网页中播放RTSP(6) WebRTC播放视频

最新推荐文章于 2024-06-14 15:55:52 发布

llhswwha

最新推荐文章于 2024-06-14 15:55:52 发布

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分类专栏： WebRTC

本文链接：https://blog.csdn.net/llhswwha/article/details/102547526

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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。

--------------------------------------------------------------------

我的代码在FireFox中无法使用，H5Sream的可以。而IE则都不行。

llhswwha

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网页中播放RTSP(6) WebRTC播放视频

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 ...
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