I Built a Real-Time Video Calling App Using WebRTC in React Native, And It Was Harder Than I Expected

Source: Dev.to

Introduction

Most developers have used apps like Zoom, Google Meet, or WhatsApp calls.
But building one? That’s a completely different story.

When I started working on my React Native WebRTC app, I thought: “It’s just video streaming, right?” I was very wrong.

The Moment I Realized This Isn’t Just Another App

In a normal app:

You send a request → server responds

In WebRTC:

Two devices talk directly
No middleman.

Just two peers trying to:

• Discover each other
• Negotiate connection
• Exchange network details
• Stream audio/video in real-time

That’s when it hit me: this isn’t purely a frontend or backend problem.

What WebRTC Actually Does

WebRTC (Web Real-Time Communication) allows devices to communicate peer‑to‑peer with ultra‑low latency.

• Your phone can directly stream video to another device without routing media through a server.
• Built‑in encryption keeps the communication secure.

Powerful, but also complex.

The Architecture Behind My WebRTC App

1. Client Layer (React Native)

• Captures camera & microphone
• Displays local & remote video
• Handles UI for calling

React Native made it easier to build cross‑platform apps for iOS and Android while still using native WebRTC performance. The UI was the easy part.

2. Signaling Server (The Unsung Hero)

“WebRTC is peer‑to‑peer, so no server needed.” – Wrong.

A signaling server is required to:

• Help users find each other
• Exchange connection data (SDP)
• Share ICE candidates

WebRTC does not define how signaling works, so you have to build it yourself. In my project, this became the coordination layer.

3. Peer Connection (The Core Engine)

After signaling:

• A peer connection is created
• Devices exchange Offer, Answer, and ICE candidates

The connection shifts from server‑mediated to direct device‑to‑device communication.

4. NAT Traversal (The Hidden Complexity)

Real‑world networks are messy. Devices sit behind routers, firewalls, and NATs. WebRTC uses:

• STUN servers → Find your public IP
• TURN servers → Relay data if a direct connection fails

Without these, many connections simply wouldn’t work.

5. Real‑Time Media Flow

Once the peer connection is established:

• Audio & video streams flow directly between peers
• No backend is involved in media transfer
• Latency stays extremely low

This is why WebRTC is used in video calls, live collaboration, and telemedicine apps.

The Full Call Flow

1. User A starts a call.
2. Signaling server sends an offer to User B.
3. User B responds with an answer.
4. Both exchange ICE candidates.
5. Peer connection is established.
6. Media streams flow directly between devices.

Not simple, but beautiful.

What Made This Project Challenging

1. Debugging Is Painful

You’re not just debugging functions; you’re debugging:

• Network states
• ICE failures
• Connection negotiation

2. Asynchronous Chaos

Everything happens in events (offers, answers, candidates, streams). Miss one step → the connection fails silently.

3. Documentation Is Scattered

WebRTC isn’t beginner‑friendly. You don’t “learn it once”; you experience it over time.

What This Project Taught Me

Before this project I thought: “Apps are about APIs and UI.”
Now I know some systems live below that layer.

Key takeaways

• Real‑time systems are fundamentally different.
• Architecture matters more than code.
• Networking knowledge is underrated.
• Peer‑to‑peer is powerful but complex.

From App Developer to System Thinker

I stopped asking, “How do I build this feature?” and started asking, “How does communication actually happen?”

That shift separates developers from engineers.

Final Thought

Building a WebRTC app isn’t just about video calling; it’s about understanding:

• Communication protocols
• Network behavior
• Real‑time systems

Once you grasp that, you stop seeing apps as screens and start seeing them as systems.

Links

• GitHub:
• Portfolio: