WebRTC is encrypted.
WebRTC is end-to-end encrypted by default. There is a signaling server that helps establishing the connections between users in a room, but after that the communication is encrypted. Also those TURN and STUN servers are only required for technical reasons to get peer-to-peer working. So no content is ever passed unencrypted.
That's the difference to other services like Zoom and Jitsi, where a server in the middle is receiving the video streams unencrypted and then redistributes. Although Jitsi is adding encryption support for that as well soon.
You're not sending unencrypted streams to Jitsi, as it uses hop-by-hop encryption, meaning you send an encrypted stream to the server where it gets decrypted and re-encrypted for the rest of the participants. At no point you are sending "plaintext" data. What you might have heard is that Jitsi will be introducing end-to-end encryption, but it doesn't mean it doesn't have encryption at the moment.[1]
I think it's a misnomer to say WebRTC end to end encrypted by default.
If you cannot guarantee end to end encryption during the negotiation phase, you cannot trust that the party you're encrypting for/receiving encrypted content from is the actual party you are expecting to communicate with, hence the need to independently verify over a trusted channel whether the fingerprint in the SDP offer/answer matches.
After the negotiation is complete, the streams will indeed be encrypted, but if the offer/answer SDP has been tampered with by the signaling server, which is under the control of a rogue/compelled provider, and another DTLS certificate has been inserted, the encryption is useless as far as complete communication security is concerned.
I believe it is correct to say, that WebRTC is end-to-end encrypted. What you are referring to is, if you can trust to have a direct connection to the other participant and that you can make sure she is the person she pretends to be. This layer is indeed missing and will be added [2]
I would very much appreciate your help on the project to eliminate those concerns.
What this means is that the packets are encrypted between you and the video bridge and the packets travel unencrypted within the videobridge infrastructure, but at no point does the outside world see the unencrypted stream. Talking to the Jitsi video bridge is just a regular WebRTC session as far as clients are concerned (so any DTLS encryption WebRTC has, the Jitsi call has), I know this because before they had a decent SDK I had to manually wedge my WebRTC calls into it.
If you're looking to add e2ee to your project, you're pretty much left to choose between Olm and Signal Protocol, which are quite similar, but libsignal is GPLv3 while libolm is Apache 2.0.
I've used Olm more intensively over the last two years. While it works, it's great, when it randomly wedges a session for no apparent reason you feel like throwing it out the window.
edit: it looks like you can find info in the RTCSessionDescription.sdp property. The remoteDescription and localDescription should each have key fingerprints in lines with "a=fingerprint:..."
It behaves quite hilariously when you switch to e2e encryption after a normal connection - the other party gets random noise for a little while.
https://www.chromestatus.com/feature/6321945865879552
Hopefully this will be implemented in other browsers as well soon.