Pion WebRTC VS livekit-server

You can run all your WebRTC traffic off a single port. You use the remote 3 tuple (IP, Port, Protocol) to demux traffic.

https://github.com/pion/webrtc/tree/master/examples/ice-sing... is one example of that.

I originally went the same route as you, and found that https://github.com/pion/webrtc is probably the best package out there for webrtc. I learned go just for it, and it paid off tenfold. Less memory, more connections, lower latency.

The "Mediasoup" project provides a high level JavaScript/TypeScript interface to the WebRTC APIs. The core logic of this project is implemented in C++/Rust. Consider taking a look at the project if you want an easy-to-use library instead of the low-level libWebRTC APIs. A notable project to mention is the Pion/webrtc project which has a Golang implementation of the WebRTC API. Of course, we should mention the rust port WebRTC.rs. Let’s keep all the rustaceans happy too!

What latency are you trying to do? Will the professor being communicating with the students while doing this? Will the students all have the same bandwidth, or will you want multiple renditions (low, med, high quality levels)?

If you want AV1 you will not be able to use RTMP. The protocol is orphaned/deprecated, so avoid if possible!

If I was building it this is what I would do, and my reasoning.

* For capture + encoding I would use OBS. You will want to use something that is easy for users to install configure. Professors will also have lots of custom requirements when it comes to layout etc... it will be tempting to do a ffmpeg command directly, but it will fall apart quick I believe.

* To get AV1 out of OBS I would use FFMPEG output. I would have it send RTP. RTP is used to carry video in a sub-second manner. This is the same protocol that WebRTC uses. You know have AV1 + low latency.

* Then for users to watch I would use WebRTC. That will allow them to watch in their web browser. Conceptually it will be like this https://github.com/pion/webrtc/tree/master/examples/rtp-to-w... this takes the RTP packets and puts them in the browser.

Lots of great projects exist that you could use for 'RTP -> WebRTC' like https://galene.org/ and https://livekit.io/ I would suggest checking them all out!

If you have more questions/want to talk to people in the video space always happy to chat on https://pion.ly/slack :)

For streaming video content your options would be HLS or WebRTC, maybe look into these gwuhaolin/livego and pion/webrtc.

WebRTC GitHub Forum --- Use your GitHub account to join WebRTC-related forums or start a discussion of your own.

Do you still see challenges with doing WebRTC on a server? I work on https://github.com/pion/webrtc so would love to hear what could be better :)

We've just released version 0.5.0 of the webrtccrate. This crate is a port of a Go project called Pion. It's a complete implementation of WebRTC in Rust, allowing you to build backends for media applications.

Check out livekit, have been working with it for 2 months and you can quite easily set up things. Really promising project: https://docs.livekit.io

Check out our docs on how the platform can support multiple (separate) rooms with 100s of members. We know how important good documentation is, so if there's any gaps you see lemme know and we will add to it asap!!

I joined an OS project called LiveKit (GH repo), a Go-based infrastructure intended to make it easy to embed real-time audio and video communications directly in any app. The world is growing increasingly remote and connected online (esp with COVID), & realtime audio and video is becoming the cornerstone of how we connect and communicate with others around the world.