WebRTC is a set of browser APIs and companion protocols for real-time audio, video, and application data—not a video-chat product or a single protocol. A typical call captures local media, negotiates a session through app-defined signaling, tests network paths with ICE, and carries media over RTP. The terms below follow that sequence.
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. The W3C defines the browser-facing APIs; the IETF specifies related protocols. They are complementary standards work, not one specification. The W3C WebRTC Recommendation was published on 13 March 2025. W3C WebRTC: Real-Time Communication in Browsers
As the IETF put it in RFC 8825, published in January 2020, the goal is to let implementations communicate using audio, video, and data sent along “the most direct possible path between the participants.” This describes the goal, not a guarantee that every call will use a direct route: a relay may be needed. IETF RFC 8825
Which browser APIs handle capture, connections, and data?
| Term | Role |
|---|---|
getUserMedia() |
Requests access to local camera and/or microphone inputs, subject to browser permission and device availability. The resulting tracks can be added to a real-time 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 tracks. |
These are distinct responsibilities: capturing media does not itself establish a call, and a data channel is not an audio or video track. MDN WebRTC API
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How do signaling, offer/answer, and SDP fit together?
Signaling
Signaling is the application’s chosen coordination channel for setup information, including session descriptions, ICE candidates, and call-control messages. WebRTC does not require a particular signaling protocol or transport. An application might use a WebSocket, but that service is part of the application architecture, not the WebRTC media transport.
Offer and answer
The initiating peer creates an SDP offer describing desired session parameters. The other peer returns an SDP answer describing the compatible negotiated session. The application passes these descriptions through its signaling channel.
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SDP and codecs
SDP, or Session Description Protocol, is a textual session-description format containing negotiation details such as media types, transport addresses, timing, and codec information. It is metadata—not the audio/video stream, a connection, or a signaling server. Peers use SDP negotiation to identify mutually supported codecs and parameters. A codec is an encoding/decoding format for media; there is no universally best codec independent of compatibility and use case. MDN: Introduction to WebRTC protocols
What is an ICE candidate?
An ICE candidate is a proposed way for peers to connect, with addressing and transport details. ICE (Interactive Connectivity Establishment) gathers possible paths and checks candidate pairs to find a working route. Candidates are exchanged as they are gathered, typically through the same application-defined signaling channel used for offer and answer.
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NAT, or Network Address Translation, lets devices on a local network share a public-facing address. This and other network restrictions can make a direct path difficult, which is why ICE considers multiple possibilities. MDN ICE glossary
What are STUN and TURN servers?
| Term | What it does | What to expect |
|---|---|---|
| STUN | Can help ICE discover the network-facing address associated with a peer. | It assists with address discovery; 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 adds an intermediary route and network overhead. |
ICE may select a direct candidate pair if checks succeed, or a TURN-relayed path when direct connectivity fails. Direct connectivity is not guaranteed; the useful trade-off is reachability through a relay versus the additional relay hop and overhead. MDN: Introduction to WebRTC protocols
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How are media tracks carried?
MediaStream and MediaStreamTrack
A MediaStream groups zero or more tracks. A MediaStreamTrack represents one media source, such as a camera or microphone; a track can also represent a conceptual source, such as a mix or composition. A stream therefore is not necessarily one physical device.
RTP and RTCP
RTP carries media packets. RTCP is its integral control and reporting protocol; they are complementary, not competing transports. Both are required in WebRTC media transport. IETF RFC 8834: Media Transport and Use of RTP in WebRTC
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How does WebRTC handle multiparty video?
Mesh and SFU forwarding
In a peer mesh, each sender must send streams to multiple participants, increasing that sender’s upload work as the group grows. A common multiparty alternative is an SFU (Selective Forwarding Unit), also called an SFM (Selective Forwarding Middlebox): it receives streams and forwards selected ones to participants. The intermediary needs forwarding infrastructure, while senders need not produce a separate stream for every recipient. Exact cost, latency, and privacy implications depend on the deployment.
Simulcast
Simulcast sends multiple simultaneous encodings of one video source at different resolutions and bitrates. An SFU can forward a suitable layer to each recipient based on that recipient’s network and device capabilities. MDN: Introduction to WebRTC protocols
What is the usual WebRTC call sequence?
- Capture, if needed: request local camera or microphone access with
getUserMedia(), then add the resulting tracks to the connection. - Create the connection: configure an
RTCPeerConnectionwith the application’s connection setup. - Negotiate: create an SDP offer, deliver it through the application’s signaling channel, and receive the other peer’s SDP answer.
- Exchange candidates: send ICE candidates through signaling as they are gathered.
- Check paths: ICE tests candidate pairs and selects a working direct or relayed route.
- Communicate: media travels using RTP with RTCP control/reporting; application messages may use a separate
RTCDataChannel.
Is WebRTC the same as a video-call app?
No. WebRTC provides browser APIs and protocol building blocks. An application still chooses its signaling service and may need backend services, such as TURN relaying or an SFU for multiparty forwarding. Different products can use WebRTC while providing different interfaces and infrastructure; the standard itself is not a particular video-chat service.
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