Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing) using webrtc and udp clients in softphone

I’m encountering an issue where I’m unable to make calls from user 300 to user 200, but I can successfully call from user 200 to user 300. This setup involves using Zoiper and MicroSIP softphones, and everything is configured to run locally on my machine.

[global]
type=global
user_agent=My Asterisk Server

[udp]
type=transport
protocol=udp
bind=0.0.0.0
tos=af42
cos=3

[ws]
type=transport
protocol=wss ; Use 'wss' for WebSocket Secure
bind=0.0.0.0 ; Replace with your Asterisk server's IP and secure port

[tcp]
type=transport
protocol=tcp
bind=0.0.0.0

[endpoint-basic](!)
type=endpoint
transport=ws
context=default ; Replace with the context you use in your dialplan
disallow=all
allow=opus,ulaw,alaw
auth=auth_user
aors=sip_user
webrtc=yes 
dtls_auto_generate_cert=yes
direct_media=no

; Authentication for the User
[auth_user](!)
type=auth
auth_type=userpass

; AOR for the User
[sip_user](!)
type=aor
max_contacts=5 ; Adjust as needed
remove_existing=yes

; Specific User Configuration
[100](endpoint-basic)
auth=auth100
aors=100

[auth100](auth_user)
username=100 ; Replace with your username
password=1234 ; Replace with a strong password

[100](sip_user)

[200](endpoint-basic)
auth=auth200
aors=200

[auth200](auth_user)
username=200 ; Replace with your username
password=1234 ; Replace with a strong password

[200](sip_user)


[300]
type=endpoint
transport=udp
context=default ; Replace with the context you use in your dialplan
disallow=all
allow=opus,ulaw,alaw
auth=300
aors=300

[300]
type=aor
max_contacts=5 ; Adjust as needed
remove_existing=yes

[300]
type=auth
auth_type=userpass
username=300
password=1234


[default]
exten => _XXX,1,NoOp(Calling: ${EXTEN}))
same => n,Dial(PJSIP/${EXTEN},30)
same => n,Hangup()

Logs when calling from user 300 to user 200:

<--- Received SIP request (1299 bytes) from UDP:172.22.160.1:50520 --->
INVITE sip:200@172.22.175.6;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.22.160.1:50520;branch=z9hG4bK-524287-1---a146da0c3e1685ac;rport
Max-Forwards: 70
Contact: <sip:300@172.22.160.1:50520;transport=UDP>
To: <sip:200@172.22.175.6>
From: <sip:300@172.22.175.6;transport=UDP>;tag=aa1a801f
Call-ID: 3xVtqG5bnCTx5uXW5NYkgg..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 629

v=0
o=Z 0 19320051 IN IP4 172.22.160.1
s=Z
c=IN IP4 172.22.160.1
t=0 0
m=audio 64337 RTP/AVPF 106 9 0 8 3 111 97 110 112 98 96 100 99 102
a=rtpmap:106 opus/48000/2
a=fmtp:106 minptime=20; useinbandfec=1
a=rtpmap:111 speex/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:110 speex/8000
a=rtpmap:112 speex/32000
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=rtpmap:100 telephone-event/16000
a=fmtp:100 0-16
a=rtpmap:99 telephone-event/32000
a=fmtp:99 0-16
a=rtpmap:102 G726-32/8000
a=sendrecv
a=rtcp-mux
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir

<--- Transmitting SIP response (498 bytes) to UDP:172.22.160.1:50520 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.22.160.1:50520;rport=50520;received=172.22.160.1;branch=z9hG4bK-524287-1---a146da0c3e1685ac
Call-ID: 3xVtqG5bnCTx5uXW5NYkgg..
From: <sip:300@172.22.175.6>;tag=aa1a801f
To: <sip:200@172.22.175.6>;tag=z9hG4bK-524287-1---a146da0c3e1685ac
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1706288276/f2e7c6b7ac8a6a3a5299de722c9d0509",opaque="2a17209b31a8b500",algorithm=MD5,qop="auth"
Server: My Asterisk Server
Content-Length:  0


