Wondering whether WebRTC or WebSockets is the right fit for your real-time app? You're not alone. Whether you're building a video conferencing tool, multiplayer game, or live chat, this decision affects everything from performance to user experience.

Let us break it down in simple terms so you can confidently choose what's best for your project.

1. What is WebRTC and When can it be Used?

WebRTC (Web Real-Time Communication) is a low-latency, open-source protocol that facilitates real-time, peer-to-peer communication directly between browsers without relying on plugins or intermediary services. It's designed for streaming audio, video, and data in real-time making it the perfect choice for applications that require ultra-low latency and high responsiveness.

You should use WebRTC when:

You’re building a video or audio calling application.
You want secure, encrypted peer-to-peer file transfers.
Your app needs screen sharing or real-time collaboration (like whiteboarding).
You require adaptive media quality based on network conditions.

WebRTC reduces server load by enabling direct communication between users, making it highly efficient for media-heavy use cases like telehealth, webinars, online tutoring, and remote team collaboration.

Key Features:
- Peer-to-peer connections (less server load)
- Adaptive video/audio quality
- Built-in encryption (DTLS + SRTP)
- NAT traversal (STUN/TURN)
- Best for video calls, telehealth, and real-time collaboration

WebRTC Diagram

Image source: TechTarget - WebRTC

2. When to Choose WebSockets Over WebRTC

WebSockets enable a continuous, two-way communication channel between client and server, perfect for real-time apps that require efficient resource handling. Unlike WebRTC, which focuses on media and peer-to-peer communication, WebSockets are best suited for structured data streams controlled from a central server.

Choose WebSockets when:

You’re building a live chat or messaging application.
You need to deliver frequent updates, such as stock tickers or live sports scores.
You’re developing collaborative tools (e.g., Google Docs-like syncing).
You're managing IoT devices that send and receive small packets of data continuously.

Because WebSockets run over TCP and handle large numbers of concurrent users well, they’re an ideal fit for real-time dashboards, multiplayer games with centralized state control, or real-time notification systems.

Key Features:
- Runs over TCP (reliable delivery)
- Supports text and binary data
- Handles thousands of simultaneous connections
- Works well through proxies and firewalls

WebSockets Diagram

Image source: Ramotion - WebSockets

Security

WebRTC enforces mandatory encryption using DTLS and SRTP. WebSockets support secure WSS connections, but it's optional.

WebRTC enhances privacy by enabling direct peer-to-peer communication, eliminating the need for routing sensitive media data through central servers.

WebRTC Use Case:

- Video/Audio Calls: Ideal for low-latency media.
- Secure File Sharing: Peer-to-peer transfers.
- Screen Sharing: Simple via getDisplayMedia().
- Real-Time Gaming: Fast and responsive.

WebSockets Use Case:

- Chat Apps: Persistent, efficient messaging.
- Live Updates: Stock tickers, notifications.
- Document Collaboration: Real-time syncing.
- IoT Devices: Lightweight persistent connections.

Hybrid Approach

Combine the best of both:
- Use WebSockets for signaling and control
- Use WebRTC for media and low-latency data

Maintain both connections for reliability and performance.

Real-World Example: Meet Hour’s Hybrid Architecture

At Meet Hour, we combine the power of WebRTC and WebSockets to deliver a seamless, real-time communication experience for video conferencing, webinars, and remote collaboration.

This hybrid architecture ensures that Meet Hour is scalable, secure, and responsive ideal for use cases like telehealth, e-learning, corporate meetings, and online events.

WebRTC is at the core of Meet Hour’s media engine, enabling ultra-low latency peer-to-peer audio and video streaming. This allows us to deliver HD video quality, screen sharing, and encrypted communication between participants, without the need for additional plugins or third-party software.
We use WebSockets for efficient signaling and real-time control, such as managing meeting states (join/leave events), chat messaging, and participant presence updates. Since WebSockets maintain a persistent connection with our servers, users experience minimal delay in interactions, even during large-scale meetings with hundreds of participants.

Final Thoughts

WebRTC is ideal if your app demands real-time media streaming and efficient peer-to-peer architecture.
Choose WebSockets if you need persistent messaging and centralized control.
These technologies aren’t rivals – they’re tools. Pick the one that suits your use case, or blend both for maximum flexibility.

Use Cases

Industries

Get Started

Resources

Webrtc Vs Websockets Choosing Your Real Time Solution

Tags :

Share :

0 Comments

Login to post your comments

Categories

Recent Posts

Tags