w3.org WebRTC 1.0

https://www.w3.org/TR/webrtc/

1. Introduction
– Connecting to remote peers using NAT-traversal technologies such as ICE, STUN, and TURN
2. Conformance
3. Terminology
– The EventHandler interface, representing a callback used for event handlers, and the ErrorEvent interface are defined in HTML
4.Peer-to-peer connections
4.1.Introduction
– Communications are coordinated by the exchange of control messages (called a signaling protocol) over a signaling channel which is provided by unspecified means, but generally by a script in the page via the server, e.g. using XMLHttpRequest [xhr] or Web Sockets
4.2.Configuration
4.2.1 RTCConfiguration Dictionary
 4.2.2 RTCIceCredentialType Enum
4.2.3 RTCIceServer Dictionary

  {urls: 'stun:stun1.example.net'},
  {urls: ['turns:turn.example.org', 'turn:turn.example.net'],
    username: 'user',
    credential: 'myPassword',
    credentialType: 'password'},

4.2.4 RTCIceTransportPolicy Enum
4.2.5 RTCBundlePolicy Enum
4.2.6 RTCRtcpMuxPolicy Enum
4.2.7 Offer/Answer Options
4.3 State Definitions
4.3.1 RTCSignalingState Enum
Non-normative signalling state transitions diagram
– Caller transition:
new RTCPeerConnection(): stable
setLocalDescription(offer): have-local-offer
setRemoteDescription(pranswer): have-remote-pranswer
setRemoteDescription(answer): stable
– Callee transition:
new RTCPeerConnection(): stable
setRemoteDescription(offer): have-remote-offer
setLocalDescription(pranswer): have-local-pranswer
setLocalDescription(answer): stable
4.3.2 RTCIceGatheringState Enum
4.3.3 RTCPeerConnectionState Enum
4.3.4 RTCPeerConnectionState Enum
4.4 RTCPeerConnection Interface
– An RTCPeerConnection object has a signaling state, a connection state, an ICE gathering state, and an ICE connection state
4.4.1.1 Constructor
4.4.1.2 Chain an asynchronous operation
-> operations chainにアルゴリズムが全て書いてある
4.4.1.3 Update the connection state
4.4.1.4 Update the ICE gathering state
4.4.1.5 Set the RTCSessionDescription
4.4.1.6 Set the configuration
4.4.2 Interface Definition
– createOffer, createAnswer, setLocalDescription, setRemoteDescription, addIceCandidate, restartIce, getConfiguration, setConfiguration, close
4.4.3 Legacy Interface Extensions
4.4.3.1 Method extensions
4.4.3.2 Legacy configuration extensions
4.4.4 Garbage collection
4.5 Error Handling
4.6 Session Description Model
4.6.1 RTCSdpType
– offer, pranswer, answer, rollback
4.6.2 RTCSessionDescription Class
4.7 RTCSessionDescription Class
– This event is fired according to the state of the connection’s negotiation-needed flag
4.7.1 Setting Negotiation-Needed
4.7.2 Setting Negotiation-Needed
4.7.3 Updating the Negotiation-Needed flag
4.8 Interfaces for Connectivity Establishment
4.8.1 RTCIceCandidate Interface
4.8.1.1 candidate-attribute Grammar
4.8.1.2 RTCIceProtocol Enum
4.8.1.3 RTCIceTcpCandidateType Enum
4.8.1.4 RTCIceCandidateType
4.82 RTCPeerConnectionIceEvent
4.8.3 RTCPeerConnectionIceErrorEvent
4.9 Certificate Management
4.9.1 RTCCertificateExpiration Dictionary
4.9.2 RTCCertificate Interface
5. RTP Media API
5.1 RTCPeerConnection Interface Extensions
5.1.1 Processing Remote MediaStreamTracks
5.2 RTCRtpSender Interface
5.2.1 RTCRtpParameters
5.2.2 RTCRtpSendParameters
5.2.3 RTCRtpReceiveParameters
5.2.4 RTCRtpCodingParameters
5.2.5 RTCRtpDecodingParameters
5.2.6 RTCRtpEncodingParameters
5.2.7 RTCRtcpParameters
5.2.8 RTCRtpHeaderExtensionParameters
5.2.9 RTCRtpCodecParameters
5.2.10 RTCRtpCapabilities
5.2.11 RTCRtpCodecCapability
5.2.12 RTCRtpHeaderExtensionCapability
5.3 RTCRtpReceiver
5.4 RTCRtpTransceiver
5.5 RTCDtlsTransport
5.5.1 RTCDtlsFingerprint
5.6 RTCIceTransport
5.6.1 RTCIceParameters
5.6.2 RTCIceCandidatePair
5.6.3 RTCIceGathererState
5.6.4 RTCIceTransportState
5.6.5 RTCIceRole
5.6.6 RTCIceComponent
5.7 RTCTrackEvent
6.Peer-to-peer Data API
6.1 RTCPeerConnection Interface Extensions
6.1.1 RTCSctpTransport
6.1.1.1 Create an instance
6.1.1.2 Update max message size
6.1.1.3 Connected procedure
6.2 RTCDataChannel
6.2.1 Creating a data channel
6.2.2 Announcing a data channel as open
6.2.3 Announcing a data channel instance
6.2.4 Closing procedure
6.2.5 Announcing a data channel as closed
6.2.6 Error on creating data channels
6.2.7 Receiving messages on a data channel
6.3 RTCDataChannelEvent
6.4 Garbage Collection
7. Peer-to-peer DTMF
7.1 RTCRtpSender Interface Extensions
7.2 RTCDTMFSender
7.3 canInsertDTMF algorithm
7.4 RTCDTMFToneChangeEvent
8. Statistics Model
8.1 Introduction
8.2 RTCPeerConnection Interface Extensions
8.3 RTCStatsReport
8.4 RTCStats
8.5 The stats selection algorithm
8.6 Mandatory To Implement Stats
9. Media Stream API Extensions for Network Use
9.2 MediaStream
9.2.1 id
9.3 MediaStreamTrack
9.3.1 MediaTrackSupportedConstraints, MediaTrackCapabilities, MediaTrackConstraints and MediaTrackSettings
10. Examples and Call Flows
10.1 Simple Peer-to-peer Example