<--- Received SIP request (346 bytes) from UDP:172.22.160.1:50520 --->
ACK sip:200@172.22.175.6;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.22.160.1:50520;branch=z9hG4bK-524287-1---a146da0c3e1685ac;rport
Max-Forwards: 70
To: <sip:200@172.22.175.6>;tag=z9hG4bK-524287-1---a146da0c3e1685ac
From: <sip:300@172.22.175.6;transport=UDP>;tag=aa1a801f
Call-ID: 3xVtqG5bnCTx5uXW5NYkgg..
CSeq: 1 ACK
Content-Length: 0


<--- Received SIP request (1594 bytes) from UDP:172.22.160.1:50520 --->
INVITE sip:200@172.22.175.6;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.22.160.1:50520;branch=z9hG4bK-524287-1---586a8e20673f88d1;rport
Max-Forwards: 70
Contact: <sip:300@172.22.160.1:50520;transport=UDP>
To: <sip:200@172.22.175.6>
From: <sip:300@172.22.175.6;transport=UDP>;tag=aa1a801f
Call-ID: 3xVtqG5bnCTx5uXW5NYkgg..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Authorization: Digest username="300",realm="asterisk",nonce="1706288276/f2e7c6b7ac8a6a3a5299de722c9d0509",uri="sip:200@172.22.175.6;transport=UDP",response="529dccb9e34fbdbe0903bdab52318ac5",cnonce="ac47f84a911f6abc1d81d6c29d27aed2",nc=00000001,qop=auth,algorithm=MD5,opaque="2a17209b31a8b500"
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 629

v=0
o=Z 0 19320051 IN IP4 172.22.160.1
s=Z
c=IN IP4 172.22.160.1
t=0 0
m=audio 64337 RTP/AVPF 106 9 0 8 3 111 97 110 112 98 96 100 99 102
a=rtpmap:106 opus/48000/2
a=fmtp:106 minptime=20; useinbandfec=1
a=rtpmap:111 speex/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:110 speex/8000
a=rtpmap:112 speex/32000
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=rtpmap:100 telephone-event/16000
a=fmtp:100 0-16
a=rtpmap:99 telephone-event/32000
a=fmtp:99 0-16
a=rtpmap:102 G726-32/8000
a=sendrecv
a=rtcp-mux
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir

<--- Transmitting SIP response (306 bytes) to UDP:172.22.160.1:50520 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.160.1:50520;rport=50520;received=172.22.160.1;branch=z9hG4bK-524287-1---586a8e20673f88d1
Call-ID: 3xVtqG5bnCTx5uXW5NYkgg..
From: <sip:300@172.22.175.6>;tag=aa1a801f
To: <sip:200@172.22.175.6>
CSeq: 2 INVITE
Server: My Asterisk Server
Content-Length:  0


[Jan 26 22:27:56] ERROR[25778]: res_pjsip_session.c:937 handle_incoming_sdp:  300: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
<--- Transmitting SIP response (360 bytes) to UDP:172.22.160.1:50520 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 172.22.160.1:50520;rport=50520;received=172.22.160.1;branch=z9hG4bK-524287-1---586a8e20673f88d1
Call-ID: 3xVtqG5bnCTx5uXW5NYkgg..
From: <sip:300@172.22.175.6>;tag=aa1a801f
To: <sip:200@172.22.175.6>;tag=8cbf404b-b4a7-496c-8b5b-126682dc2b76
CSeq: 2 INVITE
Server: My Asterisk Server
Content-Length:  0


<--- Received SIP request (347 bytes) from UDP:172.22.160.1:50520 --->
ACK sip:200@172.22.175.6;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.22.160.1:50520;branch=z9hG4bK-524287-1---586a8e20673f88d1;rport
Max-Forwards: 70
To: <sip:200@172.22.175.6>;tag=8cbf404b-b4a7-496c-8b5b-126682dc2b76
From: <sip:300@172.22.175.6;transport=UDP>;tag=aa1a801f
Call-ID: 3xVtqG5bnCTx5uXW5NYkgg..
CSeq: 2 ACK
Content-Length: 0

300 is using AVPF, and you haven’t enabled it on the endpoint. Consult the documentation for endpoint for the proper option to enable AVPF support.

