---
title: WebRTC API
slug: Web/API/WebRTC_API
tags:
  - API
  - Audio
  - Landing
  - Media
  - NeedsTranslation
  - Networking
  - TopicStub
  - Video
  - WebRTC
  - WebRTC API
  - streaming
translation_of: Web/API/WebRTC_API
---
<div>{{APIRef("WebRTC")}}</div>

<p><span class="seoSummary"><strong>WebRTC</strong> (Web Real-Time Communications) é uma tecnologia que permite aplicativos e sites da Web a capturarem e opcionalmente transmitirem mídia de áudio e/ou vídeo, assim como trocar informação arbitrária entre navegadores sem a necessidade de um intermediador. O conjunto de padrões que abrangem WebRTC possibilita o compartilhamento de informação e a realização de teleconferência peer-to-peer, dispensando a instalação de <em>plug-ins</em> ou quaisquer softwares de terceiros.</span></p>

<p>WebRTC consiste em diversas APIs e protocolos interrelacionados que trabalham juntos. A documentação que você encontrará aqui o ajudará a entender os fundamentos de WebRTC, como configurar e usar, tanto informação, quanto conexões de mídia e mais. </p>

<h2 id="WebRTC_conceitos_e_uso">WebRTC conceitos e uso</h2>

<p>WebRTC serve à diversas propostas, e sobrepõe-se substancialmente com a API de Captura e Transmissão. Juntas, provém capacidades poderosas de multimídia para a Web, incluíndo suporte para conferência em áudio e vídeo, troca de arquivos, administração de identidade, e lidando com sistemas telefônicos legados enviando sinais {{Glossary("DTMF")}}.</p>

<p>Conexões entre dois <em>peers </em>são criadas usando—e representadas por—uma interface {{DOMxRef("RTCPeerConnection")}}. Uma vez que a conexão tenha sido estabilizada e iniciada, media streams ({{DOMxRef("MediaStream")}}s) (transmissões de mídia) e/ou data channels ({{DOMxRef("RTCDataChannel")}}s) (canais de dados) podem ser adicionados à conexão.</p>

<p>Dados de mídia podem consistir em qualquer número de <em>tracks</em>(faixas) de dados de mídia; <em>tracks</em>, que são representados por objetos baseados na interface {{DOMxRef("MediaStreamTrack")}} , que podem conter um número dentre tipos de dados de mídia, incluíndo áudio, vídeo e texto (como legendas ou até mesmo nomes de capítulos). A maioria das transmissões consiste de ao menos uma faixa de áudio e comumente também uma faixa de vídeo, e podem ser usadas para enviar e receber ambas as mídias ao vivo ou dados salvos de mídia (como um filme transmitido).</p>

<p>Você também pode usar a conexão entre dois <em>peers </em>para trocar dados arbitrários binários usando a interface {{DOMxRef("RTCDataChannel")}}. Isto pode ser usado para informação de <em>back-channel </em>(canal de volta), troca de <em>metadata</em>, pacotes de <em>status</em> de games, transferência de arquivos, ou até mesmo como um canal primário para transferir dados.</p>

<p><em><strong>são necessários mais detalhes e links e tutoriais relevantes</strong></em></p>

<h2 id="WebRTC_interfaces">WebRTC interfaces</h2>

<p>Porque WebRTC provê interfaces que trabalham em conjunto para realizar uma variedade de tarefas, nós dividimos as interfaces na listagem abaixo por categoria. Por favor, atente-se a sidebar para uma listagem em ordem alfabética.</p>

<h3 id="Conexão_configuração_e_gerenciamento">Conexão, configuração e gerenciamento</h3>

<p>Estas interfaces são usadas para configurar, abrir e gerenciar conexões WebRTC.</p>

