WebRTC is a set of browser APIs and network protocols for real-time audio, video, and data communication. It does not provide a complete calling service: your application still needs a way to connect users and exchange setup information, and may need servers to relay traffic. The browser APIs and the protocols behind them are distinct parts of the WebRTC effort.
What WebRTC is—and what it is not
WebRTC combines two standards efforts. The W3C specifies browser-facing APIs that allow authorized page code to use communication features; the IETF specifies protocols for communication between implementations. As a result, WebRTC is not just a JavaScript API, one signaling protocol, or a hosted video-calling product. An implementation can use WebRTC protocols without exposing the browser JavaScript API.
The goal, as described in RFC 8825, is interoperable audio, video, and data communication along the most direct possible path between participants. The endpoints do not have to be browsers. The application decides how users find one another, how they authenticate, and how their connections are established and managed.
What the browser APIs do
Browser APIs give an application a way to request access to local media when needed, coordinate a peer connection, negotiate session descriptions, and work with media tracks and connection state. Device selection and capture behavior depend on the browser and operating system. The application should make capture controls and permission state clear to the user.
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What the protocols do
The protocols define how endpoints establish connectivity and transport media or data. Secure RTP is used for media, with DTLS used for key exchange; data channels use SCTP over DTLS over ICE. These are different transport paths within the suite, even though an application may coordinate them through the same peer connection.
How a WebRTC connection works
A useful way to understand a call is to separate the application’s signaling path from the media and data paths. Signaling establishes and manages the connection; it is not itself the audio or video transport. Once connectivity is negotiated, media may flow directly between endpoints or through a relay or other service infrastructure.
- Connect users and establish signaling. The application determines who is calling whom, authenticates participants as needed, and provides a way to exchange setup messages. WebRTC does not prescribe this channel.
- Request local media when the feature requires it. Ask for microphone or camera access at an appropriate point in the user flow. Handle denied or revoked permissions and unavailable devices rather than assuming capture will succeed.
- Negotiate the peer connection. The browser coordinates tracks, session descriptions, and connection state. The application exchanges the session descriptions with the other endpoint over its signaling path.
- Exchange ICE candidates. ICE gathers and checks possible network paths, then selects a candidate pair that can connect. Candidate information must also be conveyed through signaling.
- Send media or data on the appropriate path. Media uses secure RTP; data channels use SCTP over DTLS over ICE. Their delivery characteristics and application purposes differ, so decide deliberately whether each data channel needs ordering or reliability.
- Observe state and recover. Monitor connection and media state, respond to device changes or permission loss, and design reconnect behavior. An ICE restart may be appropriate after connectivity changes; relay use may be necessary on networks where direct connectivity fails.
What signaling is, and why WebRTC leaves it to you
Signaling is the application-level exchange used to establish, manage, and control communication paths. It commonly carries offers, answers, and ICE candidates between participants. WebRTC defines no mandatory signaling transport or message format, so applications can choose a design that fits their identity, routing, and service architecture.
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HTTPS, WebSockets, SIP, or another application design can carry signaling. WebSockets are a common option for a persistent bidirectional messaging channel, but choosing WebSockets does not supply media capture, NAT traversal, or secure RTP. Keeping signaling logic separate from the browser’s media transport responsibilities makes it easier to reason about failures in each part.
ICE, STUN, and TURN: how endpoints find a route
ICE is the framework that checks possible network paths and selects one that works through NATs and firewalls. A direct path may be possible, but it is not guaranteed, and WebRTC should not be described as always peer-to-peer in the strict sense.
STUN helps discover and check connectivity
STUN helps an endpoint discover a server-reflexive address and supports connectivity checks. A STUN server alone is not a universal fix: some network configurations prevent a usable direct path.
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TURN relays traffic when direct connectivity fails
TURN allocates a relay address so traffic can pass through a server when direct communication is not viable. RFC 8835 requires full ICE support and TURN support for cases involving endpoint-dependent NAT mappings. It also requires TURN over TCP and TURN over TLS/TCP support for networks whose firewalls block UDP.
A TURN relay adds a server to the media or data path, with corresponding operational and infrastructure costs. Conferencing services may also route media through servers intentionally for functions such as scaling, recording, moderation, or mixing. Those architectural choices do not change whether the endpoints use WebRTC protocols.
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| Technology | What it provides | Relationship to WebRTC |
|---|---|---|
| WebRTC | Browser-facing APIs plus real-time network protocols | Provides standardized capture and connection APIs and secure media and data transport. The application still chooses signaling and service architecture. |
| WebSocket | A bidirectional application messaging channel | Can carry WebRTC signaling messages. It does not itself provide media capture, ICE traversal, codecs, or secure RTP media transport. |
| SIP | A signaling protocol and broader telephony ecosystem | Can be part of a WebRTC application or gateway design, but SIP and WebRTC are not synonyms. Interworking may require compatible media negotiation, codecs, and security. |
| HTTP polling or ordinary client/server APIs | Request/response application communication | Useful for many application operations, but not a direct replacement for interactive real-time media transport. |
When choosing an implementation or architecture, compare the client types you need to support; whether media will be direct, relayed, or server-routed; behavior on restrictive networks; control and operating cost for signaling, TURN, or media servers; security and permission handling; and the need for observability, recording, moderation, or scale.
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Security and privacy decisions
WebRTC media transport is designed around secure RTP and DTLS-based key exchange. That does not make the calling application inherently trustworthy. RFC 8826 emphasizes that the web service controls signaling and, ultimately, the JavaScript application logic. The service’s security, authorization, user interface, permission handling, and protection from malicious code therefore remain consequential.
- Request microphone or camera access only when the feature needs it, and provide understandable controls for starting and stopping capture.
- Authenticate and authorize participants through the application’s own service design; transport encryption does not decide who is allowed into a call.
- Protect signaling and the application code that controls it. A secure media transport cannot compensate for compromised application logic or misleading permission flows.
- Decide deliberately whether connections should be direct, relayed, or routed through conferencing infrastructure, based on the service’s privacy and operational needs.
Deployment, reliability, and costs
WebRTC is a protocol and API foundation, not a turnkey hosted calling service. A production application commonly needs signaling and may need TURN, authentication, recording, or conferencing infrastructure. Which of those components are necessary depends on product requirements and the networks its users connect from.
There is no single WebRTC service price established here. Cost depends on the signaling and relay or media infrastructure you operate or use, along with any product-specific services. In particular, planning only for STUN can leave users on restrictive networks unable to connect; include a TURN strategy and account for relay use in your operational design.
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For browser support, verify the current W3C WebRTC specification and test the browser and operating-system combinations your application intends to support. Browser behavior changes, and a universal current compatibility matrix is not established here.
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