Thanks for the suggestion I have added use_avpf=yes in the 300 endpoint
The error is no longer there however getting

Everyone is busy/congested at this time

<--- Received SIP request (1302 bytes) from UDP:172.22.160.1:53056 --->
INVITE sip:200@172.22.175.6;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.22.160.1:53056;branch=z9hG4bK-524287-1---07382ab91466027c;rport
Max-Forwards: 70
Contact: <sip:300@172.22.160.1:53056;transport=UDP>
To: <sip:200@172.22.175.6>
From: <sip:300@172.22.175.6;transport=UDP>;tag=d44e6b4b
Call-ID: mngIRKYEiOUktf1dCHB1Mg..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 632

v=0
o=Z 0 22479254 IN IP4 172.22.160.1
s=Z
c=IN IP4 172.22.160.1
t=0 0
m=audio 50061 RTP/AVPF 106 9 0 8 3 111 97 110 112 98 101 100 99 102
a=rtpmap:106 opus/48000/2
a=fmtp:106 minptime=20; useinbandfec=1
a=rtpmap:111 speex/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:110 speex/8000
a=rtpmap:112 speex/32000
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:100 telephone-event/16000
a=fmtp:100 0-16
a=rtpmap:99 telephone-event/32000
a=fmtp:99 0-16
a=rtpmap:102 G726-32/8000
a=sendrecv
a=rtcp-mux
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir

<--- Transmitting SIP response (498 bytes) to UDP:172.22.160.1:53056 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.22.160.1:53056;rport=53056;received=172.22.160.1;branch=z9hG4bK-524287-1---07382ab91466027c
Call-ID: mngIRKYEiOUktf1dCHB1Mg..
From: <sip:300@172.22.175.6>;tag=d44e6b4b
To: <sip:200@172.22.175.6>;tag=z9hG4bK-524287-1---07382ab91466027c
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1706291434/618b50462a8b2033b7a6969d106a76fb",opaque="1c9aa8541c3626c3",algorithm=MD5,qop="auth"
Server: My Asterisk Server
Content-Length:  0


<--- Received SIP request (346 bytes) from UDP:172.22.160.1:53056 --->
ACK sip:200@172.22.175.6;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.22.160.1:53056;branch=z9hG4bK-524287-1---07382ab91466027c;rport
Max-Forwards: 70
To: <sip:200@172.22.175.6>;tag=z9hG4bK-524287-1---07382ab91466027c
From: <sip:300@172.22.175.6;transport=UDP>;tag=d44e6b4b
Call-ID: mngIRKYEiOUktf1dCHB1Mg..
CSeq: 1 ACK
Content-Length: 0


<--- Received SIP request (1597 bytes) from UDP:172.22.160.1:53056 --->
INVITE sip:200@172.22.175.6;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.22.160.1:53056;branch=z9hG4bK-524287-1---97380c8ff8874c8f;rport
Max-Forwards: 70
Contact: <sip:300@172.22.160.1:53056;transport=UDP>
To: <sip:200@172.22.175.6>
From: <sip:300@172.22.175.6;transport=UDP>;tag=d44e6b4b
Call-ID: mngIRKYEiOUktf1dCHB1Mg..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Authorization: Digest username="300",realm="asterisk",nonce="1706291434/618b50462a8b2033b7a6969d106a76fb",uri="sip:200@172.22.175.6;transport=UDP",response="928a6dcc97e6ab910fd46045227a4934",cnonce="2b57cbe1b2527fba0bd5c2591a5bbbdc",nc=00000001,qop=auth,algorithm=MD5,opaque="1c9aa8541c3626c3"
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 632