const signaling = new SignalingChannel(); // handles JSON.stringify/parse
const constraints = {audio: true, video: true};
const configuration = {iceServers: [{urls: 'stun:stun.example.org'}]};
const pc = new RTCPeerConnection(configuration);

// send any ice candidates to the other peer
pc.onicecandidate = ({candidate}) => signaling.send({candidate});

// let the "negotiationneeded" event trigger offer generation
pc.onnegotiationneeded = async () => {
  try {
    await pc.setLocalDescription(await pc.createOffer());
    // send the offer to the other peer
    signaling.send({desc: pc.localDescription});
  } catch (err) {
    console.error(err);
  }
};

// once media for a remote track arrives, show it in the remote video element
pc.ontrack = (event) => {
  // don't set srcObject again if it is already set.
  if (remoteView.srcObject) return;
  remoteView.srcObject = event.streams[0];
};

// call start() to initiate
async function start() {
  try {
    // get a local stream, show it in a self-view and add it to be sent
    const stream = await navigator.mediaDevices.getUserMedia(constraints);
    stream.getTracks().forEach((track) => pc.addTrack(track, stream));
    selfView.srcObject = stream;
  } catch (err) {
    console.error(err);
  }
}

signaling.onmessage = async ({desc, candidate}) => {
  try {
    if (desc) {
      // if we get an offer, we need to reply with an answer
      if (desc.type == 'offer') {
        await pc.setRemoteDescription(desc);
        const stream = await navigator.mediaDevices.getUserMedia(constraints);
        stream.getTracks().forEach((track) => pc.addTrack(track, stream));
        await pc.setLocalDescription(await pc.createAnswer());
        signaling.send({desc: pc.localDescription});
      } else if (desc.type == 'answer') {
        await pc.setRemoteDescription(desc);
      } else {
        console.log('Unsupported SDP type. Your code may differ here.');
      }
    } else if (candidate) {
      await pc.addIceCandidate(candidate);
    }
  } catch (err) {
    console.error(err);
  }
};