<dl>
 <dt>{{domxref("RTCPeerConnection")}}</dt>
 <dd>Representa a conexão WebRTC entre o computador local e um peer remoto. É usado para lidar com um streaming eficiente de dados entre os dois peers.</dd>
 <dt>{{domxref("RTCDataChannel")}}</dt>
 <dd>Representa um canal de dados bidirecional entre dois peers de uma conexão</dd>
 <dt>{{domxref("RTCDataChannelEvent")}}</dt>
 <dd>Represents events that occur while attaching a {{domxref("RTCDataChannel")}} to a {{domxref("RTCPeerConnection")}}. The only event sent with this interface is {{event("datachannel")}}.</dd>
 <dt>{{domxref("RTCSessionDescription")}}</dt>
 <dd>Represents the parameters of a session. Each <code>RTCSessionDescription</code> consists of a description <a href="/en-US/docs/Web/API/RTCSessionDescription/type" title="The property RTCSessionDescription.type is an enum of type RTCSdpType describing the description's type."><code>type</code></a> indicating which part of the offer/answer negotiation process it describes and of the <a href="/en-US/docs/Glossary/SDP" title="SDP: SDP (Session Description Protocol) is the standard describing a peer-to-peer connection. SDP contains the codec, source address, and timing information of audio and video.">SDP</a> descriptor of the session.</dd>
 <dt>{{domxref("RTCStatsReport")}}</dt>
 <dd>Provides information detailing statistics for a connection or for an individual track on the connection; the report can be obtained by calling {{domxref("RTCPeerConnection.getStats()")}}.</dd>
 <dt>{{domxref("RTCIceCandidate")}}</dt>
 <dd>Represents a candidate internet connectivity establishment (ICE) server for establishing an {{domxref("RTCPeerConnection")}}.</dd>
 <dt>{{domxref("RTCIceTransport")}}</dt>
 <dd>Represents information about an internet connectivity establishment (ICE) transport.</dd>
 <dt>{{domxref("RTCPeerConnectionIceEvent")}}</dt>
 <dd>Represents events that occurs in relation to ICE candidates with the target, usually an {{domxref("RTCPeerConnection")}}. Only one event is of this type: {{event("icecandidate")}}.</dd>
 <dt>{{domxref("RTCRtpSender")}}</dt>
 <dd>Manages the encoding and transmission of data for a {{domxref("MediaStreamTrack")}} on an {{domxref("RTCPeerConnection")}}.</dd>
 <dt>{{domxref("RTCRtpReceiver")}}</dt>
 <dd>Manages the reception and decoding of data for a {{domxref("MediaStreamTrack")}} on an {{domxref("RTCPeerConnection")}}.</dd>
 <dt>{{domxref("RTCTrackEvent")}}</dt>
 <dd>Indicates that a new incoming {{domxref("MediaStreamTrack")}} was created and an associated {{domxref("RTCRtpReceiver")}} object was added to the {{domxref("RTCPeerConnection")}} object.</dd>
</dl>

<h3 id="Identity_and_security">Identity and security</h3>

<p>The WebRTC API includes a number of interfaces to manage security and identity.</p>

<dl>
 <dt>{{domxref("RTCIdentityProvider")}}</dt>
 <dd>Enables a user agent is able to request that an identity assertion be generated or validated.</dd>
 <dt>{{domxref("RTCIdentityAssertion")}}</dt>
 <dd>Represents the identity of the a remote peer of the current connection. If no peer has yet been set and verified this interface returns <code>null</code>. Once set it can't be changed.</dd>
 <dt>{{domxref("RTCIdentityProviderRegistrar")}}</dt>
 <dd>Registers an  identity provider (idP).</dd>
 <dt>{{domxref("RTCIdentityEvent")}}</dt>
 <dd>Represents an identity assertion generated by an identity provider (idP). This is usually for an {{domxref("RTCPeerConnection")}}. The only event sent with this type is {{event("identityresult")}}.</dd>
 <dt>{{domxref("RTCIdentityErrorEvent")}}</dt>
 <dd>Represents an error associated with the identity provider (idP). This is usually for an {{domxref("RTCPeerConnection")}}. Two events are sent with this type: {{event("idpassertionerror")}} and {{event("idpvalidationerror")}}.</dd>
 <dt>{{domxref("RTCCertificate")}}</dt>
 <dd>Represents a certificate that an {{domxref("RTCPeerConnection")}} uses to authenticate.</dd>
</dl>