v=0
o=Z 0 22479254 IN IP4 172.22.160.1
s=Z
c=IN IP4 172.22.160.1
t=0 0
m=audio 50061 RTP/AVPF 106 9 0 8 3 111 97 110 112 98 101 100 99 102
a=rtpmap:106 opus/48000/2
a=fmtp:106 minptime=20; useinbandfec=1
a=rtpmap:111 speex/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:110 speex/8000
a=rtpmap:112 speex/32000
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:100 telephone-event/16000
a=fmtp:100 0-16
a=rtpmap:99 telephone-event/32000
a=fmtp:99 0-16
a=rtpmap:102 G726-32/8000
a=sendrecv
a=rtcp-mux
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir

<--- Transmitting SIP response (306 bytes) to UDP:172.22.160.1:53056 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.160.1:53056;rport=53056;received=172.22.160.1;branch=z9hG4bK-524287-1---97380c8ff8874c8f
Call-ID: mngIRKYEiOUktf1dCHB1Mg..
From: <sip:300@172.22.175.6>;tag=d44e6b4b
To: <sip:200@172.22.175.6>
CSeq: 2 INVITE
Server: My Asterisk Server
Content-Length:  0


    -- Executing [200@default:1] NoOp("PJSIP/300-00000000", "Calling: 200)") in new stack
    -- Executing [200@default:2] Dial("PJSIP/300-00000000", "PJSIP/200,30") in new stack
    -- Called PJSIP/200
<--- Transmitting SIP request (1546 bytes) to UDP:172.22.160.1:62753 --->
INVITE sip:200@172.22.160.1:62753;ob SIP/2.0
Via: SIP/2.0/UDP 172.22.175.6:5060;rport;branch=z9hG4bKPjb8095a33-3cce-4a3f-8928-405cd77c7f24
From: <sip:300@172.22.175.6>;tag=4addac22-f178-4d35-87a7-fa0cac98d9c6
To: <sip:200@172.22.160.1;ob>
Contact: <sip:asterisk@172.22.175.6:5060>
Call-ID: 211dfe81-7a4d-41f5-a197-dda9d6d409b3
CSeq: 23086 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: My Asterisk Server
Content-Type: application/sdp
Content-Length:   883

v=0
o=- 1080738435 1080738435 IN IP4 172.22.175.6
s=Asterisk
c=IN IP4 172.22.175.6
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio 16366 UDP/TLS/RTP/SAVPF 107 0 8 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 2B:A5:ED:AF:87:05:5E:38:2D:F0:0D:1B:B5:83:96:9D:FE:15:80:FD:D2:CD:8B:9D:81:4B:2B:20:1F:3F:E9:22
a=ice-ufrag:003c3469188a83417030bb2c6e3c8c80
a=ice-pwd:6c914225080a9f0c5653b85c280b52f3
a=candidate:Hac16af06 1 UDP 2130706431 172.22.175.6 16366 typ host
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:516156775 cname:cead6c70-4fb2-4718-b02e-af17d75ad631
a=msid:14e200d9-af49-4c86-b2be-42cac59f8ad8 3100e330-f475-444a-8ae7-7a9da5baa8aa
a=rtcp-fb:* transport-cc
a=mid:audio-0

<--- Received SIP response (383 bytes) from UDP:172.22.160.1:62753 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 172.22.175.6:5060;rport=5060;received=172.22.175.6;branch=z9hG4bKPjb8095a33-3cce-4a3f-8928-405cd77c7f24
Call-ID: 211dfe81-7a4d-41f5-a197-dda9d6d409b3
From: <sip:300@172.22.175.6>;tag=4addac22-f178-4d35-87a7-fa0cac98d9c6
To: <sip:200@172.22.160.1;ob>;tag=2d8ac8b11e4c4a7d8f9373689f30c5be
CSeq: 23086 INVITE
Content-Length:  0


