From da78a9e329e272dedb2400b79a3bdeebff387d47 Mon Sep 17 00:00:00 2001 From: Peter Bengtsson Date: Tue, 8 Dec 2020 14:42:17 -0500 Subject: initial commit --- files/id/web/api/webrtc_api/index.html | 185 +++++++++++++++++++++++++++++++++ 1 file changed, 185 insertions(+) create mode 100644 files/id/web/api/webrtc_api/index.html (limited to 'files/id/web/api/webrtc_api/index.html') diff --git a/files/id/web/api/webrtc_api/index.html b/files/id/web/api/webrtc_api/index.html new file mode 100644 index 0000000000..6d9aa7392c --- /dev/null +++ b/files/id/web/api/webrtc_api/index.html @@ -0,0 +1,185 @@ +--- +title: WebRTC API +slug: Web/API/WebRTC_API +translation_of: Web/API/WebRTC_API +--- +
{{APIRef("WebRTC")}}
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WebRTC (Web Real-Time Communications) merupakan teknologi yang memungkinkan situs dan aplikasi berbasis web menangkap dan mengirimkan audio atau video, serta bertukar data antar peramban web tanpa memerlukan adanya perantara. Kumpulan standar di dalam WebRTC membuat pertukaran data dan telekonferensi secara peer-to-peer dapat terjadi tanpa perlu memasang plugins atau aplikasi pihak ketiga.

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WebRTC terdiri dari berbagai API dan protokol yang saling terkait sehingga dapat bekerja sebagaimana mestinya. Dokumentasi ini akan membantu anda untuk memahami dasar-dasar WebRTC, bagaimana mengatur dan menggunakan sambungan data dan media, serta hal lainnya.

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Konsep dan cara penggunaan WebRTC

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WebRTC memiliki beberapa fungsi, begitu halnya juga dengan Media Capture and Streams ({{domxref("MediaDevices")}}) API. Keduanya menghadirkan kemampuan multimedia di dalam web, misalnya audio dan video conferencing, pertukaran file, identity management, serta interaksi pada jaringan telepon dengan cara mengirimkan sinyal {{Glossary("DTMF")}}. Sambungan antar peer dapat terbentuk tanpa plugins tambahan, dan sering kali dapat terbentuk tanpa memerlukan server perantara.

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Sambungan antar peer dibuat menggunakan - dan direpresentasikan oleh - {{domxref("RTCPeerConnection")}}. Ketika sebuah sambungan telah terbentuk dan terbuka, stream media ({{domxref("MediaStream")}}) atau channel data ({{domxref("RTCDataChannel")}}) dapat ditambahkan ke dalam sambungan peer.

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Stream media terdiri dari beberapa track media, yang direpresentasikan oleh sebuah objek {{domxref("MediaStreamTrack")}}, yang berisi salah satu tipe media, yaitu audio, video, dan teks (misalnya subtitle atau nama judul). Umumnya stream terdiri dari setidaknya satu track audio bahkan juga satu track video, dan dapat digunakan untuk mengirim dan menerima media secara live atau dapat berupa rekaman media (misalnya streaming film).

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Anda juga dapat menggunakan sambungan antar peer untuk bertukar sembarang data biner menggunakan {{domxref("RTCDataChannel")}}. Sambungan ini dapat digunakan untuk informasi back-channel, metadatagamingfile transfer, bahkan sebagai jalur utama pertukaran data.

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artikel ini memerlukan informasi yang lebih lanjut serta tautan untuk panduan dan tutorial yang relevan

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WebRTC interfaces

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Because WebRTC provides interfaces that work together to accomplish a variety of tasks, we have divided up the interfaces in the list below by category. Please see the sidebar for an alphabetical list.

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Connection setup and management

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These interfaces are used to set up, open, and manage WebRTC connections.

