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

jaydeepkasture · January 26, 2024, 4:58pm

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

jcolp · January 26, 2024, 5:06pm

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.

jaydeepkasture · January 26, 2024, 5:54pm

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

jcolp · January 26, 2024, 5:57pm

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

jaydeepkasture · January 26, 2024, 6:01pm

@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)

jcolp · January 26, 2024, 6:03pm

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

jaydeepkasture · January 26, 2024, 6:05pm

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

jcolp · January 26, 2024, 6:07pm

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.

jaydeepkasture · January 26, 2024, 6:13pm

@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

jcolp · January 26, 2024, 6:14pm

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.

jaydeepkasture · January 26, 2024, 6:22pm

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

jcolp · January 26, 2024, 6:31pm

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.

jaydeepkasture · January 26, 2024, 6:39pm

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

jcolp · January 26, 2024, 6:40pm

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

jaydeepkasture · January 26, 2024, 6:45pm

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

system · February 25, 2024, 6:46pm

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.

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