A Practical Guide to How Real-Time Chat & Video Platforms Work

#webrtc #webdev #tech #chat

(Plus some lessons I learned while building my own WebRTC SDK at weblivehub.com)

When developers first step into real-time communication — random chat sites, group chat rooms, or video chat — they often jump straight into WebRTC tutorials but still feel lost.

That's because WebRTC is only 20% of the full system.
The remaining 80% is all the architecture around it.

This article breaks down the pieces you actually need and shares practical lessons I learned while developing a WebRTC-based SDK (weblivehub.com). Even if you don't use my product, I hope the real-world experience helps you understand the whole picture.

1. WebRTC is transport, not the full product
WebRTC itself only provides three capabilities:

audio transport
video transport
data channel transport

That’s it.

It does not handle:
random matching
room creation
group chat
message history
identity or sessions
signaling
scaling to many users

All those parts must be built separately.

2. Features like “Stranger Chat” or “Group Chat” come from backend logic, not WebRTC

Let’s break down three common features.

✔ Stranger (Random) Chat

You need:

a matchmaking queue
WebSocket or HTTP events to notify “you’ve been matched”
and only then exchange WebRTC Offer/Answer

WebRTC does not match users for you.
The entire user flow is backend-driven.

✔ Group Chat (text)

Usually built with:

WebSocket + Redis
publish/subscribe patterns
room membership logic
message broadcast rules

WebRTC is not involved at all.

✔ Multi-person video chat

You need either:

SFU (mediasoup, ion-sfu, livekit), or
P2P for very small groups (2–4 people)

From my real tests on weblivehub.com:

1–4 people → P2P is fine
5+ people → use SFU or bandwidth explodes

And again—WebRTC alone cannot do “multi-user rooms.”
You must build the architecture around it.

3. The most confusing WebRTC parts: signaling & NAT traversal

WebRTC can't connect on its own.
It needs to exchange three things:

Offer
Answer
ICE candidates

This requires signaling, which can be implemented using:

WebSocket
HTTP polling
MQTT
Anything capable of sending messages

My own SDK uses WebSocket + distributed signaling, mostly to reduce friction for developers.

4. The real challenge is not WebRTC—it’s everything surrounding it

Here’s what a full platform needs:

The actual “WebRTC API coding” is just a small slice of the entire system.
That’s why I eventually created weblivehub.com—to encapsulate the repeated work.

5. If you want to build something like Omegle / TalkRush / random chat sites

You’ll need these skills:

Required knowledge

WebRTC basics (Offer, Answer, ICE)
WebSocket backend
Room/session logic
Queues for random pairing
TURN server (coturn)
Light UI (HTML/CSS/JS)

Typical architecture flow

User connects to WebSocket
They enter a matchmaking queue
Server pairs two users
Server tells both sides to exchange Offer/Answer
WebRTC PeerConnection is created
Chat messages + UI logic are layered on top

You’ll notice WebRTC is only one step in the middle.

6. If you want to speed up development
(This is why I built weblivehub.com)

Many developers get stuck on the same parts:

signaling
rooms
matching
SFU integration
reconnection handling

So I packaged these into a reusable SDK.
You can explore the demos or architecture breakdown—
even if you don’t use it, studying a working system helps you skip months of trial and error.

Final Thoughts

WebRTC is an amazing technology, but it does not create full products by itself. Understanding the system around it is the key to building: