Here at Velt, lots of devs ask us about the differences between WebRTC vs WebSockets for real-time collaboration features. Both are technologies that enable real-time functionality for applications, but they operate in different spaces. I'll break down the core differences between the two and when to choose which one.

TLDR:

• WebSockets allow persistent, bidirectional communication between clients and servers using TCP

• WebRTC allows peer-to-peer media and data transmission directly between browsers using UDP

• WebSockets excel at messaging, notifications, and data sync while WebRTC handles video calls and media streaming

• Most modern apps use both technologies for different features

• Implementation complexity can be eliminated with the right SDK approach

What are WebSockets

WebSockets create a persistent, two-way communication channel between your browser and a server. Think of it like having a phone line that stays open instead of hanging up after every conversation.

Unlike traditional HTTP requests where your browser asks for something and the server responds once, WebSockets maintain an always-on connection. This means the server can push updates to your browser instantly without waiting for a request.

Here's how it works: Your browser sends a special HTTP request asking to "upgrade" to a WebSocket connection. If the server agrees, they perform a handshake and create a persistent connection. From that point on, both sides can send messages back and forth freely.

The magic happens because WebSockets use TCP (Transmission Control Protocol), which guarantees that messages arrive in order and without errors. This reliability makes WebSockets perfect for applications where data integrity matters.

WebSockets reduce server overhead by up to 500x compared to traditional HTTP polling, making them important for scalable real-time applications.

We use WebSocket technology extensively in our commenting system to make sure that when someone leaves a comment, everyone sees it instantly without refreshing the page.

What is WebRTC

WebRTC (Web Real-Time Communication) takes a different approach. Instead of routing everything through a server, it lets browsers talk directly to each other in a peer-to-peer fashion.

WebRTC primarily uses UDP (User Datagram Protocol), which puts speed over reliability. Unlike TCP, UDP doesn't guarantee that every packet arrives or arrives in order. But for live video calls, it's better to skip a few frames than to pause the entire stream waiting for missing data.

The peer-to-peer nature means that once two browsers create a WebRTC connection, they can exchange audio, video, and data streams directly. No server sits in the middle processing every frame of video or audio sample.

Setting up these direct connections requires some coordination though. Browsers need to exchange connection information through a process called signaling, and they often need help from STUN and TURN servers to get through firewalls and network translation.

This is why our live huddles use WebRTC. When you're having a quick voice chat with your team, that audio streams directly between participants for the lowest possible latency.

Core Protocol Differences

The fundamental difference comes down to architecture and reliability priorities. Let's break this down:

WebSockets excel when you need guaranteed message delivery and don't mind the slight latency overhead. Every message gets acknowledged, and if something goes wrong, the protocol handles retransmission automatically.

WebRTC shines when speed trumps perfection. A video call can handle a few dropped frames, but it can't handle a 2-second delay while waiting for a lost packet to be retransmitted.

The server architecture differs dramatically too. With WebSockets, your server needs to maintain connections and route messages between clients. With WebRTC, the server mainly helps set up connections, then gets out of the way.

At Velt, we've built our platform to use both protocols strategically. We're not religious about one approach because different features have different requirements.

Data Types and Communication Models

WebSockets handle text-based data beautifully. JSON messages, chat content, notifications, and state updates all flow smoothly over WebSocket connections. The reliable delivery means you never have to worry about a chat message getting lost or arriving out of order.

WebRTC was designed for media streams. Audio samples, video frames, and real-time data channels all benefit from WebRTC's low-latency approach. When someone's speaking in a video call, you want to hear their voice immediately, even if a few milliseconds of audio get dropped.

But here's where it gets interesting: WebRTC also supports data channels for non-media content. These channels can be configured for either reliable delivery (like WebSockets) or unreliable delivery (for speed). You get to choose the tradeoff.

The communication models reflect these strengths. WebSockets typically follow request-response or broadcast patterns, where a server coordinates message distribution. WebRTC allows direct peer communication, perfect for collaborative scenarios where users interact directly.

Our commenting features use WebSockets because comments need to arrive reliably and in order. But our video features use WebRTC because real-time media can't tolerate the delays that come with guaranteed delivery.

Common WebSockets Use Cases

WebSockets power most of the real-time text-based experiences you use daily:

Live chat and messaging: Every modern chat application relies on WebSockets for instant message delivery. When you send a message in Slack or Discord, WebSockets make it appear immediately in everyone's conversation.

Real-time notifications: Push notifications, activity feeds, and status updates all benefit from WebSocket's persistent connections. No more refreshing pages to see if something new happened.

Collaborative document editing: Google Docs-style collaboration needs reliable, ordered updates. When multiple people edit the same document, WebSockets make sure everyone sees changes in the correct sequence.

Live data feeds: Stock prices, sports scores, and social media feeds use WebSockets to push updates as they happen. The persistent connection means updates arrive instantly without polling.

Presence and status: Showing who's online, typing indicators, and cursor positions all rely on WebSocket connections to broadcast user activity in real-time.

We've built our notification system and document collaboration features around WebSockets because these use cases demand reliable, ordered message delivery.

