TCP vs UDP — Low-Level Network Internals Behind Your API Calls

As developers, we often build and consume APIs without diving into the how of data transmission. But when things slow down, break unexpectedly, or you're building real-time features like video chat or live collaboration—knowing what's happening under the hood becomes invaluable.

This article explores the internals of TCP vs UDP, how they affect your API calls and app performance, and how protocols like WebRTC strategically use both to deliver real-time magic in modern web apps.

TCP vs UDP: The Core Differences

TCP (Transmission Control Protocol) and UDP (User Datagram Protocol) are transport layer protocols responsible for delivering data between clients and servers. While both handle data transmission, they prioritize different aspects of performance:

Think of TCP as a reliable delivery service that tracks every package. UDP is more like a broadcast—fast, but no promises it’ll get there.

How It Affects Your API Layer

✅ TCP for RESTful APIs
Most HTTP-based APIs use TCP underneath. Here’s what it means for your app:

• Reliable communication with retry logic
• Handshake before data transfer
• Slower startup latency (especially in poor networks)

MERN Example: Your Express.js backend serving GET /api/data via HTTPS rides entirely on TCP. Axios, Fetch, and browsers all rely on TCP to ensure delivery, retries, and order.

⚡ UDP in Real-Time Applications (And Where WebRTC Comes In)

When performance must be real-time—like in chat, video, or live location updates—latency beats reliability. That’s where UDP and WebRTC shine.

WebRTC, the backbone of video chat in tools like Google Meet, Discord, and Zoom, uses UDP as its primary transport. But here's the twist:

🔀 WebRTC actually uses multiple protocols:

• STUN (UDP-based): For NAT traversal
• TURN (TCP or UDP): For relaying media if peer-to-peer fails
• DTLS/SRTP over UDP: For encrypted audio/video streams

WebRTC selects the fastest path using ICE (Interactive Connectivity Establishment), favoring UDP for real-time speed, but falling back to TCP if needed.

📌 Practical Example: WebRTC in a MERN Stack App

Let’s say you’re building a video chat app using simple-peer or peerjs on top of WebRTC. Here’s what happens under the hood:

1. Signaling (done over TCP/WebSocket): Exchanging connection metadata.

2. STUN/TURN servers help clients punch through firewalls.

3. Media (video/audio) flows over UDP via SRTP (secure RTP).

Code snippet – Express signaling server:

// server.js (Express + Socket.IO for signaling)
const io = require('socket.io')(server, {
  cors: { origin: '*' }
});

io.on('connection', socket => {
  socket.on('signal', ({ to, data }) => {
    io.to(to).emit('signal', { from: socket.id, data });
  });
});

This code doesn’t handle actual media—just signaling. Media flows peer-to-peer over UDP once a connection is established.

🔍 Debugging WebRTC? Understand the Transport!
When WebRTC fails to connect:

• It might be blocked UDP ports or aggressive NATs.
• Knowing that WebRTC tries UDP first, then TCP (via TURN), helps you pick the right ICE config and fallback strategy.
• Inspect network traffic using chrome://webrtc-internals/ or Wireshark.

🧠 Key Differences for Devs to Know

Conclusion

As developers, we often live at the application layer—but performance, reliability, and user experience are deeply affected by transport protocols.

Whether you're making RESTful API calls or adding a real-time video chat feature, understanding the trade-offs between TCP and UDP helps you:

• Design better APIs
• Write more robust socket/WebRTC clients
• Debug low-level issues faster
• Communicate effectively with backend and DevOps teams

🔑 Key Takeaways

• TCP is great for data integrity; UDP is built for speed.
• WebRTC relies heavily on UDP to stream media in real-time.
• Understanding TCP/UDP can help you solve production bugs, optimize latency, and write more resilient real-time features.
• Tools like Wireshark or webrtc-internals are your best friends when things go wrong.

📘 Next Steps

• Try adding WebRTC to your MERN app using simple-peer or peerjs.
• Explore TURN servers like coturn and STUN services like Google’s stun.l.google.com:19302.
• Watch your network traffic using chrome://webrtc-internals/ or Wireshark to see TCP/UDP in action.