WebRTC is the set of browser APIs and related protocols that lets compatible devices exchange live audio, video, and application data. A typical call captures local media, negotiates session details through an app’s signaling channel, and uses ICE to find a working network path; RTP and RTCP then carry and manage the media. In group calls, an SFU may forward selected streams to participants.
What is WebRTC?
WebRTC means Web Real-Time Communication. It enables browser applications and other compatible devices to exchange real-time audio, video, and generic application data. It is not one video-chat product or a single protocol: the W3C defines browser APIs, while the IETF specifies underlying protocols. The W3C WebRTC Recommendation, published 13 March 2025, defines ECMAScript APIs for sending and receiving media and application data. W3C WebRTC: Real-Time Communication in Browsers
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The IETF describes the goal as enabling implementations to communicate using audio, video, and data “along the most direct possible path between the participants.” That is a design goal, not a guarantee that every call will take a direct route; ICE may select a relay when needed. IETF RFC 8825
Which browser APIs handle capture, connections, and data?
| API | What it does |
|---|---|
getUserMedia() |
Requests permission to access local camera and/or microphone inputs. Access depends on browser permission and device availability; the resulting tracks can be added to a connection. |
RTCPeerConnection |
Configures a connection with another peer and negotiates media and data capabilities. |
RTCDataChannel |
Exchanges application data messages between endpoints, separately from audio/video media tracks. |
These interfaces have different jobs: capture obtains local media, a peer connection handles communication setup, and a data channel carries application messages. The WebRTC API documentation describes these browser-facing interfaces. MDN: WebRTC API
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What is signaling?
Signaling is the application’s chosen way to coordinate call setup. Peers use it to exchange session descriptions, ICE candidates, and call-control messages. WebRTC does not require a particular signaling protocol or transport; an application might use a WebSocket service, but that is an architectural choice, not WebRTC’s media transport.
Signaling is needed to get setup information between endpoints, but it is distinct from the path that carries the call’s media. The app is responsible for providing whatever coordination service its design needs. MDN: WebRTC connectivity
What are offer and answer, SDP, and codecs?
Offer and answer
The initiating peer creates an SDP offer describing the session parameters it wants. The other peer returns an SDP answer describing the compatible session they have negotiated. The application passes these descriptions through its signaling channel; neither description is itself the audio or video.
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SDP
Session Description Protocol (SDP) is a textual format for session negotiation information, including media types, transport addresses, timing, and codec details. It is metadata used to arrange a session—not the media stream, the connection itself, or a signaling server. MDN: SDP glossary
Codec
A codec is an encoding and decoding format for media. Peers use SDP negotiation to identify mutually supported codecs and parameters. No single codec is universally best: compatibility and the use case determine which negotiated option is appropriate.
What is an ICE candidate?
An ICE candidate is a proposed way for peers to connect, with addressing and transport details. Interactive Connectivity Establishment (ICE) gathers possible paths and tests candidate pairs to find one that works. Candidates are exchanged through the application’s signaling channel as they are gathered.
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A candidate pair may enable direct connectivity, or ICE may use a TURN-relayed route. The steps are related but distinct: signaling carries candidate information between peers, while ICE evaluates possible paths. MDN: ICE glossary
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| Term | Role | What it does not do |
|---|---|---|
| STUN | Helps a peer discover the network-facing address associated with it, which can help ICE assess connectivity. | It does not relay the call’s media. |
| TURN | Relays traffic when peers cannot connect directly, for example under restrictive firewall or NAT conditions. | It is not simply another name for address discovery. |
NAT (Network Address Translation) lets devices on a local network share a public-facing address. That arrangement is one reason direct peer connectivity can be difficult. A TURN route adds an intermediary path and network overhead, but can make a call possible when a direct candidate pair fails. ICE considers both direct and relayed possibilities; applications should not assume every call will connect directly. MDN: Introduction to WebRTC protocols
How do RTP, RTCP, streams, and tracks fit together?
RTP and RTCP
RTP carries media packets. RTCP is its integral control and reporting protocol; they work together rather than serving as competing alternatives. Both are required parts of WebRTC media transport. IETF RFC 8834
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MediaStream and MediaStreamTrack
A MediaStream groups zero or more tracks. A MediaStreamTrack represents one media source, such as a camera or microphone, but it need not correspond to a single physical device: a track can also represent a conceptual source such as a mix or composition. A stream is therefore a grouping, not necessarily one device or one indivisible video feed.
How does WebRTC work in a multiparty call?
In a multiparty system, an SFU (Selective Forwarding Unit), also called an SFM (Selective Forwarding Middlebox), commonly forwards suitable streams to recipients. This avoids requiring every sender to produce a separate stream for every participant. The forwarding service still has work to do; the design shifts how media is distributed rather than removing the need to handle it.
Simulcast means sending multiple simultaneous versions of one video source at different resolutions and bitrates. An SFU can forward a layer suited to each recipient’s device capabilities and network conditions. This helps explain why participants in the same call may receive different versions of a source. MDN: Introduction to WebRTC protocols
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What is the usual order of a WebRTC connection?
- Capture media if needed: the application requests camera or microphone access with
getUserMedia(). A call that does not use local camera or microphone capture may not need this step. - Configure the peer connection: the application creates and configures an
RTCPeerConnectionfor the session. - Negotiate session parameters: one peer creates an SDP offer, the other returns an SDP answer, and the application exchanges both through its signaling channel.
- Gather and exchange ICE candidates: peers pass candidate information over signaling as ICE discovers possible paths.
- Check candidate pairs: ICE tests paths, which may be direct or use TURN if direct connectivity does not work.
- Send media or data: media travels using RTP with RTCP control and reporting; application messages can use a separate
RTCDataChannel.
In a multiparty call, an SFU may forward selected streams and, where simulcast is used, suitable video layers. These stages describe the ordinary flow; an application’s exact signaling and backend design is its own choice. MDN: Signaling and video calling
Is WebRTC the same as a video-chat service?
No. WebRTC is a browser API and protocol ecosystem that applications can use for real-time communication. A video-chat service builds an application around it and chooses its own signaling, interface, and backend architecture. WebRTC covers data messaging as well as audio and video; a particular product may use only some of those capabilities.
Further reading
- W3C WebRTC Recommendation
- IETF RFC 8825: Overview of browser-based real-time protocols
- IETF RFC 8834: Media transport and RTP in WebRTC
- MDN WebRTC API overview
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