Common WebRTC Use Cases

WebRTC dominates scenarios where direct, low-latency communication matters:

Video conferencing: Zoom, Google Meet, and similar platforms use WebRTC for the actual audio and video streams. The peer-to-peer connections minimize latency and reduce server costs.

Screen sharing: When you share your screen in a meeting, WebRTC streams those pixels directly to other participants. The real-time nature means viewers see your actions as they happen.

File transfers: Large file sharing applications use WebRTC data channels to transfer files directly between browsers, bypassing server storage limitations.

Live streaming: Interactive streaming platforms use WebRTC to minimize the delay between streamers and viewers, allowing real-time interaction.

Gaming applications: Multiplayer browser games use WebRTC for voice chat and real-time game state synchronization where speed matters more than perfect reliability.

Our live video huddles use WebRTC to send video and audio media streams over a low-latency connection from one client to another.

Performance and Scalability Considerations

WebSockets face traditional server scaling challenges. Each connection consumes server resources, and you need to plan for connection limits. A single server might be able to handle 10,000 concurrent WebSocket connections, but beyond that, you need load balancing and connection distribution strategies.

The advantage is predictable scaling patterns. You can calculate server requirements based on concurrent users and message frequency. Horizontal scaling works well with proper session management and message routing.

WebRTC reduces server load dramatically once connections are set up. The peer-to-peer architecture means your servers don't process media streams. However, you still need signaling servers and STUN/TURN infrastructure for connection setup and NAT traversal.

WebRTC can reduce bandwidth costs by up to 90% for media-heavy applications by eliminating server-side media processing.

The complexity comes from managing peer connections. Network conditions vary between users, and you need fallback strategies when direct connections fail. Some connections require TURN servers, which do consume server resources.

At Velt, we handle these scaling complexities so you don't have to. Our infrastructure automatically manages connection limits, load balancing, and failover scenarios across both WebSocket and WebRTC connections.

How Velt Makes Real-Time Communication Simple

Building real-time features from scratch means months of infrastructure work. You need WebSocket servers, connection management, message queuing, WebRTC signaling servers, STUN/TURN infrastructure, and solid error handling.

We've abstracted all that complexity into simple SDK components. Want real-time comments? Add our comment component and we handle the WebSocket connections, message ordering, and offline synchronization automatically.

Need video video calls for collaboration? Our video huddles manage WebRTC peer connections, media capture, and stream processing without exposing the underlying complexity.

The magic is in choosing the right technology for each feature. Comments and notifications use WebSockets for reliability. Video calls and screen sharing use WebRTC for performance. You get both without implementing either.

Our SDK provides 25+ collaboration features that would typically require separate implementations of both protocols. Instead, you get enterprise-grade real-time functionality with full customization options.

Implementation Complexity and Development Time

Implementing WebSockets from scratch requires building server infrastructure, managing connection lifecycles, handling reconnection logic, and implementing message queuing for offline users. You're looking at weeks of development for basic functionality.

WebRTC is even more complex. You need signaling servers for connection establishment, STUN servers for NAT traversal, TURN servers for firewall bypass, and solid error handling for network failures. Most teams spend 2-3 months just getting basic video calls working reliably.

Then there's the ongoing maintenance. Connection management, scaling infrastructure, handling edge cases, and debugging network issues across different browsers and network conditions.

We've seen teams spend 6+ months building what our SDK provides out of the box. Our implementation comparison shows the dramatic difference in development time and ongoing maintenance requirements.

The complexity goes beyond technical challenges. You need expertise in networking protocols, media processing, and real-time systems architecture. Most product teams want to focus on their core features, not become WebRTC experts.

FAQ

What's the main difference between WebSockets and WebRTC?

WebSockets create persistent connections between browsers and servers for reliable messaging, while WebRTC allows direct peer-to-peer communication for media and data streams with lower latency.

Can I use both WebSockets and WebRTC in the same application?

Absolutely. Most modern real-time applications use both technologies for different features. WebSockets handle messaging and notifications while WebRTC manages video calls and media sharing.

Which protocol is better for real-time chat?

WebSockets are better for text-based chat because they guarantee message delivery and ordering. WebRTC is overkill for simple messaging and adds unnecessary complexity.

Do I need special servers for WebRTC?

Yes, WebRTC requires signaling servers to create connections and often needs STUN/TURN servers to handle firewalls and NAT traversal. This infrastructure complexity is why many teams choose SDK solutions.

How do these technologies affect mobile performance?

WebSockets are lightweight and mobile-friendly. WebRTC can be more resource-intensive on mobile devices, especially for video processing, but modern smartphones handle it well with proper optimization.

Final thoughts on choosing between WebSockets and WebRTC

You don't need to choose between WebSockets and WebRTC, you need both working together smoothly. WebSockets handle reliable messaging while WebRTC powers low-latency media streaming. Building this real time communication infrastructure from scratch takes months of complex development. Velt combines both protocols into simple SDK components that integrate with any JavaScript framework so you don't have to worry about building it yourself.