<h3 id="Telephony">Telephony</h3>

<p>These interfaces are related to interactivity with public-switched telephone networks (PTSNs).</p>

<dl>
 <dt>{{domxref("RTCDTMFSender")}}</dt>
 <dd>Manages the encoding and transmission of dual-tone multi-frequency (DTMF) signaling for an {{domxref("RTCPeerConnection")}}.</dd>
 <dt>{{domxref("RTCDTMFToneChangeEvent")}}</dt>
 <dd>Indicates an occurrence of a of dual-tone multi-frequency (DTMF). This event does not bubble (except where otherwise stated) and is not cancelable (except where otherwise stated).</dd>
</dl>

<h2 id="Guides">Guides</h2>

<dl>
 <dt><a href="/en-US/docs/Web/API/WebRTC_API/Protocols">Introduction to WebRTC protocols</a></dt>
 <dd>This article introduces the protocols on top of which the WebRTC API is built.</dd>
 <dt><a href="/en-US/docs/Web/API/WebRTC_API/Connectivity">WebRTC connectivity</a></dt>
 <dd>A guide to how WebRTC connections work and how the various protocols and interfaces can be used together to build powerful communication apps.</dd>
 <dt><a href="/en-US/docs/Web/API/WebRTC_API/Session_lifetime">Lifetime of a WebRTC session</a></dt>
 <dd>WebRTC lets you build peer-to-peer communication of arbitrary data, audio, or video—or any combination thereof—into a browser application. In this article, we'll look at the lifetime of a WebRTC session, from establishing the connection all the way through closing the connection when it's no longer needed.</dd>
 <dt><a href="/en-US/docs/Web/API/WebRTC_API/Signaling_and_video_calling">Signaling and two-way video calling</a></dt>
 <dd>A tutorial and example which turbs a WebSocket-based chat system created for a previous example and adds support for opening video calls among participants. The chat server's WebSocket connection is used for WebRTC signaling.</dd>
 <dt><a href="/en-US/docs/Web/API/WebRTC_API/Using_data_channels">Using WebRTC data channels</a></dt>
 <dd>This guide covers how you can use a peer connection and an associated {{domxref("RTCDataChannel")}} to exchange arbitrary data between two peers.</dd>
 <dt><a href="/en-US/docs/Web/API/WebRTC_API/Using_DTMF">Using DTMF with WebRTC</a></dt>
 <dd>WebRTC's support for interacting with gateways that link to old-school telephone systems includes support for sending DTMF tones using the {{domxref("RTCDTMFSender")}} interface. This guide shows how to do so.</dd>
</dl>

<h2 id="Tutorials">Tutorials</h2>

<dl>
 <dt><a href="/en-US/docs/Web/API/WebRTC_API/adapter.js">Improving compatibility using WebRTC adapter.js</a></dt>
 <dd>The WebRTC organization <a href="https://github.com/webrtc/adapter/">provides on GitHub the WebRTC adapter</a> to work around compatibility issues in different browsers' WebRTC implementations. The adapter is a JavaScript shim which lets your code to be written to the specification so that it will "just work" in all browsers with WebRTC support.</dd>
 <dt><a href="/en-US/docs/Web/API/WebRTC_API/Taking_still_photos">Taking still photos with WebRTC</a></dt>
 <dd>This article shows how to use WebRTC to access the camera on a computer or mobile phone with WebRTC support and take a photo with it.</dd>
 <dt><a href="/en-US/docs/Web/API/WebRTC_API/Simple_RTCDataChannel_sample">A simple RTCDataChannel sample</a></dt>
 <dd>The {{domxref("RTCDataChannel")}} interface is a feature which lets you open a channel between two peers over which you may send and receive arbitrary data. The API is intentionally similar to the <a href="/en-US/docs/Web/API/WebSocket_API">WebSocket API</a>, so that the same programming model can be used for each.</dd>
</dl>