10.2 Advanced Peer-to-peer Example with Warm-up

const signaling = new SignalingChannel();
const configuration = {iceServers: [{urls: 'stun:stun.example.org'}]};
const audio = null;
const audioSendTrack = null;
const video = null;
const videoSendTrack = null;
const started = false;
let pc;

// Call warmup() to warm-up ICE, DTLS, and media, but not send media yet.
async function warmup(isAnswerer) {
  pc = new RTCPeerConnection(configuration);
  if (!isAnswerer) {
    audio = pc.addTransceiver('audio');
    video = pc.addTransceiver('video');
  }

  // send any ice candidates to the other peer
  pc.onicecandidate = (event) => {
    signaling.send(JSON.stringify({candidate: event.candidate}));
  };

  // let the "negotiationneeded" event trigger offer generation
  pc.onnegotiationneeded = async () => {
    try {
      await pc.setLocalDescription(await pc.createOffer());
      // send the offer to the other peer
      signaling.send(JSON.stringify({desc: pc.localDescription}));
    } catch (err) {
      console.error(err);
    }
  };

  // once media for the remote track arrives, show it in the remote video element
  pc.ontrack = async (event) => {
    try {
      if (event.track.kind == 'audio') {
        if (isAnswerer) {
          audio = event.transceiver;
          audio.direction = 'sendrecv';
          if (started && audioSendTrack) {
            await audio.sender.replaceTrack(audioSendTrack);
          }
        }
      } else if (event.track.kind == 'video') {
        if (isAnswerer) {
          video = event.transceiver;
          video.direction = 'sendrecv';
          if (started && videoSendTrack) {
            await video.sender.replaceTrack(videoSendTrack);
          }
        }
      }

      // don't set srcObject again if it is already set.
      if (!remoteView.srcObject) {
        remoteView.srcObject = new MediaStream();
      }
      remoteView.srcObject.addTrack(event.track);
    } catch (err) {
      console.error(err);
    }
  };

  try {
    // get a local stream, show it in a self-view and add it to be sent
    const stream = await navigator.mediaDevices.getUserMedia({audio: true,
                                                              video: true});
    selfView.srcObject = stream;
    audioSendTrack = stream.getAudioTracks()[0];
    if (started) {
      await audio.sender.replaceTrack(audioSendTrack);
    }
    videoSendTrack = stream.getVideoTracks()[0];
    if (started) {
      await video.sender.replaceTrack(videoSendTrack);
    }
  } catch (err) {
    console.error(err);
  }
}

// Call start() to start sending media.
function start() {
  started = true;
  signaling.send(JSON.stringify({start: true}));
}

signaling.onmessage = async (event) => {
  if (!pc) warmup(true);

  try {
    const message = JSON.parse(event.data);
    if (message.desc) {
      const desc = message.desc;

      // if we get an offer, we need to reply with an answer
      if (desc.type == 'offer') {
        await pc.setRemoteDescription(desc);
        await pc.setLocalDescription(await pc.createAnswer());
        signaling.send(JSON.stringify({desc: pc.localDescription}));
      } else {
        await pc.setRemoteDescription(desc);
      }
    } else if (message.start) {
      started = true;
      if (audio && audioSendTrack) {
        await audio.sender.replaceTrack(audioSendTrack);
      }
      if (video && videoSendTrack) {
        await video.sender.replaceTrack(videoSendTrack);
      }
    } else {
      await pc.addIceCandidate(message.candidate);
    }
  } catch (err) {
    console.error(err);
  }
};

10.3 Simulcast Example
10.4 Peer-to-peer Data Example
10.5 Call Flow Browser to Browser

10.6 DTMF Example
11. Error Handling

-> ICE CandidateはIPアドレス、プロトコル(TCP/UDP)だけでなく、ポート番号、コンポーネント、タイプ(host/srlfx/relay)、優先度(type preference, local preference, component IDなど)、ファウンデーション(ホールパンチ効率化)、ベース(無駄な候補を省く)も含まれる
-> 収集(IP&Portを集める)、交換(候補を交換)、整頓(相手の候補と自分の候補をペアにしてuniq & sort)、穴開(ペアに対して接続試行・ホールパンチ)、集結(候補を決定)
-> ローカル候補を取得(host)、STUNでNAT外部候補を取得(srlfx)、TurnにAllocation Request(relay)

