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1.还是基于h5stream的h5splayer.js学习,还是基于websocket。
与原来的不同,在onopen里面要发送一些open指令
- ws.onopen = function () {
- inc.innerHTML += '.. connection open<br/>';
- var t = {
- type: "open"
- };
- ws.send(JSON.stringify(t))
- };
然后会onreceive里面会受到一些信息。
- connecting to server ..
- .. connection open
- receive:{
- "sdp" : "v=0\r\no=- 2612379890657624089 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE video\r\na=msid-semantic: WMS token1\r\nm=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 127\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:M7/v\r\na=ice-pwd:s+xvE6/hacBd5++3xPE+1qpT\r\na=ice-options:trickle\r\na=fingerprint:sha-256 44:D2:9C:A6:A3:9F:01:5C:AD:CE:86:E6:2F:E8:EF:C0:6D:26:68:F5:2E:6A:82:89:C8:87:74:42:C8:FC:7F:F5\r\na=setup:actpass\r\na=mid:video\r\na=extmap:2 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:4 urn:3gpp:video-orientation\r\na=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\na=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type\r\na=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/video-timing\r\na=extmap:10 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07\r\na=sendonly\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:96 H264/90000\r\na=rtcp-fb:96 goog-remb\r\na=rtcp-fb:96 transport-cc\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=fmtp:96 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f\r\na=rtpmap:97 rtx/90000\r\na=fmtp:97 apt=96\r\na=rtpmap:98 H264/90000\r\na=rtcp-fb:98 goog-remb\r\na=rtcp-fb:98 transport-cc\r\na=rtcp-fb:98 ccm fir\r\na=rtcp-fb:98 nack\r\na=rtcp-fb:98 nack pli\r\na=fmtp:98 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\r\na=rtpmap:99 rtx/90000\r\na=fmtp:99 apt=98\r\na=rtpmap:100 red/90000\r\na=rtpmap:101 rtx/90000\r\na=fmtp:101 apt=100\r\na=rtpmap:127 ulpfec/90000\r\na=ssrc-group:FID 3833096967 1073194061\r\na=ssrc:3833096967 cname:C1L/zoQ17dXx0SDt\r\na=ssrc:3833096967 msid:token1 video_label\r\na=ssrc:3833096967 mslabel:token1\r\na=ssrc:3833096967 label:video_label\r\na=ssrc:1073194061 cname:C1L/zoQ17dXx0SDt\r\na=ssrc:1073194061 msid:token1 video_label\r\na=ssrc:1073194061 mslabel:token1\r\na=ssrc:1073194061 label:video_label\r\n",
- "type" : "offer"
- }
- type:string length:2130
- receive:{
- "candidate" : "candidate:725387133 1 tcp 1518280447 169.254.250.71 50001 typ host tcptype passive generation 0 ufrag M7/v network-id 1",
- "sdpMLineIndex" : 0,
- "sdpMid" : "video",
- "type" : "remoteice"
- }
- type:string length:215
- receive:{
- "candidate" : "candidate:3525817373 1 tcp 1518214911 192.168.6.12 50001 typ host tcptype passive generation 0 ufrag M7/v network-id 2",
- "sdpMLineIndex" : 0,
- "sdpMid" : "video",
- "type" : "remoteice"
- }
- type:string length:214
- receive:{
- "candidate" : "candidate:2564955588 1 tcp 1518149375 192.168.1.16 50001 typ host tcptype passive generation 0 ufrag M7/v network-id 3",
- "sdpMLineIndex" : 0,
- "sdpMid" : "video",
- "type" : "remoteice"
- }
- type:string length:214
从type来看,第一个是offer,后面三个是remoteice
对应不同的消息,做不同的处理
对于offer(请求),创建RTCPeerConnection,参考:https://developer.mozilla.org/zh-CN/docs/Web/API/RTCPeerConnection
一个基本的RTCPeerConnection使用需要协调本地机器以及远端机器的连接,它可以通过在两台机器间生成Session Description的数据交换协议来实现。呼叫方发送一个offer(请求),被呼叫方发出一个answer(应答)来回答请求。双方-呼叫方以及被呼叫方,最开始的时候都要建立他们各自的RTCPeerConnection对象。
setRemoteDescription=>onaddstream=>URL.createObjectURL,video.play()
createAnswer,setLocalDescription
代码 rtc.html,虽然功能还没有实现。
- <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
- <html>
-
- <head>
- <title>websocket client</title>
- <script src="js/jquery-3.1.1.js"></script>
- <script src="js/adapter.js"></script>
- <script type="text/javascript">
-
- var count=0;
- function getImageString(arrayBuffer){
- var bytes = new Uint8Array(arrayBuffer);
- var binary= "";
- var len = bytes.byteLength;
- for (var i = 0; i < len; i++) {
- binary += String.fromCharCode( bytes[ i ] )
- }
- var src="https://img-blog.csdnimg.cn/2022010702372264874.png"+window.btoa( binary );//png,jpg都没有关系
- return src;
- }
-
- function loadImage(imgId,data){
- var reader = new FileReader();
