I am trying to create a multiple-user audio streaming web app and I have been making research on how to go about this.
I got a clue in using WEBRTC or RTMP and audio encoders, but I don't really understand these technologies and which will be better for broadcasting clear HD audio sound to millions of users, please I need any clue on how to go about this to allow multiple users broadcast at the same time on the web app.
I am using JavaScript for this project.
RTMP is a dead protocol that was used for Flash. The only reason you see it today is for support for legacy encoders (usually on the video side) that don't support modern protocols.
WebRTC is a whole technology stack designed for streaming the lowest possible latency of media streams. Its original goal was peer-to-peer streaming, but it has picked up in usage for server-to-client streaming. If you're streaming to millions of users, you'll undoubtedly not be using WebRTC.
With that kind of traffic, you'll probably want to use existing file-based CDNs, which are a lot more affordable than WebRTC streaming. To utilize these CDNs, you'll need a segmented streaming protocol such as HLS or DASH. Due to Apple's anti-competitive policies on iOS, you're stuck with HLS between the two (unless you choose to support both).
I should say... a million listeners is a lot. Those sorts of numbers are very rare online. If you have more reasonable expectations, you might also consider Icecast, which is very easy to set up and has a lot of affordable hosting available.