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Scan for outdated or missing drivers - takes under a minuteDriver Scan →Repair Windows errors before they cause bigger problemsFix Now →Fix the driver behind crashes, sound loss and screen glitchesFind Drivers →WebRTC (Web Real-Time Communication) 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 coordinates a connection and sends the media or data once a viable network route is found.
What WebRTC does—and what it does not do
WebRTC provides browser APIs and communication protocols for real-time exchanges between compatible browsers or devices. It can support voice and video calls, screen sharing, file exchange, and other application data. The W3C Recommendation describes APIs for sending and receiving media and generic application data between browsers or devices that implement the relevant protocols (W3C WebRTC Recommendation, 13 March 2025).
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WebRTC is not a complete calling product. An application still needs to provide the interface, decide whom users can contact, request permissions, and arrange the initial connection setup. Nor does WebRTC mean that every call travels directly from one device to another: network conditions may require a relay.
What the main WebRTC components do
- Media capture APIs: Browser APIs can request access to a camera and microphone. Screen capture is another supported way to supply media. The application determines what to capture and when to request permission (WebRTC.org’s getting-started guide).
RTCPeerConnection: This API manages the connection between endpoints, including session setup and the exchange of media tracks.RTCDataChannel: This can carry application data, not just audio or video. It can be used for exchanges such as files or other data supported by the app.- Signaling: The app’s separate mechanism for exchanging setup information between endpoints. WebRTC does not prescribe a particular signaling service or transport.
- ICE, STUN, and TURN: ICE tests possible network routes. STUN helps an endpoint discover the address it appears to have from outside its local network; TURN relays traffic when a direct route cannot be established.
The MDN WebRTC API overview covers the browser interfaces and their role in applications.
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How a WebRTC connection is established
- Capture media if the app needs it. The app requests camera, microphone, or screen access as appropriate. A data-only exchange may not need media capture.
- Create a peer connection. The app creates an
RTCPeerConnectionand configures any needed connection services. - Create and set a session description. One endpoint prepares an offer describing the session; the other endpoint returns an answer. The descriptions are applied to their respective peer connections.
- Exchange setup information through signaling. The app sends the offer and answer through its own signaling channel, which may use a web service or another out-of-band method.
- Exchange ICE candidates. Each endpoint shares candidate routes as they are discovered. This is commonly done over the signaling channel. With trickle ICE, candidates can be sent as discovery proceeds rather than waiting for the full set.
- Let ICE test routes. ICE checks candidate pairs and selects a working path if one is available. Once connected, the endpoints can send media tracks or use a data channel.
People often group offer-and-answer exchange and candidate exchange under the term “signaling.” Signaling carries the information needed to arrange a connection; ICE performs the connectivity checks. The distinction is explained in WebRTC.org’s peer-connection guide and MDN’s WebRTC connectivity guide.
Why WebRTC uses STUN and TURN
Devices are often behind routers, firewalls, or other network arrangements that make it difficult for one endpoint to reach another directly. ICE tests the available routes rather than assuming that a direct connection will work.
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- A direct route: If ICE finds a viable path between endpoints, media can travel without being relayed by a TURN server.
- STUN assistance: STUN helps a device learn how its address appears from outside its local network. It helps with route discovery; it does not carry the call’s media.
- TURN relay: If network restrictions prevent a direct connection, TURN can relay traffic between endpoints. That fallback requires relay infrastructure, and the media then passes through it.
A simple analogy: signaling exchanges the details for arranging the call; ICE tries possible routes; STUN helps identify an endpoint’s outward-facing address; TURN provides a relay route when needed. Not every WebRTC connection uses TURN, and “peer-to-peer” does not guarantee that all traffic is physically direct.
Is WebRTC secure?
The W3C WebRTC Recommendation states that a user agent always encrypts data, using per-session keying with DTLS-SRTP for media, and describes consent and congestion safeguards (W3C WebRTC Recommendation). Encryption protects data in transit, but it does not by itself prove who the other participant is or explain how a particular application handles media.
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Pay attention to the application’s identity and privacy practices, and grant camera or microphone access only when you intend to use those devices. Permission to use a microphone is not verification of the person on the other end. Network observers may also be able to tell that communication is taking place, even when its data is encrypted.
What equipment and browser support do you need?
WebRTC itself does not require a special headset. A microphone is useful when an app sends audio; built-in laptop or phone equipment may be sufficient for a given setup. A headset with a microphone is an optional accessory, not a WebRTC requirement.
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WebRTC support is broad across browsers, but implementations and platform behavior are not identical. Check that the browser and device you plan to use support the specific APIs your application needs. The MDN WebRTC API documentation is a useful starting point for browser-facing API information.
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