express.ioによるchat room

### server.js

var express = require('express.io');
var app = express();
var PORT = 3000;

var fs = require("fs");
var https = require("https");
var options = {
	key: fs.readFileSync('key.pem'),
	cert: fs.readFileSync('server.crt')
}
app.https(options).io();

console.log('server started' + PORT);

app.use(express.static(__dirname + '/public'));
app.get('/', function(req, res){
	res.render('index.ejs');
});

app.io.route('ready', function(req){
	req.io.join(req.data)
	app.io.room(req.data).broadcast('announce', {
		message: 'New client in the ' + req.data + ' room.'
	})
})

app.io.route('send', function(req){
	app.io.room(req.data.room).broadcast('message', {
		message: req.data.message,
		author: req.data.author,
	})
})

app.listen(PORT);

### index.ejs

<!DOCTYPE html>
<html lang="en">
<head>
	<meta charset="UTF-8">
	<title>WebRTC</title>
	<link rel="stylesheet" type="text/css" href="public/styles.css">
	<script src="/socket.io/socket.io.js"></script>
</head>
<body>
	<p><button id="takeProfilePicture" type="button" autofocus="true">Create Profile Picture</button></p>
	<video id="videoTag" autoplay></video>
	<div>
		<label>Your Name</label><input id="myName" type="text">
		<label>Message</label><input id="myMessage" type="text">
		<input id="sendMessage" type="submit">
		<div id="chatArea">Message: output:<br></div>
	</div>

	<script>
		navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || window.navigator.mozGetUserMedia;
		var constraints = {audio: false, video: {
				mandatory: {
					maxWidth: 240,
					maxHeight: 240
				}
			}
		};
		var videoArea = document.querySelector("video");
		var myName = document.querySelector("#myName");
		var myMessage = document.querySelector("#myMessage");
		var sendMessage = document.querySelector("#sendMessage");
		var chatArea = document.querySelector("#chatArea");
		var ROOM = "chat";

		io = io.connect();
		io.emit('ready', ROOM);

		io.on('announce', function(data){
			displayMessage(data.message);
		});

		io.on('message', function(data){
			displayMessage(data.author + ": " + data.message);
		});

		sendMessage.addEventListener('click', function(ev){
			io.emit('send', {"author":myName.value, "message":myMessage.value, "room":ROOM});
			ev.preventDefault();
		}, false);

		function displayMessage(message){
			chatArea.innerHTML = chatArea.innerHTML + "<br>" + message;
		}

		navigator.getUserMedia(constraints, onSuccess, onError);

		function onSuccess(stream){
			console.log("Success! we have a stream!");
			// videoArea.src = window.URL.createObjectURL(stream);
			// videoArea.className = "grayscale_filter";
			videoArea.srcObject = stream;
		}

		function onError(error){
			console.log("Error with getUserMedia: ", error);
		}
	</script>
</body>
</html>

chat room top -> chat room create -> insert into mysql -> chat room display -> other user join って流れか
socket ioは、HTTP通信ではなく、WebSocketによって通信を行なっている

あれ、待てよ、webrtcだと、SDPの送信が必要なわけだけど、Socket.ioは、TURN serverは提供していないわけだから、TURNで取得したpublic ipをsocket.ioで転送するってこと?

Express.ioを使ってみる

express.io = express + socket.io

express.ioをインストールして、サーバーを立てるまで

### express.io install
package.json

{
	"name": "test-webrtc",
	"version": "0.0.1",
	"private": true,
	"dependencies": {
		"express": "4.x",
		"ejs": "3.0.1",
		"express.io": "1.x",
		"coffee-script": "~1.6.3",
		"connect": "*"
	}
}

$ npm update

### server.js

var express = require('express.io');
var app = express();
var PORT = 3000;

var fs = require("fs");
var https = require("https");
var options = {
	key: fs.readFileSync('key.pem'),
	cert: fs.readFileSync('server.crt')
}
app.https(options).io();

console.log('server started' + PORT);

app.use(express.static(__dirname + '/public'));
app.get('/', function(req, res){
	res.render('index.ejs');
});

app.io.route('ready', function(req){
	req.io.join(req.data)
	app.io.room(req.data).broadcast('announce', {
		message: 'New client in the ' + req.data + ' room.'
	})
})

app.listen(PORT);