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{{domxref("RTCPeerConnection")}}
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Represents a WebRTC connection between the local computer and a remote peer. It is used to handle efficient streaming of data between the two peers.
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{{domxref("RTCDataChannel")}}
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Represents a bi-directional data channel between two peers of a connection.
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{{domxref("RTCDataChannelEvent")}}
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Represents events that occur while attaching a {{domxref("RTCDataChannel")}} to a {{domxref("RTCPeerConnection")}}. The only event sent with this interface is {{event("datachannel")}}.
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{{domxref("RTCSessionDescription")}}
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Represents the parameters of a session. Each RTCSessionDescription consists of a description type indicating which part of the offer/answer negotiation process it describes and of the SDP descriptor of the session.
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{{domxref("RTCStatsReport")}}
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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()")}}.
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{{domxref("RTCIceCandidate")}}
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Represents a candidate internet connectivity establishment (ICE) server for establishing an {{domxref("RTCPeerConnection")}}.
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{{domxref("RTCIceTransport")}}
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Represents information about an internet connectivity establishment (ICE) transport.
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{{domxref("RTCPeerConnectionIceEvent")}}
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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")}}.
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{{domxref("RTCRtpSender")}}
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Manages the encoding and transmission of data for a {{domxref("MediaStreamTrack")}} on an {{domxref("RTCPeerConnection")}}.
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{{domxref("RTCRtpReceiver")}}
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Manages the reception and decoding of data for a {{domxref("MediaStreamTrack")}} on an {{domxref("RTCPeerConnection")}}.
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{{domxref("RTCTrackEvent")}}
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Indicates that a new incoming {{domxref("MediaStreamTrack")}} was created and an associated {{domxref("RTCRtpReceiver")}} object was added to the {{domxref("RTCPeerConnection")}} object.
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Identity and security

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The WebRTC API includes a number of interfaces to manage security and identity.

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{{domxref("RTCIdentityProvider")}}
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Enables a user agent is able to request that an identity assertion be generated or validated.
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{{domxref("RTCIdentityAssertion")}}
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Represents the identity of the a remote peer of the current connection. If no peer has yet been set and verified this interface returns null. Once set it can't be changed.
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{{domxref("RTCIdentityProviderRegistrar")}}
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Registers an  identity provider (idP).
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{{domxref("RTCIdentityEvent")}}
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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")}}.
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{{domxref("RTCIdentityErrorEvent")}}
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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")}}.
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{{domxref("RTCCertificate")}}
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Represents a certificate that an {{domxref("RTCPeerConnection")}} uses to authenticate.
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Telephony

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

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{{domxref("RTCDTMFSender")}}
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Manages the encoding and transmission of dual-tone multi-frequency (DTMF) signaling for an {{domxref("RTCPeerConnection")}}.
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{{domxref("RTCDTMFToneChangeEvent")}}
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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).
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Guides

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Introduction to WebRTC protocols
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This article introduces the protocols on top of which the WebRTC API is built.
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WebRTC connectivity
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A guide to how WebRTC connections work and how the various protocols and interfaces can be used together to build powerful communication apps.
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Lifetime of a WebRTC session
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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.
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Signaling and two-way video calling
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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.
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Using WebRTC data channels
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This guide covers how you can use a peer connection and an associated {{domxref("RTCDataChannel")}} to exchange arbitrary data between two peers.
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Using DTMF with WebRTC
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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.
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Tutorials

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Improving compatibility using WebRTC adapter.js
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The WebRTC organization provides on GitHub the WebRTC adapter 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.
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Taking still photos with WebRTC
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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.
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A simple RTCDataChannel sample
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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 WebSocket API, so that the same programming model can be used for each.
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Resources

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Protocols

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WebRTC-proper protocols

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Specifications

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SpecificationStatusComment
{{SpecName('WebRTC 1.0')}}{{Spec2('WebRTC 1.0')}}The initial definition of the API of WebRTC.
{{SpecName('Media Capture')}}{{Spec2('Media Capture')}}The initial definition of the object conveying the stream of media content.
{{SpecName('Media Capture DOM Elements')}}{{Spec2('Media Capture DOM Elements')}}The initial definition on how to obtain stream of content from DOM Elements
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In additions to these specifications defining the API needed to use WebRTC, there are several protocols, listed under resources.

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