如何使用webbrtc和基于服务器的Peer连接录制摄像头和音频

  <video autoplay></video>
<script language="javascript" type="text/javascript">
function onVideoFail(e) {
    console.log('webcam fail!', e);
  };
function hasGetUserMedia() {
  // Note: Opera is unprefixed.
  return !!(navigator.getUserMedia || navigator.webkitGetUserMedia ||
            navigator.mozGetUserMedia || navigator.msGetUserMedia);
}
if (hasGetUserMedia()) {
  // Good to go!
} else {
  alert('getUserMedia() is not supported in your browser');
}
window.URL = window.URL || window.webkitURL;
navigator.getUserMedia  = navigator.getUserMedia || navigator.webkitGetUserMedia ||
                          navigator.mozGetUserMedia || navigator.msGetUserMedia;
var video = document.querySelector('video');
var streamRecorder;
var webcamstream;
if (navigator.getUserMedia) {
  navigator.getUserMedia({audio: true, video: true}, function(stream) {
    video.src = window.URL.createObjectURL(stream);
    webcamstream = stream;
//  streamrecorder = webcamstream.record();
  }, onVideoFail);
} else {
    alert ('failed');
}
function startRecording() {
    streamRecorder = webcamstream.record();
    setTimeout(stopRecording, 10000);
}
function stopRecording() {
    streamRecorder.getRecordedData(postVideoToServer);
}
function postVideoToServer(videoblob) {
/*  var x = new XMLHttpRequest();
    x.open('POST', 'uploadMessage');
    x.send(videoblob);
*/
    var data = {};
    data.video = videoblob;
    data.metadata = 'test metadata';
    data.action = "upload_video";
    jQuery.post("http://www.foundthru.co.uk/uploadvideo.php", data, onUploadSuccess);
}
function onUploadSuccess() {
    alert ('video uploaded');
}
</script>
<div id="webcamcontrols">
    <a class="recordbutton" href="javascript:startRecording();">RECORD</a>
</div>

你一定要去看看Kurento。它提供了一个webbrtc服务器基础设施，允许您从webbrtc馈送和更多的记录。您还可以在这里找到您正在规划的应用程序的一些示例。在演示中添加录制功能并将媒体文件存储在URI(本地磁盘或其他地方)中非常容易。

该项目在LGPL Apache 2.0

下获得许可

---

编辑1

自从这篇文章以来，我们增加了一个新的教程，展示了如何在几个场景中添加记录器

• kurento-hello-world-recording:简单的录音教程，展示了录音端点的不同功能。
• kurento- one21 -recording:如何在媒体服务器上录制一对一的通信。
• kurento-hello-world-repository:使用外部存储库记录文件。

免责声明:我是Kurento开发团队的一员。

我认为使用kurento或其他mcu只是为了录制视频会有点过分，特别是考虑到Chrome从v47和Firefox v25开始就有MediaRecorder API支持的事实。所以在这个交叉点，你可能甚至不需要一个外部的js库来做这项工作，试试我做的这个演示，用MediaRecorder录制视频/音频:

演示 -将在chrome和firefox中工作(故意省略将blob推送到服务器代码)

Github Code Source

如果运行firefox，你可以在这里测试它(chrome需要https):

'use strict'
let log = console.log.bind(console),
  id = val => document.getElementById(val),
  ul = id('ul'),
  gUMbtn = id('gUMbtn'),
  start = id('start'),
  stop = id('stop'),
  stream,
  recorder,
  counter = 1,
  chunks,
  media;
gUMbtn.onclick = e => {
  let mv = id('mediaVideo'),
    mediaOptions = {
      video: {
        tag: 'video',
        type: 'video/webm',
        ext: '.mp4',
        gUM: {
          video: true,
          audio: true
        }
      },
      audio: {
        tag: 'audio',
        type: 'audio/ogg',
        ext: '.ogg',
        gUM: {
          audio: true
        }
      }
    };
  media = mv.checked ? mediaOptions.video : mediaOptions.audio;
  navigator.mediaDevices.getUserMedia(media.gUM).then(_stream => {
    stream = _stream;
    id('gUMArea').style.display = 'none';
    id('btns').style.display = 'inherit';
    start.removeAttribute('disabled');
    recorder = new MediaRecorder(stream);
    recorder.ondataavailable = e => {
      chunks.push(e.data);
      if (recorder.state == 'inactive') makeLink();
    };
    log('got media successfully');
  }).catch(log);
}
start.onclick = e => {
  start.disabled = true;
  stop.removeAttribute('disabled');
  chunks = [];
  recorder.start();
}
stop.onclick = e => {
  stop.disabled = true;
  recorder.stop();
  start.removeAttribute('disabled');
}
function makeLink() {
  let blob = new Blob(chunks, {
      type: media.type
    }),
    url = URL.createObjectURL(blob),
    li = document.createElement('li'),
    mt = document.createElement(media.tag),
    hf = document.createElement('a');
  mt.controls = true;
  mt.src = url;
  hf.href = url;
  hf.download = `${counter++}${media.ext}`;
  hf.innerHTML = `donwload ${hf.download}`;
  li.appendChild(mt);
  li.appendChild(hf);
  ul.appendChild(li);
}

