What is WebRTC and How Does It Work?

 WebRTC or Web Real-Time Connection is an open-source project that allows real-time communication between devices and web browsers, You may confuse it with WebSockets, which also allow users to communicate in real-time. But what is the difference? Is WebRTC a websocket? What is WebRTC and why is it used in many places in place of a web socket? This is what we're gonna learn today.

WebSocket vs WebRTC

Feature	WebSocket	WebRTC
Communication Model	Client-Server	Peer-to-Peer
Primary Use	Real-time data exchange between client and server	Real-time audio, video, and data between peers
Latency	Low, but depends on server routing	Generally lower due to direct P2P connection
Data Types	Text and binary	Media streams (audio/video) and data channels
Setup Complexity	Relatively simple	More complex (requires signaling, NAT traversal)
Server Requirements	Requires a server to manage connections and relay messages	An initial signaling server is needed, but media/data flows directly between peers

Use Cases comparison:

Both of them have different use cases that they are designed for, and since they're not using the same connection mode;

WebSocket is mostly used for:

1. Chat Applications: Real-time messaging between users.
2. Live Notifications: Instant updates like stock tickers or social media feeds.
3. Online Gaming: Real-time game state synchronization.
4. Collaborative Tools: Real-time document editing or whiteboards.

While WebRTC is used for:

1. Video Conferencing: Platforms like Zoom or Google Meet leverage WebRTC for real-time video and audio.
2. Voice Calls: VoIP applications using browsers.
3. P2P File Sharing: Direct transfer of files between users.
4. Interactive Gaming: Real-time multiplayer games requiring low latency.

When to Use Each Technology

Use WebSocket when:

1. You need real-time communication between a client and a server.
2. Implementing features like live chat, real-time dashboards, or notifications.
3. The communication does not involve high-bandwidth media streams.

Use WebRTC when:

1. You require real-time peer-to-peer media communication (audio/video).
2. Building applications like video conferencing, live streaming, or P2P file sharing.
3. You want to reduce server load by handling media directly between clients.

 

What is WebRTC anyway?

Okay, so to explain it simply, it's a technology that provides a collection of APIs and protocols that allow for Peer-to-peer (P2P) exchange of audio, video, and data in general. It was mainly designed to handle video, audio, and generally heavy real-time communication, unlike WebSockets that are used for mostly text, and binary, and are not as frequent. 

synchronizing video or audio online, particularly on a server, can be resource-consuming due to the potential for high traffic from millions of users. To address this issue, WebRTC was developed. With WebRTC, a server is only needed for the initial signaling process, while the actual communication occurs through Peer-to-Peer connections. I've talked about P2P more in this What is Peer-to-peer (P2P) network? if you're interested.

 Key WebRTC APIs are:

1. MediaStream API (getUserMedia): For capturing audio and video from local devices.
2. RTCPeerConnection API: For managing connections between peers.
3. RTCDataChannel API: For exchanging arbitrary data between peers.

Why would you use WebRTC instead of Websockets?

WebRTC's Main purpose in its life is to handle real-time audio and video streams, it has built-in support for codecs, synchronization, and media quality adjustments. While WebSockets are not optimized for media. In addition to that, Peer2Peer communication allows for lower latency and low resource consumption. 

On a WebSocket, as I mentioned earlier, each bit and frame of a video will have to pass through the server, which is very inefficient. WebRTC offers data channels that are optimized for real-time data transfer, with lower latency than Websockets. While WebSockets are capable of handling real-time data, it's not to the level that WebRTC was built for.

Conclusion

In conclusion, You may need to use them together, for example, you need to signal the WebRTC to start the data exchange and establish the peer-to-peer connection. You could even use them in parallel, WebRTC handles the video/audio or file transfer while WebSockets handles anything that is not that. that's all there is to it. I guess, Thank you for reading. You can check out official docs whenever you think you need to get in-depth in it.