- reader.onload = () => {
- var img = document.getElementById(imgId);
- img.src = getImageString(reader.result);
- };
- reader.readAsArrayBuffer(data);
- }
-
- function processRTCOffer(offer){
- var description=new RTCSessionDescription(offer);
- console.log("ProcessRTCOffer", offer);
- var config={M:[]};
- var option={optional:[{DtlsSrtpKeyAgreement: true}]};
- var connection=new RTCPeerConnection(config,option);
-
- connection.onicecandidate=function(event){//只要本地代理ICE 需要通过信令服务器传递信息给其他对等端时就会触发
- console.log("------------- RTCPeerConnection.onicecandidate",event);
-
- // if (event.candidate) {
- // var candidate;
- // //console.log("onIceCandidate currentice", event.candidate);
- // candidate = event.candidate;
- // //console.log("onIceCandidate currentice",candidate, JSON.stringify(candidate));
- // var candidateObj = JSON.parse(JSON.stringify(candidate));
- // candidateObj.type = "remoteice";
- // //console.log("onIceCandidate currentice new", candidateObj, JSON.stringify(candidateObj));
- // //ws.send(JSON.stringify(candidateObj));//h5spalyer.js里面有发送,但是测试发现这里不发送也能获取到视频。
- // }
- // else {
- // console.log("End of candidates.");
- // }
- };
- connection.onaddstream=function(mediaStreamEvent){//
- console.log("------------- RTCPeerConnection.onaddstream",mediaStreamEvent);
- var stream=mediaStreamEvent.stream;
- var video=document.getElementById('video1');
- video.src=URL.createObjectURL(stream);
- //需要引用Adapter.js,默认的URL.createObjectURL不支持MediaSource类型
- //Uncaught TypeError: Failed to execute 'createObjectURL' on 'URL': No function was found that matched the signature provided.
-
- video.play();//开始播放
- //Uncaught (in promise) DOMException
- //需要将video设置为muted(静音),或者由用户点击来播放。
- };
- connection.oniceconnectionstatechange=function(event){
- console.log("------------- RTCPeerConnection.oniceconnectionstatechange state: " + connection.iceConnectionState)
- };
- connection.setRemoteDescription(description); //=>onaddstream
- var offerOptions={mandatory:{offerToReceiveAudio: true, offerToReceiveVideo: true}};
- connection.createAnswer(offerOptions).then(function(answer){
- console.log("Create answer:",answer);//answer is RTCSessionDescription
- connection.setLocalDescription(answer,function(){ //=>onicecandidate
- console.log("ProcessRTCOffer createAnswer", answer);
- ws.send(JSON.stringify(answer));//=>onicecandidate
- });
- });
- ws.connection=connection;
- }
-
- function processRemoteIce(remoteIce){
- var connection=ws.connection;
- try {
- var iceCandidate = new RTCIceCandidate({
- sdpMLineIndex: remoteIce.sdpMLineIndex, candidate: remoteIce.candidate
- });
- console.log("ProcessRemoteIce", iceCandidate);
- //console.log("Adding ICE candidate :" + JSON.stringify(iceCandidate));
- //当本机当前页面的 RTCPeerConnection 接收到一个从远端页面通过信号通道发来的新的 ICE 候选地址信息,本机可以通过调用RTCPeerConnection.addIceCandidate() 来添加一个 ICE 代理
- connection.addIceCandidate(iceCandidate, function () {
- console.log(" addIceCandidate OK")
- },
- function (error) {
- console.log("addIceCandidate error:" + JSON.stringify(error))
- })
- }
- catch (err) {
- alert("connect ProcessRemoteIce error: " + err)
- }
- }
-
- var startPlay = function (url) {
- var inc = document.getElementById('incomming');
- var wsImpl = window.WebSocket || window.MozWebSocket;
- var form = document.getElementById('sendForm');
- var input = document.getElementById('sendText');
-
- inc.innerHTML += "connecting to server ..<br/>";
-
- window.video=document.getElementById('video1');
- window.ws = new wsImpl(url);
- //To receive the image as ArrayBuffer on the client side, you have to specify the binaryType after creating a WebSocket:
- ws.binaryType = "arraybuffer";//有了这个,就不用FileReader读取Blob的内容了。
- // when data is comming from the server, this metod is called
- ws.onmessage = function (evt) {
- console.log('>>>>> websocket.onmessage');
- //console.log('onmessage',evt.data);
- var dataObj = JSON.parse(evt.data);
- if(dataObj.type =='offer'){
- processRTCOffer(dataObj);//第一次
- }else if (dataObj.type=='remoteice'){
- processRemoteIce(dataObj);//后面3次
- }
- };
-
- window.isVideoPlaying=false;
-
