Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)

Welcome everyone. I get an error in Asterisk Log Files when I try to call.

[2023-03-31 14:08:41] ERROR[10349] res_pjsip_session.c:  1000: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)

The error will not occur if you call from the UCP phone widget. Specifically, when calling via SIPML5 or JsSip/sip.js.

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WSS 3s347lasgs7c.invalid;branch=z9hG4bK1512593
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=98gj505r84
CSeq: 2 INVITE
Call-ID: pcnrn80v9e1l0k7jlf44
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username="1000", realm="asterisk", nonce="1679985883/56e15f6ae196f19b53ca1fcdec9e6d30", uri="sip:[email protected]", response="c95c5b820bd84de465323ca0ff96e00e", opaque="1acc15291a227f92", qop=auth, cnonce="jas4e2saf41e", nc=00000001
Contact: <sip:[email protected];transport=ws;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.21.1
Content-Type: application/sdp
Content-Length: 1877

v=0
o=- 2559441414679085464 2 IN IP4 xxx.xxx.xxx.xxx
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS a78c2747-b6c7-401c-9db8-a6ad6192d45c
m=audio 54274 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 xxx.xxx.xxx.xxx
a=rtcp:9 IN IP4 x.x.x.x
a=candidate:3614347725 1 udp 2122260223 x.x.x.x 54273 typ host generation 0 network-id 1
a=candidate:261984363 1 udp 2122194687 x.x.x.x 54274 typ host generation 0 network-id 2 network-cost 10
a=candidate:700728665 1 tcp 1518280447 x.x.x.x 9 typ host tcptype active generation 0 network-id 1
a=candidate:4046931199 1 tcp 1518214911 x.x.x.x 9 typ host tcptype active generation 0 network-id 2 network-cost 10
a=candidate:777302348 1 udp 1685987071 x.x.x.x 54274 typ srflx raddr x.x.x.x rport 54274 generation 0 network-id 2 network-cost 10
a=ice-ufrag:obWx
a=ice-pwd:rkwLcolQquFf8jYk+zkOo41K
a=ice-options:trickle
a=fingerprint:sha-256 7B:9D:EE:D2:9A:B0:A1:A9:46:C2:78:15:75:B3:CC:A8:3D:A4:01:DE:E3:F1:98:EA:DE:3D:A8:6D:88:8C:6E:84
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
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:a78c2747-b6c7-401c-9db8-a6ad6192d45c 7e23e343-91df-4fae-8d92-855cd3f7c630
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:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
a=ssrc:3693460245 cname:6YBkZCJp7kfGeo1o
a=ssrc:3693460245 msid:a78c2747-b6c7-401c-9db8-a6ad6192d45c 7e23e343-91df-4fae-8d92-855cd3f7c630


rollbar.umd.min.js:1 Tue Mar 28 2023 12:44:43| sip.Transport | Received WebSocket text message:

SIP/2.0 100 Trying
Via: SIP/2.0/WSS 3s347lasgs7c.invalid;rport=64700;received=x.x.x.x;branch=z9hG4bK1512593
Call-ID: pcnrn80v9e1l0k7jlf44
From: <sip:[email protected]>;tag=98gj505r84
To: <sip:[email protected]>
CSeq: 2 INVITE
Server: FPBX-16.0.39(18.16.0)
Content-Length:  0



rollbar.umd.min.js:1 Tue Mar 28 2023 12:44:43 | sip.Inviter | Inviter.onTrying
rollbar.umd.min.js:1 Tue Mar 28 2023 12:44:43 | sip.Transport | Received WebSocket text message:

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WSS 3s347lasgs7c.invalid;rport=64700;received=x.x.x.x;branch=z9hG4bK1512593
Call-ID: pcnrn80v9e1l0k7jlf44
From: <sip:[email protected]>;tag=98gj505r84
To: <sip:[email protected]>;tag=c5eaa980-b1d9-4d5e-b14d-b2f950bfc78b
CSeq: 2 INVITE
Server: FPBX-16.0.39(18.16.0)
Content-Length:  0

That will occur if you are trying to use a configured endpoint that is not for WebRTC. The SDP from a normal one and from WebRTC are different and incompatible, and require different configuration on the server side.

Hi, thanks for the response. In my case these steps helped me

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