libdatachannel VS Pion WebRTC

Node.js node JavaScript runtime itself provides a means to interface with native applications. Plenty of people write their own WebSocket or WebRTC data channel libraries https://github.com/paullouisageneau/libdatachannel.

Fourtunately some technologies and API's are extensible enough to do whatever you want in spite of and to overcome limitations; e.g., WebRTC (particularly data channels https://github.com/paullouisageneau/libdatachannel), Native Messaging, WebTransport, fetch(), Trasnferable Streams, Media Capture Transform, File API, File System Access API, Web extensions.

To be fair, C++ has libdatachannel which I like using too.

https://github.com/paullouisageneau/libdatachannel ?

Or use something specific like libdatachannel https://libdatachannel.org/

If you’re using WebRTC data channels in a client server config then infra is no more complex than a regular setup because the server will almost certainly have an external facing IP address.

People should really try using WebRTC as well because you can get the full pipeline setup for a toy demo in a couple of slow days. I did whilst also writing my own signalling server in Go without having used Go before!

It’s certainly not any more complex than rolling all the stuff the article is talking about on top of raw UDP.

This does use a real WebRTC stack (specifically libdatachannel - https://libdatachannel.org). Under the hood this is making real WebRTC connections.

But I guess you mean why it doesn't mock using real peers in the browser where the tests are happening, using the browser's existing stack there?

There's a few reasons I've avoided that:

* Primarily: I want to do things that the browser APIs won't let you do. E.g. sending invalid data, or intentionally causing various types of errors. The browser APIs are designed to stop you doing the wrong thing, by keeping the raw data at arm's length. Very sensible for normal WebRTC usage, but bad for testing edge cases and breaking things.

Building WebRTC tends to be hell. If your use case is simple enough, it might be interesting to switch to an alternative C++ library like libdatachannel (I'm the author).

You can, in theory, get UDP-like characteristics out of a different web API - with a WebRTC datachannel configured to be unreliable and unordered. I haven't seen anybody use that API just because it's a pretty big pain to use. The developer of Agar IO called it too hard to use in a HackerNews comment, but nowadays there's simpler libraries than the ones he had in 2016 like webrtc-unreliable and libdatachannel. You still need to run a STUN/TURN server and the integration is easier than it was back then, but still much harder than raw UDP sockets though.

exit(EXIT_SUCCESS); } char message[message_length]; fread(message, message_length, sizeof(char), stdin); string m(message, message + sizeof message / sizeof message[0]); return json::parse(m); } // Encode a message for transmission, given its content. message_t encode_message(json content) { string encoded_content = content.dump(); message_t m; m.content = encoded_content; m.length = (uint32_t)encoded_content.length(); return m; } // Send an encoded message to stdout. void send_message(message_t encoded_message) { char *raw_length = reinterpret_cast(&encoded_message.length); fwrite(raw_length, 4, sizeof(char), stdout); fwrite(encoded_message.content.c_str(), encoded_message.length, sizeof(char), stdout); fflush(stdout); } int main(int argc, char *argv[]) { while (true) { json message = get_message(); auto command = message.get(); // https://github.com/paullouisageneau/libdatachannel/issues/544#issuecomment-1024739214 // https://www.reddit.com/r/cpp\_questions/comments/swr1tk/comment/hxral9f/ FILE *pipe = popen(command.c_str(), "r"); while (true) { size_t bufferSize = 1024; byte buffer[bufferSize]; size_t count; while ((count = fread(buffer, 1, bufferSize, pipe)) > 0) { stringstream data; // S16NE PCM from parec to integers in JSON array data << "["; for (size_t i = 0; i < count; i++) { data << (int)buffer[i]; if (i < count - 1) { data << ","; } } data << "]"; // send JSON array to Native Messaging local client send_message(encode_message(data.str())); } } } }

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.