Repro fiddle: https://jsfiddle.net/xL95p1t7/

This is what fails:
https://cs.chromium.org/chromium/src/third_party/webrtc/pc/webrtcsdp.cc?type=cs&sq=package:chromium&l=2526

I was confused because the codecs 9, 0 and 8 does have a=rtpmap lines, but I debugged and what fails is codec 111.
There's a "a=rtcp-fb:111 transport-cc" line but 111 is not listed in m=audio. When I removed it, the promise resolves.

Fiddle that resolves: https://jsfiddle.net/xL95p1t7/1/

So I think this is not a bug, but invalid SDP.
Or what do you think deadbeef@, should the promise resolve with that line?

Are you deleting the comments or is there an issue with the site? Someone else mentioned their comments auto-deleting. I can see from the email thread you wrote:

'v=0\r\n' + 
'o=- 964396985 1 IN IP4 151.237.238.50\r\n' + 
's=-\r\n' + 
't=0 0\r\n' + 
'a=ice-lite\r\n' + 
'm=audio 65362 RTP/SAVPF 111 103 104 9 0 8 106 105 13 112 113 126\r\n' + 
'c=IN IP4 151.237.238.50\r\n' + 
'a=rtpmap:111 opus/48000/2\r\n' + 
'a=fmtp:111 minptime=10;useinbandfec=1\r\n' + 
'a=rtpmap:103 ISAC/16000\r\n' + 
'a=rtpmap:104 ISAC/32000\r\n' + 
'a=rtpmap:9 G722/8000\r\n' + 
'a=rtpmap:0 PCMU/8000\r\n' + 
'a=rtpmap:8 PCMA/8000\r\n' + 
'a=rtpmap:106 CN/32000\r\n' + 
'a=rtpmap:105 CN/16000\r\n' + 
'a=rtpmap:13 CN/8000\r\n' + 
'a=rtpmap:112 telephone-event/32000\r\n' + 
'a=rtpmap:113 telephone-event/16000\r\n' + 
'a=rtpmap:126 telephone-event/8000\r\n' + 
'a=sendrecv\r\n' + 
'a=rtcp-mux\r\n' + 
'a=ice-options:trickle\r\n' + 
'a=mid:audio\r\n' + 
'a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\n' + 
'a=rtcp-fb:111 transport-cc\r\n' + 
'a=ssrc:3842950873 cname:ofV5OEGGCprVKDVp\r\n' + 
'a=ssrc:3842950873 msid:dsLGt81LRTryFupx7UKIDtioaby5c9N2d5gK 874e575f-a7db-47ba-bbc8-6d06d3d9e3bb\r\n' + 
'a=ssrc:3842950873 mslabel:dsLGt81LRTryFupx7UKIDtioaby5c9N2d5gK\r\n' + 
'a=ssrc:3842950873 label:874e575f-a7db-47ba-bbc8-6d06d3d9e3bb\r\n' + 
'a=setup:actpass\r\n' + 
'a=fingerprint:sha-256 FD:0D:48:C8:3A:A7:7B:80:0B:AF:3D:C8:0D:65:0B:80:3B:9A:08:D8:52:BC:64:82:14:0F:58:D2:98:9D:3C:1A\r\n' + 
'a=ice-ufrag:GIyd\r\n' + 
'a=ice-pwd:+CYWu2zMvv+dtAA0QIrzgqQ99Tm4b9\r\n' + 
'a=candidate:8Os/q 1 UDP 659136 151.237.238.50 65362 typ host\r\n' + 
'a=candidate:8Os/q 2 UDP 659134 151.237.238.50 65363 typ host\r\n',

So with the 111 stuff added back in so that it is a valid line I am still getting the following rejection:

jssip.min.js?v100.0.0:15 rtcninja:ERROR:RTCPeerConnection setRemoteDescription() | error: +1ms DOMException: Failed to parse SessionDescription. m=audio 65362 RTP/SAVPF 111 103 104 9 0 8 106 105 13 113 126 Invalid value: .
s @ jssip.min.js?v100.0.0:15
(anonymous) @ jssip.min.js?v100.0.0:16
r.setRemoteDescription @ jssip.min.js?v100.0.0:16
t @ jssip.min.js?v100.0.0:13
(anonymous) @ jssip.min.js?v100.0.0:16

Deleted cause I was being dumb, there was a line in my code I didn't notice before it was passing the SDP that was editing out one of the m=audio values but leaving double spaces where it removed numbers. That created genuinely invalid SDP. I didn't initially notice that the error message I was getting this time was more informative compared to the 111 thing (that was a separate issue where the PBX is not properly stripping codecs).

I do think that the m=audio line with two spaces used to be accepted but its definitely not a bug if its not.

Alright cool, thanks!