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WebRTC is the collection of browser APIs and related protocols that lets compatible devices exchange real-time audio, video, and application data. A typical call uses local capture, an app-provided signaling channel to negotiate the session, and ICE to find a working network path; RTP carries media once that path is selected.
What is WebRTC?
WebRTC means Web Real-Time Communication. It enables browser applications and other compatible devices to communicate with real-time media and data. Despite the name, it is not limited to video: applications can exchange audio, video, and generic application data.
WebRTC is not a video-chat product, a single server, or one protocol. The W3C WebRTC Recommendation, published 13 March 2025, defines browser-facing APIs for sending and receiving media and application data. The IETF specifies related protocols. The two standards efforts work together, but they are not the same specification.
In IETF RFC 8825, published in January 2020, the goal is described as enabling implementations to communicate using audio, video, and data “sent along the most direct possible path between the participants.” ICE may find a direct path, but a direct connection is not guaranteed; a TURN relay can be used when needed.
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How a WebRTC call is put together
- Capture local media, if the application needs it. The browser asks for permission to use a camera or microphone, then provides media tracks.
- Create and configure a peer connection. The application uses
RTCPeerConnectionto set up negotiation and connection options. - Exchange session descriptions. One endpoint creates an SDP offer; the other responds with an SDP answer. The application passes these through its own signaling channel.
- Exchange and test ICE candidates. Candidates describe possible network paths. The peers check candidate pairs to find a working route, direct or relayed.
- Send media over the selected path. RTP carries media packets, while RTCP provides related control and reporting. An application can also use a data channel for non-media messages.
This is a useful mental model rather than a requirement that every application follow one identical implementation. For example, an application may not need camera capture, and WebRTC does not prescribe a particular signaling transport.
Browser APIs and media objects
getUserMedia()
A browser media-capture API that requests access to local camera and/or microphone inputs. The request is subject to user permission, browser support, and device availability. Tracks obtained from capture can be added to a real-time connection.
RTCPeerConnection
The browser interface used to configure and manage a connection with another peer and negotiate media or data capabilities. It participates in session negotiation and connectivity setup; it is not itself the signaling service.
RTCDataChannel
An API for exchanging application data messages between endpoints. It is separate from audio and video media tracks, so an application can use it for data rather than treating it as another media stream.
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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 directly to one physical device: tracks can also represent conceptual sources such as a mix or composition.
Signaling, offer/answer, and SDP
Signaling
Signaling is the application’s chosen coordination channel for exchanging setup information. It commonly carries session descriptions, ICE candidates, and call-control messages. WebRTC does not mandate a particular signaling protocol or transport; an application could, for example, use a WebSocket service, but that is an application architecture choice rather than WebRTC’s 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 delivers both descriptions through its signaling channel.
SDP (Session Description Protocol)
SDP is a textual format for session-description information, including items such as media types, transport addresses, timing, and codec information. It is negotiation metadata—not the audio or video, the connection itself, or the signaling server.
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Codec
A codec is an encoder/decoder format used for media. During SDP negotiation, peers identify codecs and parameters they support in common. There is no universally best codec independent of compatibility and use case.
ICE, candidates, STUN, and TURN
ICE (Interactive Connectivity Establishment)
ICE is the connectivity framework peers use to gather and test possible network paths. It considers candidate paths that may be direct or relayed, then checks combinations to find a working route.
What is an ICE candidate?
An ICE candidate is a proposed connection method with addressing and transport details. Candidates are exchanged as they are gathered, then checked in candidate pairs to determine whether a path works. A candidate is a possibility, not proof that a connection has been established.
What are STUN and TURN servers?
- 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.
- TURN is a relay service that forwards traffic when peers cannot connect directly, including in some restrictive firewall or NAT conditions. A relayed route adds an intermediary hop and network overhead.
NAT (Network Address Translation)
NAT allows devices on a local network to share a public-facing address. Because a device’s local address may not be directly reachable from the other participant, NAT is one reason ICE techniques can be needed to find a working path.
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Direct path versus TURN relay
| Path | When it can work | What to keep in mind |
|---|---|---|
| Direct | An ICE candidate pair succeeds without a relay. | It avoids the extra relay hop, but restrictive network conditions can prevent a direct path. |
| TURN-relayed | Direct connectivity fails and a relay path is available. | TURN forwards traffic through an intermediary, adding a network hop and overhead. |
ICE handles discovery and checking; an application should not promise that every call will connect directly.
Independent reader supportYour contribution helps us test, update, and keep practical guides available for everyone.Media transport: RTP, RTCP, and tracks
RTP and RTCP
In WebRTC, RTP carries media packets. RTCP is its integral control and reporting protocol. They are complementary parts of media transport, not competing alternatives. IETF RFC 8834, published in January 2021, specifies media transport and the use of RTP in WebRTC.
How media tracks relate to transport
A media stream groups tracks for application use; the tracks represent individual sources. The media itself is transported over the selected connection path using RTP, with RTCP providing control and reporting. SDP describes negotiated session details but does not contain the media.
Multiparty calls: SFU and simulcast
SFU/SFM (Selective Forwarding Unit or Middlebox)
An SFU is an intermediary commonly used in multiparty video systems. Rather than requiring each sender to produce a separate stream for every participant, it can select and forward suitable streams to recipients. Its role is forwarding; deployment-specific cost, latency, and privacy depend on the system and cannot be inferred from the term alone.
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Simulcast
Simulcast sends multiple simultaneous encodings of one video source at different resolutions and bitrates. An SFU can forward a version suited to each recipient’s network and device capabilities. This can help a multiparty system serve participants with differing conditions without implying that every WebRTC call uses simulcast.
Mesh and SFU-style forwarding
| Approach | Sender’s upload work | Forwarding and adaptation |
|---|---|---|
| Peer mesh | A sender may need to send separate streams to multiple participants, increasing its upload work as peers are added. | Participants connect to one another; there is no SFU selecting and forwarding streams. |
| SFU-style | A sender can send a stream or simulcast layers to the intermediary rather than a separate stream for every recipient. | The intermediary forwards selected streams or layers to recipients. Simulcast can provide different resolution and bitrate options. |
The exact trade-offs depend on the application’s deployment and configuration; the architecture labels alone do not establish a universal cost, latency, or privacy outcome.
Where StreamNeo fits—and where it does not
StreamNeo is a separate YouTube service for keeping an uploaded video or playlist live 24/7 from the cloud. It is not a WebRTC glossary term, and it does not capture live camera video. If your goal is a continuous YouTube stream rather than understanding browser-to-browser real-time communication, see StreamNeo: upload a recording or build a playlist, add your YouTube stream key, and go live. Your computer does not have to stay on. StreamNeo supports uploaded quality up to 4K 60fps at one flat price per slot, can automatically recover if YouTube drops the stream, and offers a free first day with no card. Start at StreamNeo’s free trial.
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Product prices and availability are accurate as of the date/time indicated and are subject to change. Any price and availability information displayed on Amazon at the time of purchase will apply.
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