简单的WebRTC例子
简单的WebRTC例子
webrtc网上封装的很多，demo很多都是一个页面里实现的，今天实现了个完整的 ， A 发视频给 B。
A offer.html作为offer
<!DOCTYPE html>
<html id="home" lang="en">
<head>
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
<meta charset="utf-8">
<meta name="viewport" content="width=device-width, initial-scale=1.0, user-scalable=no">
<meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
<style>
p { padding: 1em; }
li {
border-bottom: 1px solid rgb(189, 189, 189);
border-left: 1px solid rgb(189, 189, 189);
padding: .5em;
}
</style>
</head>
<body>
<script>
var mediaConstraints = {
optional: [],
mandatory: {
OfferToReceiveAudio: false,
OfferToReceiveVideo: true
}
};
</script>
<script>
var offerer
,answererWin
window.RTCPeerConnection = window.mozRTCPeerConnection || window.webkitRTCPeerConnection;
window.RTCSessionDescription = window.mozRTCSessionDescription || window.RTCSessionDescription;
window.RTCIceCandidate = window.mozRTCIceCandidate || window.RTCIceCandidate;
navigator.getUserMedia = navigator.mozGetUserMedia || navigator.webkitGetUserMedia;
window.URL = window.webkitURL || window.URL;
window.iceServers = {
iceServers: [{
url: 'stun:23.21.150.121'
}
]
};
</script>
<script>
/* offerer */
function offererPeer(video_stream) {
offerer = new RTCPeerConnection(window.iceServers)
offerer.addStream(video_stream)
offerer.onaddstream = function (event) {
// 本地显示video
}
offerer.onicecandidate = function (event) {
if (!event || !event.candidate) return
sendToP2({
'action' : 'candidate',
'candidate' :event.candidate
})
}
offerer.createOffer(function (offer) {
offerer.setLocalDescription(offer)
sendToP2({
'action' : 'create',
'offer':offer
})
}, function() {}, mediaConstraints)
}
</script>
<script>
var video_constraints = {
mandatory: {},
optional: []
}
function getUserMedia(callback) {
var n = navigator
n.getMedia = n.webkitGetUserMedia || n.mozGetUserMedia
n.getMedia({
audio: false,
video: video_constraints
}, callback, onerror)
function onerror(e) {
alert(JSON.stringify(e, null, '\t'))
}
}
</script>
<script>
function sendToP2(data){
answererWin.postMessage(JSON.stringify(data) ,window.location)
}
function receiveMessage(data){
data = JSON.parse(data.data)
switch ( data.action) {
case 'answer' :
offerer.setRemoteDescription(new RTCSessionDescription(data.answer))
break
case "candidate":
offerer.addIceCandidate(new RTCIceCandidate(data.candidate))
break
}
console.log('msg' ,data)
}
window.addEventListener("message", receiveMessage, false)
answererWin = window.open('answer.html' ,'t')
getUserMedia(function (video_stream) {
offererPeer(video_stream)
});
</script>
</body>
</html>
B answer.html 作为answer
<!DOCTYPE html>
<html id="home" lang="en">
<head>
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
<meta charset="utf-8">
<meta name="viewport" content="width=device-width, initial-scale=1.0, user-scalable=no">
<meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
</head>
<body>
<article>
<div style="text-align:center;">
<div class="videos-container">
<video id="peer1-to-peer2" autoplay controls></video>
<h2>Offerer-to-Answerer</h2>
<h2>此页面刷新之后，必须重新刷新一下Offer页面</h2>
</div>
</div>
<script>
var mediaConstraints = {
optional: [],
mandatory: {
OfferToReceiveAudio: true,
OfferToReceiveVideo: true
}
};
</script>
<script>
var offerer, answerer;
var offererToAnswerer = document.getElementById('peer1-to-peer2');
window.RTCPeerConnection = window.mozRTCPeerConnection || window.webkitRTCPeerConnection;
window.RTCSessionDescription = window.mozRTCSessionDescription || window.RTCSessionDescription;
window.RTCIceCandidate = window.mozRTCIceCandidate || window.RTCIceCandidate;
navigator.getUserMedia = navigator.mozGetUserMedia || navigator.webkitGetUserMedia;
window.URL = window.webkitURL || window.URL;
window.iceServers = {
iceServers: [{
url: 'stun:23.21.150.121'
}
]
};
</script>
<script>
/* answerer */
function answererPeer(offer, video_stream) {
answerer = new RTCPeerConnection(window.iceServers);
// answerer.addStream(video_stream);
answerer.onaddstream = function (event) {
offererToAnswerer.src = URL.createObjectURL(event.stream);
offererToAnswerer.play();
};
answerer.onicecandidate = function (event) {
if (!event || !event.candidate) return;
sendToP1({
'action' : 'candidate',
'candidate' :event.candidate
})
//offerer.addIceCandidate(event.candidate);
};
answerer.setRemoteDescription(new RTCSessionDescription(offer));
answerer.createAnswer(function (answer) {
answerer.setLocalDescription(answer);
sendToP1({
'action' : 'answer' ,
'answer' : answer
})
//offerer.setRemoteDescription(answer);
}, function() {}, mediaConstraints);
}
function receiveMessage(data){
data = JSON.parse(data.data)
console.log(data)
switch(data.action){
case "create":
answererPeer(data.offer , data.stream)
break
case "candidate":
answerer.addIceCandidate(new RTCIceCandidate(data.candidate))
break
}
}
window.addEventListener("message", receiveMessage, false)
function sendToP1(data) {
opener.postMessage(JSON.stringify(data) , window.location)
}
</script>
</article>
</body>
</html>
demo用 postMessage传递数据， 业务使用可以用websocket
A 先 createOffer ，生成的offer 供自己setLocalDescription ，并发给B
B 拿A的offer ，setRemoteDescription(offer) , 然后 createAnswer ，生成的answer
供自己setLocalDescription ，并发给A
A 拿B的answer 设置 setRemoteDescription（answer)
A onicecandidate 事件被触发 将得到的通道发给B
B addIceCandidate(new RTCIceCandidate(candidate)) 建立通道
B onicecandidate 事件被触发 将得到的通道发给A
A addIceCandidate(new RTCIceCandidate(candidate)) 建立通道
通道建立后视频就可以共享了
使用时，打开：offer.html 即可。
简单的WebRTC例子
https://www.dearcloud.cn/2016/09/12/20200310-cnblogs-old-posts/20160912-简单的WebRTC例子/