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What is WebRTC live streaming?
WebRTC is not one hosted streaming service or a single protocol. It is a coordinated set of browser APIs and real-time protocols that applications use to send and receive audio, video, and data. The World Wide Web Consortium (W3C) describes its role this way: “The WebRTC 1.0 specification exposes an API to control protocols (defined within the IETF) necessary to establish real-time audio, video and data exchange.” W3C WebRTC Recommendation, dated 13 March 2025.
The W3C defines the browser-facing API; IETF specifications define transport and related protocols. “Live streaming” can refer to different delivery patterns: a one-to-one call, a small direct exchange, or a multi-party session distributed through an intermediary. The architecture determines how media reaches recipients.
How does real-time video work?
- Obtain or select media. The application may capture a camera and microphone or choose another available media source. WebRTC works with media tracks represented by
MediaStreamTrackobjects. - Add tracks to a peer connection. The application attaches tracks to an
RTCPeerConnection, the browser API object used to manage sending and receiving media. - Exchange connection descriptions. The endpoints negotiate which media and connection details they can use. They need a signaling mechanism to pass offers, answers, and later ICE candidates to one another.
- Find a network route with ICE. ICE gathers possible connection routes and tests compatible options. STUN helps discover network-facing addresses and restrictions. If direct connectivity cannot be established, TURN can relay the media.
- Send and receive media securely. Once a usable candidate pair is selected, media can flow. RFC 8835 specifies secure RTP for media and DTLS-SRTP for key exchange; WebRTC data channels use SCTP over DTLS over ICE. IETF RFC 8835, published January 2021.
- Distribute streams to multiple participants when needed. For multi-party conferencing, an SFU (Selective Forwarding Unit) can receive and forward participant streams to recipients. It is an architecture option, not a requirement for every WebRTC connection.
Signaling is separate from the media connection
WebRTC does not prescribe how signaling messages travel. The application must provide a way for endpoints to exchange their offers, answers, and ICE candidates. A signaling server can route those messages without interpreting their contents; WebSocket is one possible transport, not a WebRTC requirement. MDN’s signaling and video-calling guide describes the exchange.
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Signaling traffic and media traffic are distinct. A server may help two endpoints coordinate even when their media later travels directly. Conversely, ICE may select a TURN relay, or an SFU may sit in the media path by design. “Peer-to-peer” therefore does not mean every session is guaranteed to send media directly between users.
Direct peer delivery and SFU delivery
Direct delivery can suit a small exchange: endpoints send media to one another without an SFU forwarding the streams. As the number of recipients grows, a sender in a mesh arrangement may need to send a separate stream to each recipient. A common multi-party alternative is an SFU, which forwards streams to participants.
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| Consideration | Direct peer delivery | SFU-mediated delivery |
|---|---|---|
| How media reaches recipients | Endpoints send media directly to one another when a route is available. | An intermediary forwards participant streams to recipients. |
| Sender’s recipient fan-out | In a mesh, a sender may need to send separate streams to multiple recipients. | The SFU handles forwarding to recipients. |
| Operational components | The application still needs signaling and may need TURN for relayed connectivity. | The application needs signaling and an SFU; TURN may also be needed for network connectivity. |
| When to consider it | When a small group can exchange media directly and that delivery pattern suits the application. | When an application needs an intermediary to forward streams among multiple participants. |
These are qualitative architecture choices, not a universal participant-count rule. The right design depends on recipient fan-out, network variation, media needs, and the services the application can operate. The available standards and implementation guides do not establish a universal crossover point, capacity, latency, or bandwidth saving for choosing one architecture over the other. MDN discusses SFUs in its WebRTC protocols overview.
What WebRTC does—and does not—guarantee
- It provides APIs and protocols, not a complete hosted application. The application still has to handle signaling and decide how participants and media are managed.
- It attempts to find a usable route. Direct connectivity is not guaranteed; ICE can use a TURN relay.
- It supports secure media transport. RFC 8835 specifies secure RTP and DTLS-SRTP key exchange.
- It does not guarantee a fixed latency or quality. Network conditions and implementation choices affect a session. Do not assume one performance figure applies to every deployment.
- It is not the same as a one-to-many prerecorded broadcast service. WebRTC is commonly used for interactive real-time exchange; the delivery architecture and media source depend on the application.
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