      button {
        margin: 10px 5px;
      }
      li {
        margin: 10px;
      }
      body {
        width: 90%;
        max-width: 960px;
        margin: 0px auto;
      }
      #btns {
        display: none;
      }
      h1 {
        margin-bottom: 100px;
      }

<link type="text/css" rel="stylesheet" href="https://maxcdn.bootstrapcdn.com/bootstrap/3.3.6/css/bootstrap.min.css">
<h1> MediaRecorder API example</h1>
<p>For now it is supported only in Firefox(v25+) and Chrome(v47+)</p>
<div id='gUMArea'>
  <div>
    Record:
    <input type="radio" name="media" value="video" checked id='mediaVideo'>Video
    <input type="radio" name="media" value="audio">audio
  </div>
  <button class="btn btn-default" id='gUMbtn'>Request Stream</button>
</div>
<div id='btns'>
  <button class="btn btn-default" id='start'>Start</button>
  <button class="btn btn-default" id='stop'>Stop</button>
</div>
<div>
  <ul class="list-unstyled" id='ul'></ul>
</div>
<script src="https://code.jquery.com/jquery-2.2.0.min.js"></script>
<script src="https://maxcdn.bootstrapcdn.com/bootstrap/3.3.6/js/bootstrap.min.js"></script>

请检查RecordRTC

RecordRTC是MIT在github上授权的。

是的，如您所知，MediaStreamRecorder目前未实现。

MediaStreamRecorder是一个用于记录getUserMedia()流的WebRTC API。它允许web应用程序从实时音频/视频会话中创建文件。

或者你可以这样做http://ericbidelman.tumblr.com/post/31486670538/creating-webm-video-from-getusermedia但音频是缺失的一部分。

您可以使用RecordRTC-together，它是基于RecordRTC的。

它支持在单独的文件中录制视频和音频。您将需要像ffmpeg这样的工具将两个文件合并为服务器上的一个文件。

Web Call Server 4可以录制webbrtc的音视频到WebM容器。录音是使用Vorbis编解码器的音频和VP8编解码器的视频。最初的WebRTC编解码器是Opus或G.711和VP8。因此，服务器端记录需要Opus/G。711到Vorbis服务器端转码或VP8-H。264转码，如果有必要使用另一个容器，即AVI。

我也没有足够的知识，

但是我在Git hub上找到了这个-

<!DOCTYPE html>
 <html>
<head>
  <title>XSockets.WebRTC Client example</title>
  <meta charset="utf-8" />

<style>
body {
  }
.localvideo {
position: absolute;
right: 10px;
top: 10px;
}
.localvideo video {
max-width: 240px;
width:100%;
margin-right:auto;
margin-left:auto;
border: 2px solid #333;
 }
 .remotevideos {
height:120px;
background:#dadada;
padding:10px; 
}
.remotevideos video{
max-height:120px;
float:left;
 }
</style>
</head>
<body>
<h1>XSockets.WebRTC Client example </h1>
<div class="localvideo">
    <video autoplay></video>
</div>
<h2>Remote videos</h2>
<div class="remotevideos">
</div>
<h2>Recordings  ( Click on your camera stream to start record)</h2>
<ul></ul>