<--- Transmitting SIP request (413 bytes) to UDP:172.22.160.1:62753 --->
ACK sip:200@172.22.160.1:62753;ob SIP/2.0
Via: SIP/2.0/UDP 172.22.175.6:5060;rport;branch=z9hG4bKPjb8095a33-3cce-4a3f-8928-405cd77c7f24
From: <sip:300@172.22.175.6>;tag=4addac22-f178-4d35-87a7-fa0cac98d9c6
To: <sip:200@172.22.160.1;ob>;tag=2d8ac8b11e4c4a7d8f9373689f30c5be
Call-ID: 211dfe81-7a4d-41f5-a197-dda9d6d409b3
CSeq: 23086 ACK
Max-Forwards: 70
User-Agent: My Asterisk Server
Content-Length:  0


  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [200@default:3] Hangup("PJSIP/300-00000000", "") in new stack
  == Spawn extension (default, 200, 3) exited non-zero on 'PJSIP/300-00000000'
<--- Transmitting SIP response (384 bytes) to UDP:172.22.160.1:53056 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 172.22.160.1:53056;rport=53056;received=172.22.160.1;branch=z9hG4bK-524287-1---97380c8ff8874c8f
Call-ID: mngIRKYEiOUktf1dCHB1Mg..
From: <sip:300@172.22.175.6>;tag=d44e6b4b
To: <sip:200@172.22.175.6>;tag=43e0b21e-8071-4f3d-ada2-0dab6407ac97
CSeq: 2 INVITE
Server: My Asterisk Server
Reason: Q.850;cause=58
Content-Length:  0


<--- Received SIP request (347 bytes) from UDP:172.22.160.1:53056 --->
ACK sip:200@172.22.175.6;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.22.160.1:53056;branch=z9hG4bK-524287-1---97380c8ff8874c8f;rport
Max-Forwards: 70
To: <sip:200@172.22.175.6>;tag=43e0b21e-8071-4f3d-ada2-0dab6407ac97
From: <sip:300@172.22.175.6;transport=UDP>;tag=d44e6b4b
Call-ID: mngIRKYEiOUktf1dCHB1Mg..
CSeq: 2 ACK
Content-Length: 0


<--- Received SIP request (982 bytes) from UDP:172.22.160.1:53056 --->
REGISTER sip:172.22.175.6;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.22.160.1:53056;branch=z9hG4bK-524287-1---f495b828e353710e;rport
Max-Forwards: 70
Contact: <sip:300@172.22.160.1:53056;rinstance=ad2aa9d627081b06;transport=UDP>
To: <sip:300@172.22.175.6;transport=UDP>
From: <sip:300@172.22.175.6;transport=UDP>;tag=ef5c1b56
Call-ID: Zo5mvo302WZVlXUReebP_Q..
CSeq: 3 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Authorization: Digest username="300",realm="asterisk",nonce="1706291420/2d8924aa0fe672aa322ed82121da0de5",uri="sip:172.22.175.6;transport=UDP",response="f3bcb5d13d7acff83c127f208ca9996f",cnonce="0d16a18b5fb383d613e072b04f78ef5c",nc=00000002,qop=auth,algorithm=MD5,opaque="492a1a944c37b423"
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0


<--- Transmitting SIP response (511 bytes) to UDP:172.22.160.1:53056 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.22.160.1:53056;rport=53056;received=172.22.160.1;branch=z9hG4bK-524287-1---f495b828e353710e
Call-ID: Zo5mvo302WZVlXUReebP_Q..
From: <sip:300@172.22.175.6>;tag=ef5c1b56
To: <sip:300@172.22.175.6>;tag=z9hG4bK-524287-1---f495b828e353710e
CSeq: 3 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1706291473/59b4f3dd2715159b2feddcadfa470cfb",opaque="1011ad0b745e0078",stale=true,algorithm=MD5,qop="auth"
Server: My Asterisk Server
Content-Length:  0


