/ArWebSDK

anyRTC Web SDK 开发的下一代Web RTC SDK。支持语音通话、视频通话、音频互动直播、视频互动直播等场景。anyrtc Web SDK 是一个全量重构的版本,主要针对 API 的易用性和内部架构做了较大的调整。

ArRTCWebSDK

English | 简体中文

The anyrtc Web SDK is the next-generation SDK of the current anyrtc Web SDK, enabling audio and video real-time communications based on anyrtc SD-RTNTM and implementing scenarios such as voice-only calls, video call, voice-only interactive broadcast, and video interactive broadcast. The anyrtc Web SDK makes full-scale refactoring to the internal architecture of the anyrtc Web SDK and improves usability of APIs.

npm install ar-rtc-sdk --save

Documentation Website

We provide some basic demos. For the online website, check out here. For the source code, check out here.

If you have some problems when using the anyrtc Web SDK, or have any suggestions, you can post new issue in this repo and we will reply as soon as possoble.

Overview

For detailed introduction and documentation, please go to Documentation Website. Here we briefly introduce the features of the anyrtc Web SDK:

  • Support Typescript
  • Using ES6 Promise
  • Track-based media objects

Here is the sample code to join the channel and publish local media automatically

import ArRTC from "ar-rtc-sdk"

const client = ArRTC.createClient()

async function startCall() {
  await client.join("APPID", "CHANNEL", "TOKEN");
  const audioTrack = await ArRTC.createMicrophoneAudioTrack();
  const videoTrack = await ArRTC.createCameraVideoTrack();

  await client.publish([audioTrack, videoTrack]);
}

startCall().then(/** ... **/).catch(console.error);