<h2>Trace</h2>
<div id="immediate"></div>
<script src="XSockets.latest.js"></script>
<script src="adapter.js"></script>
<script src="bobBinder.js"></script>
<script src="xsocketWebRTC.js"></script>
<script>
    var $ = function (selector, el) {
        if (!el) el = document;
        return el.querySelector(selector);
    }
    var trace = function (what, obj) {
        var pre = document.createElement("pre");
        pre.textContent = JSON.stringify(what) + " - " + JSON.stringify(obj || "");
        $("#immediate").appendChild(pre);
    };
    var main = (function () {
        var broker;
        var rtc;
        trace("Ready");
        trace("Try connect the connectionBroker");
        var ws = new XSockets.WebSocket("wss://rtcplaygrouund.azurewebsites.net:443", ["connectionbroker"], {
            ctx: '23fbc61c-541a-4c0d-b46e-1a1f6473720a'
        });
        var onError = function (err) {
            trace("error", arguments);
        };
        var recordMediaStream = function (stream) {
            if ("MediaRecorder" in window === false) {
                trace("Recorder not started MediaRecorder not available in this browser. ");
                return;
            }
            var recorder = new XSockets.MediaRecorder(stream);
            recorder.start();
            trace("Recorder started.. ");
            recorder.oncompleted = function (blob, blobUrl) {
                trace("Recorder completed.. ");
                var li = document.createElement("li");
                var download = document.createElement("a");
                download.textContent = new Date();
                download.setAttribute("download", XSockets.Utils.randomString(8) + ".webm");
                download.setAttribute("href", blobUrl);
                li.appendChild(download);
                $("ul").appendChild(li);
            };
        };
        var addRemoteVideo = function (peerId, mediaStream) {
            var remoteVideo = document.createElement("video");
            remoteVideo.setAttribute("autoplay", "autoplay");
            remoteVideo.setAttribute("rel", peerId);
            attachMediaStream(remoteVideo, mediaStream);
            $(".remotevideos").appendChild(remoteVideo);
        };
        var onConnectionLost = function (remotePeer) {
            trace("onconnectionlost", arguments);
            var peerId = remotePeer.PeerId;
            var videoToRemove = $("video[rel='" + peerId + "']");
            $(".remotevideos").removeChild(videoToRemove);
        };
        var oncConnectionCreated = function () {
            console.log(arguments, rtc);
            trace("oncconnectioncreated", arguments);
        };
        var onGetUerMedia = function (stream) {
            trace("Successfully got some userMedia , hopefully a goat will appear..");
            rtc.connectToContext(); // connect to the current context?
        };
        var onRemoteStream = function (remotePeer) {
            addRemoteVideo(remotePeer.PeerId, remotePeer.stream);
            trace("Opps, we got a remote stream. lets see if its a goat..");
        };
        var onLocalStream = function (mediaStream) {
            trace("Got a localStream", mediaStream.id);
            attachMediaStream($(".localvideo video "), mediaStream);
            // if user click, video , call the recorder
            $(".localvideo video ").addEventListener("click", function () {
                recordMediaStream(rtc.getLocalStreams()[0]);
            });
        };
        var onContextCreated = function (ctx) {
            trace("RTC object created, and a context is created - ", ctx);
            rtc.getUserMedia(rtc.userMediaConstraints.hd(false), onGetUerMedia, onError);
        };
        var onOpen = function () {
            trace("Connected to the brokerController - 'connectionBroker'");
            rtc = new XSockets.WebRTC(this);
            rtc.onlocalstream = onLocalStream;
            rtc.oncontextcreated = onContextCreated;
            rtc.onconnectioncreated = oncConnectionCreated;
            rtc.onconnectionlost = onConnectionLost;
            rtc.onremotestream = onRemoteStream;
            rtc.onanswer = function (event) {
            };
            rtc.onoffer = function (event) {
            };
        };
        var onConnected = function () {
            trace("connection to the 'broker' server is established");
            trace("Try get the broker controller form server..");
            broker = ws.controller("connectionbroker");
            broker.onopen = onOpen;
        };
        ws.onconnected = onConnected;
    });
    document.addEventListener("DOMContentLoaded", main);
</script>

在第89行，在我的情况下代码OnrecordComplete实际上附加了一个记录器文件的链接，如果你点击那个链接，它将开始下载，你可以将该路径保存到你的服务器作为一个文件。

记录代码看起来像这样

recorder.oncompleted = function (blob, blobUrl) {
                trace("Recorder completed.. ");
                var li = document.createElement("li");
                var download = document.createElement("a");
                download.textContent = new Date();
                download.setAttribute("download", XSockets.Utils.randomString(8) + ".webm");
                download.setAttribute("href", blobUrl);
                li.appendChild(download);
                $("ul").appendChild(li);
            };

blobUrl保存路径。我用这个解决了我的问题，希望有人会发现这个有用

目前浏览器支持客户端录音

https://webrtc.github.io/samples/

可以在连接结束后通过一些HTTP请求上传将录制的文件推送到服务器。

https://webrtc.github.io/samples/src/content/getusermedia/record/https://github.com/webrtc/samples/tree/gh-pages/src/content/getusermedia/record

这有一些缺点，如果用户只是关闭选项卡，不在后端运行这些操作，它可能不会将文件完全上传到服务器。

作为一个更稳定的解决方案，Ant Media Server可以在服务器端记录流，记录功能是Ant Media Server的基本功能之一。

antmedia.io

注:我是蚂蚁媒体团队成员。

从技术上讲，您可以在后端使用FFMPEG来混合视频和音频