- // when the connection is established, this method is called
- ws.onopen = function () {
- console.log('>>>>> websocket.onopen',video);
- inc.innerHTML += '.. connection open<br/>';
- ws.send(JSON.stringify({type: "open"})); //必须发送open指令,不然后续的都无法开始,onmessage将收不到offer
- };
-
- // when the connection is closed, this method is called
- ws.onclose = function () {
- inc.innerHTML += '.. connection closed<br/>';
- }
- }
- window.onload = function(){
- var url=$('#inputUrl').val();
- $('#btnPlay').click(function(){
- console.log('play',url);
- var inc = document.getElementById('incomming');
- inc.innerHTML="";
- startPlay(url);
- });
- }
- </script>
- </head>
- <body>
- <input id='inputUrl' value='ws://localhost:8085/api/v1/h5srtcapi?token=token1&profile=main&session=null' style='width:100%;'/>
- <button id='btnPlay'>play</button>
- <pre id="incomming"></pre>
- <video class="h5video1" id="video1" autoplay webkit-playsinline playsinline>
-
- </video>
- </body>
- </html>
其中在video.src=URL.createObjectURL(stream);卡了一下,原来需要引用adapter.js,不然不支持MediaStream类型的参数,导致提示,URL没有createObjectURL函数。
adapter.js里面有个
- URL.createObjectURL = function(stream) {
- if ('getTracks' in stream) {
- var url = 'polyblob:' + (++newId);
- streams.set(url, stream);
- utils.deprecated('URL.createObjectURL(stream)',
- 'elem.srcObject = stream');
- return url;
- }
- return nativeCreateObjectURL(stream);
- };
结果是src="polyblob:1"
------------------------------------------------------------------------
然后在video.play卡住了。
然后不知怎么,多刷新几次,可以了,有一定概率可以播放。
此时:chrome://webrtc-internals/
还是那些代码。
----------------------------------------------------------------------------------------
Uncaught (in promise) DOMException问题处理
参考1:Uncaught (in promise) DOMException谷歌浏览器js报错分析
输入chrome://flags/#autoplay-policy后,我的和它的结果不一样
参考2:解决Chrome浏览器无法自动播放音频视频的问题,Uncaught (in promise) DOMException
解决办法 1.静音 2.让用户手动点击
感觉现在的方式都是让用户点击吧
界面上加个按钮,而之所以我的代码会出错,h5s的代码没这个问题是它原本就是让人点击再播放的。
- window.onload = function(){
- var url=$('#inputUrl').val();
- $('#btnPlay').click(function(){
- console.log('play',url);
- var inc = document.getElementById('incomming');
- inc.innerHTML="";
- startPlay(url);
- });
- }
- </script>
- </head>
- <body>
- <input id='inputUrl' value='ws://localhost:8085/api/v1/h5srtcapi?token=token1&profile=main&session=null' style='width:100%;'/>
- <button id='btnPlay'>play</button>
- <pre id="incomming"></pre>
- <video class="h5video1" id="video1" autoplay webkit-playsinline playsinline>
-
- </video>
- </body>
- </html>
------------------------------------------------------------------------------------
总之就是通过几次请求(offer)、应答(answer),建立的webrtc连接。
最后wireshark里面发现有RTP协议的包
就是不知道是发给浏览器的呢,还是发给H5Stream的呢。
-----------------------------------------------------------------------------------
整理一下代码的过程,
1.通过websocket连接一个ws地址
- window.ws = new wsImpl('ws://localhost:8085/api/v1/h5srtcapi?token=token1&profile=main&session=null');
- //To receive the image as ArrayBuffer on the client side, you have to specify the binaryType after creating a WebSocket:
- ws.binaryType = "arraybuffer";//有了这个,就不用FileReader读取Blob的内容了。
2.在onopen里面发送open消息
- ws.onopen = function () {
- console.log('>>>>> websocket.onopen',video);
- inc.innerHTML += '.. connection open<br/>';
- ws.send(JSON.stringify({type: "open"})); //必须发送open指令,不然后续的都无法开始,onmessage将收不到offer
- };
3.在onmessage中处理收到的消息,建立webrtc通道
- ws.onmessage = function (evt) {
- console.log('>>>>> websocket.onmessage');
- //console.log('onmessage',evt.data);
- var dataObj = JSON.parse(evt.data);
- if(dataObj.type =='offer'){
- processRTCOffer(dataObj);//第一次
- }else if (dataObj.type=='remoteice'){
- processRemoteIce(dataObj);//后面3次
- }
- };
3.1 第一次返回的类型是offer,处理
创建RTCPeerConnection,
3.1.1.根据offer,setRemoteDescription,触发onaddstream,用mediastream设置video.src。
3.1.2.createAnswer,根据新的answer设置setLocalDescription,ws发送answer
3.1.3.触发onicecandidate,ws发送candidate(发现不发送也可以)
3.2 后面有3次返回remoteice,处理
3.2.1 RTCPeerConnection添加addIceCandidate
整个过程的事件在chrome://webrtc-internals/里面其实就列出来了。
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我的代码在FireFox中无法使用,H5Sream的可以。而IE则都不行。
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