FAQ: What Is WebRTC? A Complete Guide to Real-Time Communication on the Web

Question 1

What does WebRTC stand for?

Accepted Answer

WebRTC stands for Web Real-Time Communication. It is a collection of open web standards and APIs that allow browsers and applications to communicate with each other in real time using audio, video, and data—without requiring plugins or additional software.

Question 2

Is WebRTC free to use?

Accepted Answer

Yes. WebRTC is completely open-source and free. The core WebRTC project is maintained by a large community and supported by major companies such as Google, Mozilla, Apple, and Microsoft. However, running supporting infrastructure like TURN servers may introduce operational costs.

Question 3

Does WebRTC work in all browsers?

Accepted Answer

WebRTC is supported by all modern major browsers, including:
Google Chrome
Mozilla Firefox
Microsoft Edge
Apple Safari
Mobile browsers on Android and iOS also support WebRTC, although there may be minor implementation differences.

Question 4

Is WebRTC secure?

Accepted Answer

Yes. WebRTC is secure by design. All WebRTC communications use mandatory encryption:
DTLS for data channels
SRTP for audio and video streams
Browsers will refuse to establish unencrypted WebRTC connections, making it suitable for enterprise and privacy-sensitive applications.

Question 5

Do I need a server to use WebRTC?

Accepted Answer

Yes and no.
Peer-to-peer media transfer happens directly between users.

However, you still need servers for:

Signaling (WebSocket, HTTP, etc.)
STUN/TURN services
Peer-to-peer media transfer happens directly between users.
However, you still need servers for:
Signaling (WebSocket, HTTP, etc.)
STUN/TURN services
In production environments, a TURN server is essential to guarantee reliable connectivity.

Question 6

What is the difference between WebRTC and WebSockets?

Accepted Answer

WebRTC and WebSockets serve different purposes:
WebRTC is optimized for low-latency, real-time audio, video, and peer-to-peer data.
WebSockets are best for structured, client-server messaging.
They are often used together, with WebSockets handling signaling and WebRTC handling media.

Question 7

Can WebRTC be used for live streaming?

Accepted Answer

Yes. WebRTC is widely used for ultra-low-latency live streaming, especially where real-time interaction is required. However, for large-scale one-to-many broadcasting, WebRTC is often combined with media servers or SFU/MCU architectures.

Question 8

Is WebRTC suitable for large-scale applications?

Accepted Answer

Yes, but not in a pure peer-to-peer form. Large-scale WebRTC applications typically rely on:
SFU (Selective Forwarding Units)
Media servers like Mediasoup, Janus, or LiveKit
These architectures enable WebRTC to scale to thousands or millions of users.

Question 9

What are common problems developers face with WebRTC?

Accepted Answer

Common challenges include:
NAT and firewall traversal
Debugging connection failures
Browser compatibility differences
TURN server configuration and cost
Despite these challenges, mature tooling and frameworks significantly reduce development complexity.

Question 10

Is WebRTC the future of real-time communication on the web?

Accepted Answer

WebRTC is already a cornerstone of real-time web communication. With ongoing improvements in codecs, scalability, and browser APIs, WebRTC is expected to remain a foundational technology for modern web-based communication platforms.

Alyvro

FAQ: What Is WebRTC? A Complete Guide to Real-Time Communication on the Web

Alyvro

Home

Products

Packages

Social

Subscribe to our newsletter

FAQ: What Is WebRTC? A Complete Guide to Real-Time Communication on the Web