在一篇古老而已被废弃的文章中,我写了一篇关于通过websockets从画布上广播视频是多么容易和轻松的事情。在那篇文章中,我简要讨论了如何使用MediaStream API捕获来自摄像机的视频和来自麦克风的声音,如何对接收到的流进行编码并将其通过websocket发送到服务器。但是,实际上,他们不是这样做的,对于广播,他们使用需要安装和配置的特殊软件:可以是Open Broadcast Software,也可以使用WebRTC(即开即用),也就是说,它不需要安装任何插件(例如Flash Player),已经从12月开始从Chromium浏览器中删除。
今天我们将讨论WebRTC。
Web实时通信(WebRTC)并不是一个协议,它是标准,协议和JavaScript API的完整集合,它们共同提供实时的对等视频-音频通信,还可以用于传输任何二进制数据... 通常,浏览器充当对等方,但是例如,它也可以是移动应用程序。为了组织客户端之间的p2p通信,浏览器需要支持各种类型的视频和音频编码,支持许多网络协议,以确保硬件与浏览器的交互(通过OS层):网络摄像头,声卡。为了开发人员的方便,整个技术杂乱都隐藏在JavaScript API抽象的后面。
归结为三个API:
MediaStream API-我们上次对其进行了分析,今天我将详细介绍它。服务接收来自硬件的视频/音频流
RTCPeerConnection-提供两个客户端之间的通信(p2p)
RTCDataChannel-用于在两个客户端之间传输任意数据
准备音频和视频流以进行传输
"" . , : , , , . , -. , ( , ), . . 1:
, . . 2020 . , MediaStream API, . IE .
: , , , "" Media Stream <video> html. canvas , WebGL CSS3, , canvas , ( bigo live, twitch ). , , :
https://jeeliz.com/ - realtime CV Javascript. js- canvas: , , (, ) . , .
Canvas captureStream API - API canvas. Chrome, Opera Firefox
RTCPeerConnection
, ? RTCPeerConnection. , RTCPeerConnection:
const peerConnection = new RTCPeerConnection({
iceServers: [{
urls: 'stun:stun.l.google.com:19302'
}]
});
iceServers - , , NAT'. : ip , NAT ? ICE , , ICE WebRTC, .
Usermedia :
navigator.mediaDevices.getUserMedia({ video: true, audio: true }).then(stream => {
// Usermedia-,
const tracks = stream.getTracks();
for (const track of tracks) {
// peerConnection
peerConnection.addTrack(track);
}
}).catch(console.error);
peerConnection onnegotiationneeded, offer ( SDP - Session Description Protocol) peerConnection setLocalDescription. SDP - offer answer - .
LocalDescription peerConnection, "" ice-, NAT. onicegatheringstatechange. onicegatheringstatechange webrtc-signaling- stream Session Description :
peerConnection.oniceconnectionstatechange = (event) => {
console.log('Connection state: ', peerConnection.iceConnectionState);
if (peerConnection.iceConnectionState === 'connected') {
// Start broadcast
setBroadcasting(true);
setBroadcastingBtnActive(true);
}
};
// , peerConnection
peerConnection.onnegotiationneeded = (event) => {
// SDP offer
peerConnection.createOffer().
then((offer) => peerConnection.setLocalDescription(offer)).
catch(console.error);
};
// , ICE
peerConnection.onicegatheringstatechange = (ev) => {
let connection = ev.target;
// Now we can activate broadcast button
if (connection.iceGatheringState === 'complete') {
let delay = 50;
let tries = 0;
let maxTries = 3;
let timerId = setTimeout(function allowStreaming() {
if (isOnline) {
setBroadcastingBtnActive(true);
return;
}
if (tries < maxTries) {
tries += 1;
delay *= 2;
timerId = setTimeout(allowStreaming, delay);
} else {
// TODO: show user notification
console.error("Can't connect to server");
alert("Can't connect to server");
}
}, delay);
}
};
webrtc-signaling- - , session description , websocket xhr- . : session description .
Session descriptions , , ontrack peerConnection, , <video> . .
:
https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection - RTCPeerConnection
https://github.com/pion/webrtc - WebRTC go
https://webrtcforthecurious.com/ - pion
https://hpbn.co/ - High Perfomance Browser Networking. web-. WebRTC. (2013), .
pion, HLS ffmpeg .
: react pion twitch ( ).