# young-webrtc-sdk

**Repository Path**: BluesYoung-web/young-webrtc-sdk

## Basic Information

- **Project Name**: young-webrtc-sdk
- **Description**: 对于 webrtc 操作的封装
- **Primary Language**: TypeScript
- **License**: MIT
- **Default Branch**: master
- **Homepage**: None
- **GVP Project**: No

## Statistics

- **Stars**: 1
- **Forks**: 0
- **Created**: 2021-11-21
- **Last Updated**: 2025-01-23

## Categories & Tags

**Categories**: Uncategorized

**Tags**: None

## README

# 浏览器使用 webrtc 推拉媒体流

- [代码参考内容](http://ossrs.net/players/js/srs.sdk.js)

## RTC 对等连接建立流程

```js
const localPC = new RTCPeerConnection();
const remotePC = new RTCPeerConnection();
// 创建本地 offer
const localOffer = await localPC.createOffer();
// 设置本地描述
localPC.setLocalDescription(localOffer);
// 发送本地 offer 的 sdp 给对端，返回对端的 answer -> session
const session = sendSdp(localOffer.sdp);

// 对端接受 offer，作为远程描述
await remotePC.setRemoteDescription(new RTCSessionDescription({
  type: 'offer',
  sdp: localOffer.sdp
}));
// 对端创建 answer，返回给本地
const answer = await remotePC.createAnswer();

// 将对端的 answer sdp 设置为远程描述
localPC.setRemoteDescription(new RTCSessionDescription({
  type: 'answer',
  sdp: session.sdp
}));
```

## `/lib/reciver.ts`

- 接收(播放)部分

```ts
import { WebRtcReciver } from './lib/reciver';

const rtc = new WebRtcReciver();
const play = async () => {
  const video = document.querySelector('#live_video') as HTMLVideoElement;
  // 先将直播流对象赋值给 video 的属性
  video.srcObject = rtc.stream;
  try {
    await rtc.play('webrtc://192.168.31.216:1985/live/450564');
    await video.play();
  } catch (error) {
    console.log('------播放出错------');
    console.log(error);
  }
};
```

## `/lib/pusher.ts`

- 推送(推流)部分

```ts
import { WebRtcPusher } from './lib/pusher';

const rtc = new WebRtcPusher();
const push = async () => {
  try {
    await rtc.push('webrtc://192.168.31.216:1985/live/450564');
  } catch (error) {
    console.log('------推流出错------');
    console.log(error);
  }
}
```