<--- Received SIP request (982 bytes) from UDP:172.22.160.1:53056 --->
REGISTER sip:172.22.175.6;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.22.160.1:53056;branch=z9hG4bK-524287-1---7c46257120319b00;rport
Max-Forwards: 70
Contact: <sip:300@172.22.160.1:53056;rinstance=ad2aa9d627081b06;transport=UDP>
To: <sip:300@172.22.175.6;transport=UDP>
From: <sip:300@172.22.175.6;transport=UDP>;tag=ef5c1b56
Call-ID: Zo5mvo302WZVlXUReebP_Q..
CSeq: 4 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Authorization: Digest username="300",realm="asterisk",nonce="1706291473/59b4f3dd2715159b2feddcadfa470cfb",uri="sip:172.22.175.6;transport=UDP",response="1e69e046719c9210b395d4757e29ae44",cnonce="50ddb49f1a51e0b4cba12144b5a51aec",nc=00000001,qop=auth,algorithm=MD5,opaque="1011ad0b745e0078"
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0


<--- Transmitting SIP response (485 bytes) to UDP:172.22.160.1:53056 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.160.1:53056;rport=53056;received=172.22.160.1;branch=z9hG4bK-524287-1---7c46257120319b00
Call-ID: Zo5mvo302WZVlXUReebP_Q..
From: <sip:300@172.22.175.6>;tag=ef5c1b56
To: <sip:300@172.22.175.6>;tag=z9hG4bK-524287-1---7c46257120319b00
CSeq: 4 REGISTER
Date: Fri, 26 Jan 2024 17:51:13 GMT
Contact: <sip:300@172.22.160.1:53056;transport=UDP;rinstance=ad2aa9d627081b06>;expires=59
Expires: 60
Server: My Asterisk Server
Content-Length:  0


Whatever you have called has rejected it. Why that is, no idea. Maybe it doesn’t speak WebRTC SDP.

@jcolp Thanks for the reply
I do not understand why I can not call from 300(a udp user) to 200(a webrtc user) but can call from 200(a webrtc user) to 300(a udp user)

I can only answer with the information given and the configuration given, and have no insight into the behavior of either of those.

@jcolp
I have only configured pjsip and the dialplan and nothing else so there shouldn’t be any more configurations

Well, you’ve apparently configured it in such a way that what you are calling doesn’t like it. You have to understand what the other side expects in order to configure things properly.

@jcolp I have one question
does the webrtc user only work in a browser phone and not with an external softphone.
the reason I am asking this question because when I using the same configuration and use the webrtc user in a browser phone which might us jssip its work fine however when I try to use in a softphone it always give some error or waring and does not allow to connect the call

If you configure an endpoint as WebRTC, it will only work when talking to something that is WebRTC. It will not work with a non-WebRTC device.

Just another question out of curiosity
So what will be a non-webrtc device?
is it a softphone in this case even though it has a valid webrtc user logged in it?

And what will be a webrtc device?
any valid webrtc user who is operated by a browser-based sip phone

Something that doesn’t implement the WebRTC specifications? A softphone most likely, that isn’t in a browser or using a browser in some way. I don’t know what “valid webrtc user logged in”. You can register if something has webrtc enabled on it and you aren’t using WebRTC, but calling is problematic.

When I refer to a valid webrtc user logged I mean the endpoint which has webrtc enabled and transport has wss(websocket secure)
And that endpoint is consumed by a softphone or a browser-based sip phone

There can’t be an or. If it’s configured as WebRTC, it can only be used by a browser-based WebRTC client.

@jcolp Thank you so much for your time I got my answer
now I can sleep with satisfaction.