WebRTC is a set of browser APIs and related protocols for real-time audio, video, and application data. A typical call captures local media, negotiates a session through app-defined signaling, checks possible network paths with ICE, and then carries media over the selected path. The terms below follow that sequence and distinguish the browser interfaces from the protocols and services around them.
What is WebRTC?
WebRTC means Web Real-Time Communication. It enables browser applications and 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 the browser APIs, while the IETF specifies underlying protocols. An application still needs to provide signaling and any backend services its design requires.
The IETF’s RFC 8825, published in January 2020, describes the goal as communication using audio, video, and data sent along “the most direct possible path between the participants.” That is a design goal, not a guarantee that every call will use a direct connection; ICE may select a relay when necessary.
Which browser APIs handle capture, connections, and data?
getUserMedia()
This media-capture API requests access to local camera and/or microphone inputs. Capture depends on user permission, browser support, and available devices. The resulting media tracks can be added to a real-time connection; camera capture is not required for every WebRTC application.
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RTCPeerConnection
This browser interface configures and manages a connection with another peer. It is used to negotiate media and data capabilities, exchange the information needed to establish connectivity, and handle the peer connection. It is not itself the signaling service or the video stream.
RTCDataChannel
This API carries application data messages between endpoints. It is separate from audio and video media tracks, so an application can use a data channel for information such as in-call messages or game state without treating that information as a media track.
How do signaling, offer/answer, and SDP fit together?
Signaling
Signaling is the application’s chosen coordination channel for exchanging session descriptions, ICE candidates, and call-control messages. WebRTC does not mandate a signaling protocol or transport. An application might use a WebSocket service, but that is an architectural choice; signaling is not the WebRTC media transport.
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Offer and answer
The initiating peer creates an SDP offer describing proposed session parameters. The other peer responds with an SDP answer describing the compatible negotiated session. The application delivers the offer and answer through its signaling channel.
SDP (Session Description Protocol)
SDP is a textual session-description format. It can contain information such as media types, transport addresses, timing, and codec details. It is negotiation metadata—not the audio or video, not the peer connection, and not a signaling server.
Codec
A codec is an encoding and decoding format for media. During negotiation, peers use SDP to identify mutually supported codecs and parameters. There is no universally best codec independent of compatibility and application needs.
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What is an ICE candidate?
ICE and candidate gathering
ICE means Interactive Connectivity Establishment. It is the framework peers use to gather and test possible network paths, including direct paths and relayed options. An ICE candidate is a proposed connection method with addressing and transport details. Candidates are exchanged as they are gathered, then candidate pairs are checked to find a working path.
NAT
Network Address Translation (NAT) lets devices on a local network share a public-facing address. Because a device’s local address may not be directly reachable from another network, NAT is one reason peers need ICE techniques to discover and test viable paths.
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STUN
STUN is a protocol service ICE can use to help discover the network-facing address associated with a peer. STUN assists with address discovery; it does not relay the call’s media.
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TURN
TURN is a relay service that forwards traffic when peers cannot connect directly, for example under restrictive firewall or NAT conditions. A TURN path adds an intermediary and network overhead, but can make a call possible when a direct candidate pair fails.
| Path outcome | When it can work | Trade-off |
|---|---|---|
| Direct path | ICE finds a viable candidate pair between peers. | A direct pair may not be reachable under the networks’ firewall or NAT conditions. |
| TURN-relayed path | A relay is available and ICE selects it when direct connectivity does not succeed. | Traffic passes through an intermediary, adding a network hop and overhead. |
ICE discovers and checks the options; an application should not assume that every call will be direct.
Independent reader supportYour contribution helps us test, update, and keep practical guides available for everyone.How do WebRTC media and tracks work?
RTP and RTCP
RTP carries media packets in WebRTC, while RTCP is its integral control protocol for control and reporting. They work together; they are not competing choices for transporting a call.
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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, and can also represent a conceptual source such as a mix or composition. A stream is therefore not necessarily one physical device.
How does a typical WebRTC connection come together?
- Capture, if the application needs local media. Request camera or microphone access with
getUserMedia(), subject to permission and device availability. - Create the peer connection. Configure an
RTCPeerConnectionand add the media tracks the application intends to send. The application may also establish anRTCDataChannelfor application messages. - Negotiate session capabilities. One peer creates an SDP offer; the other returns an SDP answer. The application passes both descriptions over its signaling channel.
- Exchange and test ICE candidates. Peers send gathered candidates through signaling and check candidate pairs. STUN can assist address discovery, while TURN can provide a relay path if direct connectivity fails.
- Send media over the selected path. Once a working path is established, RTP carries media and RTCP provides control and reporting.
How do multiparty calls use an SFU and simulcast?
SFU/SFM
A Selective Forwarding Unit (SFU), also called a Selective Forwarding Middlebox (SFM), is an intermediary commonly used for multiparty video. Rather than requiring each sender to produce a separate stream for every participant, it can select and forward streams to recipients. This supports recipients with different network conditions and device capabilities; specific cost, latency, or privacy consequences 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 version to each recipient based on that recipient’s network and device capabilities.
How is WebRTC different from a video-chat app?
WebRTC is the browser API and protocol foundation an application can use, not a brand or a complete hosted calling service. A specific app decides how users find one another, how signaling works, what backend services it runs, and whether calls are direct or use intermediaries such as TURN or an SFU.
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StreamNeo is a separate kind of service: it keeps uploaded videos live on YouTube from the cloud, rather than establishing browser-to-browser calls. It is not a WebRTC video-chat tool. Learn more at StreamNeo.
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