### index.ejs

<!DOCTYPE html>
<html lang="en">
<head>
	<meta charset="UTF-8">
	<title>WebRTC</title>
	<link rel="stylesheet" type="text/css" href="public/styles.css">
	<script src="/socket.io/socket.io.js"></script>
</head>
<body>
	<p><button id="takeProfilePicture" type="button" autofocus="true">Create Profile Picture</button></p>
	<video id="videoTag" autoplay></video>
	<div>
		<label>Your Name</label><input id="myName" type="text">
		<label>Your Name</label><input id="myMessage" type="text">
		<input id="sendMessage" type="submit">
		<div id="chatArea">Message: output:<br></div>
	</div>

	<script>
		navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || window.navigator.mozGetUserMedia;
		var constraints = {audio: false, video: {
				mandatory: {
					maxWidth: 240,
					maxHeight: 240
				}
			}
		};
		var videoArea = document.querySelector("video");
		var myName = document.querySelector("#myName");
		var myMessage = document.querySelector("#myMessage");
		var sendMessage = document.querySelector("#sendMessage");
		var chatArea = document.querySelector("#chatArea");
		var ROOM = "chat";

		io = io.connect();
		io.emit('ready', ROOM);

		io.on('announce', function(data){
			displayMessage(data.message);
		});

		function displayMessage(message){
			chatArea.innerHTML = chatArea.innerHTML + "<br>" + message;
		}

		navigator.getUserMedia(constraints, onSuccess, onError);

		function onSuccess(stream){
			console.log("Success! we have a stream!");
			// videoArea.src = window.URL.createObjectURL(stream);
			// videoArea.className = "grayscale_filter";
			videoArea.srcObject = stream;
		}

		function onError(error){
			console.log("Error with getUserMedia: ", error);
		}
	</script>
</body>
</html>

$npm server.js

socket ioとは、web soketにより、双方向通信を簡単に記述できる
複数ブラウザでテストし、io.emit(‘ready’, ROOM);となった際に、chatArea.innerHTML = chatArea.innerHTML + “
” + message;で、’New client in the ‘ + req.data + ‘ room.’が追加される

Vagrant環境(amazon linux2)で、Expressを使ってhttpsサーバーを立てる

まず、sslモジュールをinstall
$ sudo yum install mod_ssl

続いてkeyとcertを作成して読み込む
# 手順
## certificate file作成
openssl req -newkey rsa:2048 -new -nodes -keyout key.pem -out csr.pem
openssl x509 -req -days 365 -in csr.pem -signkey key.pem -out server.crt

## package.json

{
	"name": "test-webrtc",
	"version": "0.0.1",
	"private": true,
	"dependencies": {
		"express": "4.x",
		"ejs": "3.0.1"
	}
}

$ npm install

## server.js

var express = require('express');
var app = express();

var fs = require("fs");
var https = require("https");
var options = {
	key: fs.readFileSync('key.pem'),
	cert: fs.readFileSync('server.crt')
}
var server = https.createServer(options, app);

console.log('server started');

app.get('/', function(req, res){
	res.render('index.ejs');
});

server.listen(3000);

$ node server.js

# 駄目な方法
## certificate file作成
$ openssl genrsa > server.key
$ openssl req -new -key server.key > server.csr
$ openssl x509 -req -signkey server.key < server.csr > server.crt

var express = require('express');
var app = express();

var fs = require("fs");
var https = require("https");
var options = {
	key: fs.readFileSync('server.key'),
	cert: fs.readFileSync('server.crt')
}
var server = https.createServer(options, app);

console.log('server started');

app.get('/', function(req, res){
	res.writeHead(200);
	res.render('index.ejs');
});

server.listen(3000);

