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What Is WebRTC? A Simple Guide to Real-Time Streaming

WebRTC lets browsers and compatible apps exchange live audio, video, and data. Understand signaling, ICE, STUN and TURN, security, and equipment basics.

By VGSources Team 5 min read
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WebRTC is a set of web technologies that lets browsers and compatible apps exchange live audio, video, and data. It is not a video-call service by itself: an app handles permissions and signaling, while WebRTC establishes a secure connection between devices—directly when possible, or through a relay when necessary.

What WebRTC is—and what it is not

WebRTC stands for Web Real-Time Communication. The W3C Recommendation describes it as ECMAScript APIs that let media and application data be sent to and received from another browser or device using compatible real-time protocols. W3C WebRTC Recommendation, published 13 March 2025.

It is a collection of APIs and protocols, not a standalone calling app or a promise that every connection will be direct. Services use it to build voice and video calls, conferencing, screen sharing, file exchange, and other real-time data features. WebRTC.org

What WebRTC can carry

  • Audio and video: Apps can capture camera and microphone media through browser media APIs, then send media tracks over a connection.
  • Screen sharing: A compatible app can capture a screen or window and send it as media.
  • Application data: An app can use RTCDataChannel to exchange data that is not an audio or video track.

RTCPeerConnection manages the connection and its media or data channels. Which features are available and how they behave can vary by browser and platform, so check compatibility for the specific devices and APIs your application needs. MDN Web Docs: WebRTC API

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How a WebRTC connection is arranged

A useful way to picture setup is that the app first exchanges the information needed to arrange a connection; then ICE tries possible network routes. The signaling and connection checks are related parts of setup, but they are not the same job.

  1. Request media if needed. The app asks for camera, microphone, or screen access when its feature requires it. A data-only application may not need to capture local media.
  2. Create a peer connection. The app creates an RTCPeerConnection and adds media tracks or configures a data channel as appropriate.
  3. Create and exchange session descriptions. One endpoint creates an offer and sets its session description. The app sends that description to the other endpoint through its signaling system; the other endpoint returns an answer.
  4. Exchange ICE candidates. Each side shares candidate network routes through signaling. With trickle ICE, candidates can be sent as they are discovered rather than waiting for the full set.
  5. Let ICE test routes. The endpoints perform connectivity checks, and ICE selects a working candidate pair if one is available.
  6. Send media or data. Once a usable path is established, the configured tracks or data channel can carry communication.

That sequence is the typical shape rather than a requirement that every app implement its signaling identically. WebRTC does not specify the signaling transport: an application may use a regular web service or another out-of-band mechanism to pass session descriptions and candidates. WebRTC.org: Getting started with WebRTC · WebRTC.org: Getting started with peer connections

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Why signaling, STUN, TURN, and ICE exist

Signaling exchanges setup information

Before two endpoints can coordinate a session, they need to exchange session descriptions and ICE candidates. The application’s signaling system carries that information; it is separate from the WebRTC APIs and does not itself establish the media route.

ICE tests possible routes

ICE (Interactive Connectivity Establishment) gathers and tests candidate paths so endpoints can find one that works across their network conditions. People sometimes use “signaling” broadly for the exchange of descriptions and candidates, but ICE is the process that checks connectivity.

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STUN helps discover an address as seen from outside

A STUN service helps an endpoint learn how its address appears from the public side of its local network. It helps with route discovery; it does not carry the call’s media.

TURN relays when direct connectivity fails

If network restrictions prevent a direct connection, a TURN service can relay traffic between endpoints. This provides a fallback route but requires relay infrastructure and means the media traffic passes through that relay. Not every session needs TURN.

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Route What happens Trade-off
Direct path ICE selects a path between endpoints without relaying the media through a TURN service. Can avoid TURN relay infrastructure, but may not work on restrictive networks.
TURN relay A relay forwards traffic when a direct path cannot be established. Offers a fallback, but requires relay infrastructure and carries traffic through the relay.

For that reason, “peer-to-peer” describes the connection model, not a guarantee that packets always travel directly from one device to the other. MDN Web Docs: WebRTC connectivity

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Does WebRTC need a server?

WebRTC needs a way for the endpoints to exchange setup information, but the standard does not require one particular signaling server or transport. Applications commonly provide signaling through their own web service or another communication channel. STUN and TURN are separate network services used for route discovery and relaying; a TURN relay is needed only when a usable direct path cannot be established. The signaling service and the media route should not be conflated.

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Is WebRTC secure?

The W3C specification states that a user agent will always encrypt data, using per-session keying with DTLS-SRTP for media transport, and describes consent and congestion safeguards. Encryption protects the transport; it does not verify who the other participant is, certify an app’s privacy practices, or hide the fact that communication is taking place from network observers. W3C WebRTC Recommendation

  • Review the browser’s camera, microphone, and screen-sharing permission prompts, and grant only the access needed for the feature.
  • Use the app’s own identity and account safeguards to confirm whom you are communicating with; transport encryption alone does not do that.
  • Check the application’s privacy information to understand how it handles media and related data.

What equipment do you need?

WebRTC itself does not impose a headset requirement. A microphone is useful when you want to send audio, and a camera is needed for camera video; screen sharing uses screen-capture access instead. A headset with a microphone is an optional setup choice, not a WebRTC prerequisite. Actual device requirements depend on the app and the feature you use. WebRTC.org: Getting started with WebRTC

WebRTC for live streaming: know the distinction

WebRTC supports real-time communication between browsers and compatible applications, including live audio and video exchange. That does not make it a complete live-streaming service: an application still has to provide the capture, signaling, connection setup, and any audience or distribution features it needs. If your goal is to keep a pre-recorded video running as a 24/7 YouTube live stream, that is a different workflow from a browser-to-browser WebRTC call.

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