<h2 id="Resources">Resources</h2>

<h3 id="Protocols">Protocols</h3>

<h4 id="WebRTC-proper_protocols">WebRTC-proper protocols</h4>

<ul>
 <li><a href="http://datatracker.ietf.org/doc/draft-ietf-rtcweb-alpn/"><cite>Application Layer Protocol Negotiation for Web Real-Time Communications</cite></a></li>
 <li><a href="http://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio/"><cite>WebRTC Audio Codec and Processing Requirements</cite></a></li>
 <li><a href="http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/"><cite>RTCWeb Data Channels</cite></a></li>
 <li><a href="http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/"><cite>RTCWeb Data Channel Protocol</cite></a></li>
 <li><a href="http://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage/"><cite>Web Real-Time Communication (WebRTC): Media Transport and Use of RTP</cite></a></li>
 <li><a href="http://datatracker.ietf.org/doc/draft-ietf-rtcweb-security-arch/"><cite>WebRTC Security Architecture</cite></a></li>
 <li><a href="http://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/"><cite>Transports for RTCWEB</cite></a></li>
</ul>

<h4 id="Related_supporting_protocols">Related supporting protocols</h4>

<ul>
 <li><a href="https://tools.ietf.org/html/rfc5245">Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocol</a></li>
 <li><a href="https://tools.ietf.org/html/rfc5389"><cite>Session Traversal Utilities for NAT (STUN)</cite></a></li>
 <li><a href="https://tools.ietf.org/html/rfc7064"><cite>URI Scheme for the Session Traversal Utilities for NAT (STUN) Protocol</cite></a></li>
 <li><a href="https://tools.ietf.org/html/rfc7065"><cite>Traversal Using Relays around NAT (TURN) Uniform Resource Identifiers</cite></a></li>
 <li><a href="https://tools.ietf.org/html/rfc3264"><cite>An Offer/Answer Model with Session Description Protocol (SDP)</cite></a></li>
 <li><a href="https://datatracker.ietf.org/doc/draft-ietf-tram-turn-third-party-authz/"><cite>Session Traversal Utilities for NAT (STUN) Extension for Third Party Authorization</cite></a></li>
</ul>

<h2 id="Specifications">Specifications</h2>

<table class="standard-table">
 <thead>
  <tr>
   <th scope="col">Specification</th>
   <th scope="col">Status</th>
   <th scope="col">Comment</th>
  </tr>
 </thead>
 <tbody>
  <tr>
   <td>{{SpecName('WebRTC 1.0')}}</td>
   <td>{{Spec2('WebRTC 1.0')}}</td>
   <td>The initial definition of the API of WebRTC.</td>
  </tr>
  <tr>
   <td>{{SpecName('Media Capture')}}</td>
   <td>{{Spec2('Media Capture')}}</td>
   <td>The initial definition of the object conveying the stream of media content.</td>
  </tr>
  <tr>
   <td>{{SpecName('Media Capture DOM Elements')}}</td>
   <td>{{Spec2('Media Capture DOM Elements')}}</td>
   <td>The initial definition on how to obtain stream of content from DOM Elements</td>
  </tr>
 </tbody>
</table>

<p>In additions to these specifications defining the API needed to use WebRTC, there are several protocols, listed under <a href="#Protocols">resources</a>.</p>

<h2 class="Related_Topics" id="See_also">See also</h2>

<ul>
 <li>{{domxref("MediaDevices")}}</li>
 <li>{{domxref("MediaStreamEvent")}}</li>
 <li>{{domxref("MediaStreamConstraints")}}</li>
 <li>{{domxref("MediaStreamTrack")}}</li>
 <li>{{domxref("MessageEvent")}}</li>
 <li>{{domxref("MediaStream")}}</li>
 <li><a href="https://hacks.mozilla.org/2015/06/firefox-multistream-and-renegotiation-for-jitsi-videobridge/">Firefox multistream and renegotiation for Jitsi Videobridge</a></li>
 <li><a href="https://hacks.mozilla.org/2015/04/peering-through-the-webrtc-fog-with-socketpeer/">Peering Through the WebRTC Fog with SocketPeer</a></li>
 <li><a href="https://hacks.mozilla.org/2014/04/inside-the-party-bus-building-a-web-app-with-multiple-live-video-streams-interactive-graphics/">Inside the Party Bus: Building a Web App with Multiple Live Video Streams + Interactive Graphics</a></li>
</ul>