Video and audio stream - server to clients only

Question

Is there a way to stream a video and audio on a website just to the clients, using a camera installed on the server - for instance, like youtube does ?

I've started reading webrtc, but if I use webrtc I should create a stun/turn server and other things, which for one way stream I think is not necessary (this is just my understanding of the things..) because I don't need anything from the clients, literally, neither their video, or audio..

So is there a way to achieve this using html5, streaming just in one direction:

server (camera) -> clients

Is there something about this out there, or should I stick with webrtc ?

Answer 1

I'm going to explain a possible solution for this scenario, there might be others, but I hope mine gives you a rough idea of how you could do it and a start point to explore more about the amazing possibilities of WebRTC. Please let me know if something is not understood.

So, WebRTC is a free, open project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. Sweet, that is: WebRTC has a quite good browser support (not in every browser though, Safari just started supporting it a month ago with Safari 11). But in this case we want to use WebRTC in the server side. At the end of the day we can still think about peer-to-peer real time communication, where one of our peers is the server.

I don't know if you are familiar with Node.js, but I recommend you to write your Server app with it (<3 Javascript!):

• There are a few libraries that wrap WebRTC functionality to be used in the server side, like node-webrtc and node-rtc-peer-connection.

• But I recommend you to to take a look at electron-werbrtc, since the others might be using deprecated methods or be incomplete. electron-webrtc runs a headless Electron client in the background to use Chromium's built-in WebRTC implementation. So with it you should be able to access the Camera in your server and create a stream to be served to the other peer (the browser).

All above would be the WebRTC related tasks, in this case: streaming video peer(server)-to-peer(browser).

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Now, let's talk about signaling process, stun and turn.

• Signaling: imagine now a scenario peer-to-peer with 2 browsers, they want to establish a direct connection and stream video and audio between each other. But they don't know each other, like if I don't know your home address, I can't send you a letter. So they need a service that helps them know each other, so they can have the other's IP. This should be done by what is called "a signaling server". If somehow you know the other peer IP, you wouldn't need a signaling server.

• STUN/TURN: the scheme above works perfectly in a local area network where each peer has its own IP address and there are no firewalls and routers between them. But otherwise, you can have peers behind a NAT or firewalls, and then your signaling server won't be able to make both peers to discover themselves. If you have peers behind a NAT, you'll need a STUN server, and if you have peers behind firewalls you'll need a TURN server. This is a bit simplified, but I just want you to have the general picture of when you might need STUN/TURN servers.

To better understand Signaling, STUN and TURN, there is a very graphic article that explains them perfectly.

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Now, for your scenario:

• I think you prob don't need STUN/TURN servers and also you prob don't need to implement the signaling process, because the browsers that are supposed to receive the stream from the server will know that server address, right? So they can establish a WebRTC connection with it.
• EDIT: it is likely that you will need to implement some sort of handshake between the server and the clients (browsers), so this will be the signaling process. This is not part of WebRTC and this is why you need to implement it yourself. As I said, it is the way 2 peers can discover each other, but they also exchange information as their local media conditions, like codecs, resolutions they can handle, etc. For your case, your signaling server could be hosted in the same server you use to strea: you can build a small node.js app that runs there and that manages all the signaling process easily, it is not a big deal. I recommend you to read this article, and specially the section "How can I build a signaling service?". In general all WebRTC articles from that site are very helpful.

Does this make sense to you? I think with it you can start digging a little bit more and see if with this is enough or you need to implement more stuff. Hope it helps!