host generation 0 network-id 3
a=candidate:894974034 1 tcp 1518280447 198.18.21.196 9 typ host tcptype active generation 0 network-id 1 network-cost 50
a=candidate:3858614177 1 tcp 1518217471 2408:8234:9c13:3d0:7846:3761:9e7d:78e7 9 typ host tcptype active generation 0 network-id 4
a=candidate:4105680969 1 tcp 1518151935 2408:8234:9c13:3d0:fc57:4322:3eb5:d239 9 typ host tcptype active generation 0 network-id 5
a=candidate:2837422564 1 tcp 1518083839 170.170.0.51 9 typ host tcptype active generation 0 network-id 2
a=candidate:3467823988 1 tcp 1518018303 192.168.0.155 9 typ host tcptype active generation 0 network-id 3
a=candidate:4203987478 1 udp 1686052607 66.154.105.4 56547 typ srflx raddr 198.18.21.196 rport 56547 generation 0 network-id 1 network-cost 50
a=candidate:294397120 1 udp 1685855999 58.245.96.42 56550 typ srflx raddr 170.170.0.51 rport 56550 generation 0 network-id 2
a=ice-ufrag:/m0T
a=ice-pwd:ySu5mM/C2MkAkxVWmRcBwCch
a=ice-options:trickle
a=fingerprint:sha-256 BB:1B:23:A2:28:65:18:FF:DC:3D:03:51:1A:16:81:86:58:C7:57:AE:71:4C:BF:F2:84:15:3A:C0:21:3B:31:28
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 docs/native-code/rtp-hdrext/abs-send-time - src - Git at Google
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:c7bc33f4-06fb-40cb-910d-d92f67ddefcf 296e671a-5c0c-4f66-9226-528f0a4a2ddf
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:1131245714 cname:ALKrG9zzJEC6d0yO
a=ssrc:1131245714 msid:c7bc33f4-06fb-40cb-910d-d92f67ddefcf 296e671a-5c0c-4f66-9226-528f0a4a2ddf

<— Transmitting SIP response (303 bytes) to WSS:66.154.105.4:25357 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 192.0.2.20;rport=25357;received=66.154.105.4;branch=z9hG4bK6957869
Call-ID: ccdnovef64n75st5dg2g
From: “2022207” sip:2022207@rus.za;tag=ltl3sm2l95
To: sip:102030@rus.za
CSeq: 2 INVITE
Server: FPBX-16.0.21.3(16.25.0)
Content-Length: 0

[2022-07-22 08:21:53] ERROR[29974]: res_pjsip_session.c:937 handle_incoming_sdp: 2022207: Couldn’t negotiate stream 0:audio-0:audio:sendrecv (nothing)
<— Transmitting SIP response (357 bytes) to WSS:66.154.105.4:25357 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WSS 192.0.2.20;rport=25357;received=66.154.105.4;branch=z9hG4bK6957869
Call-ID: ccdnovef64n75st5dg2g
From: “2022207” sip:2022207@rus.za;tag=ltl3sm2l95
To: sip:102030@rus.za;tag=fa6db4d5-3257-425c-a3ac-07396b323ef4
CSeq: 2 INVITE
Server: FPBX-16.0.21.3(16.25.0)
Content-Length: 0

<— Received SIP request (311 bytes) from WSS:66.154.105.4:25357 —>
ACK sip:102030@rus.za SIP/2.0
Via: SIP/2.0/WSS 192.0.2.20;branch=z9hG4bK6957869
To: sip:102030@rus.za;tag=fa6db4d5-3257-425c-a3ac-07396b323ef4
From: “2022207” sip:2022207@rus.za;tag=ltl3sm2l95
Call-ID: ccdnovef64n75st5dg2g
CSeq: 2 ACK
Max-Forwards: 70
Content-Length: 0

<— Transmitting SIP request (488 bytes) to WSS:66.154.105.4:16356 —>
OPTIONS sip:okl07kn0@66.154.105.4:16356;transport=ws SIP/2.0
Via: SIP/2.0/WSS 31.42.177.246:8089;rport;branch=z9hG4bKPj7cea2191-f544-4918-bd74-7f9d070eb548;alias
From: sip:147563@freepbx.sangoma.local;tag=514c9123-4090-4d26-99be-f008fde0a7b1
To: sip:okl07kn0@66.154.105.4
Contact: sip:147563@freepbx.sangoma.local:5060;transport=ws
Call-ID: 631ac518-5e48-4452-bd8a-83f4886f0b8b
CSeq: 41215 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-16.0.21.3(16.25.0)
Content-Length: 0

<— Received SIP response (498 bytes) from WSS:66.154.105.4:16356 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS 31.42.177.246:8089;rport;branch=z9hG4bKPj7cea2191-f544-4918-bd74-7f9d070eb548;alias
To: sip:okl07kn0@66.154.105.4;tag=ai7e7cm0fl
From: sip:147563@freepbx.sangoma.local;tag=514c9123-4090-4d26-99be-f008fde0a7b1
Call-ID: 631ac518-5e48-4452-bd8a-83f4886f0b8b
CSeq: 41215 OPTIONS
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Accept: application/sdp,application/dtmf-relay
Supported: outbound
User-Agent: SIP.js/0.10.0
Content-Length: 0

log messsage have an answer

[2022-07-22 08:21:53] ERROR[29974]: res_pjsip_session.c:937 handle_incoming_sdp: 2022207: Couldn’t negotiate stream 0:audio-0:audio:sendrecv (nothing)

yes. I know.
But I dont know the way to solve the problem

It would be the underlying endpoint configuration on the Asterisk side, or not all required modules (such as res_srtp) are loaded.

how can i check this?
could you tell me guide shortly?

PJSIP configuration is commonly done in the pjsip.conf file. I should also add that leaping into WebRTC without a solid VoIP and Asterisk foundation is extremely difficult and prone to issues.