## server.js
keyがpemファイルでないので、エラーが出ます
$ node server.js
_tls_common.js:88
c.context.setCert(options.cert);
^

Error: error:0906D06C:PEM routines:PEM_read_bio:no start line
at Object.createSecureContext (_tls_common.js:88:17)
at Server (_tls_wrap.js:819:25)
at new Server (https.js:60:14)
at Object.createServer (https.js:82:10)
at Object. (/home/vagrant/webrtc/server.js:10:20)
at Module._compile (module.js:653:30)
at Object.Module._extensions..js (module.js:664:10)
at Module.load (module.js:566:32)
at tryModuleLoad (module.js:506:12)
at Function.Module._load (module.js:498:3)

vagrantでhttpsの環境を作ろうとした時、opensslとphpのビルトインサーバーでhttps環境を作っていましたが、フロントエンドだけならexpressで十分だということがわかりました。
expressはhttpのみかと勘違いしていたが、よくよく考えたら、できないわけない😂😂😂

Signalingの仕組み

## basic signaling structure

“Offer” & “Answer” : Session Description Protocol(SDP), video codecs resolution format

How to connect : Websockets, socket.io, Publish/Subscribe, commercial providers

To communicate within Firewall of Private Networks
1. Connection over Plubic IP’s
2. STUN server ※NATを通過するためのポートマッピング
3. TURN server ※Firewallを越えるための、TURNによるリレーサーバーを介した中継通信

STUN, TURNでSDPを交換してから、ICE(Internet Connectivity Establishment) Candidateで接続する

PCカメラでの自撮画像 作成方法

– そろそろプロフィール写真を変えたいが、スマホで撮って一々転送するのはめんどくさいので、自分のPCで自撮画像を撮りたいと思ってる方に朗報
– 以下のコードでPCから自撮して画像化できます
– canvasでvideoタグをdrawImageして、 toDataURL(‘image/png’) で画像化してダウンロードできるようにしています

<!DOCTYPE html>
<html lang="en">
<head>
	<meta charset="UTF-8">
	<title>WebRTC</title>
	<link rel="stylesheet" type="text/css" href="public/styles.css">
</head>
<body>
	<p><button id="takeProfilePicture" type="button" autofocus="true">Create Profile Picture</button></p>
	<video id="videoTag" autoplay></video>
	<canvas id="profilePicCanvas" style="display: none;"></canvas>
	<div>
		<img id="profilePictureOutput">
	</div>
	<div class="download" style="display: none;">
		<a id="download" href="#" download="canvas.jpg">download</a>
	</div>
	<script>
		navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || window.navigator.mozGetUserMedia;
		var constraints = {audio: false, video: {
				mandatory: {
					maxWidth: 240,
					maxHeight: 240
				}
			}
		};
		var videoArea = document.querySelector("video");
		var profilePicCanvas = document.querySelector("#profilePicCanvas");
		var profilePictureOutput = document.querySelector("#profilePictureOutput");
		var takePicButton = document.querySelector("#takeProfilePicture");
		var videoTag = document.querySelector("#videoTag");
		var width = 240;
		var height = 0;
		var streaming = false;

		takePicButton.addEventListener('click', function(ev){
			takeProfilePic();
			ev.preventDefault();
		}, false);

		videoTag.addEventListener('canplay', function(ev){
			if(!streaming){
				height = videoTag.videoHeight / (videoTag.videoWidth/width);
				if (isNaN(height)){
					height = width / (4/3);
				}
				videoTag.setAttribute('width', width);
				videoTag.setAttribute('height', height);
				profilePicCanvas.setAttribute('width', width);
				profilePicCanvas.setAttribute('height', height);
				streaming = true;
			}
		}, false);

		function takeProfilePic(){
			var context = profilePicCanvas.getContext('2d');
			if (width && height){
				profilePicCanvas.width = width;
				profilePicCanvas.height = height;
				context.drawImage(videoTag, 0, 0, width, height);

				var data = profilePicCanvas.toDataURL('image/png');
				profilePictureOutput.setAttribute('src', data);
				document.querySelector(".download").style.display = "block";
				document.getElementById("download").href = data;
			}
		}

		navigator.getUserMedia(constraints, onSuccess, onError);

		function onSuccess(stream){
			console.log("Success! we have a stream!");
			// videoArea.src = window.URL.createObjectURL(stream);
			// videoArea.className = "grayscale_filter";
			videoArea.srcObject = stream;
		}

		function onError(error){
			console.log("Error with getUserMedia: ", error);
		}
	</script>
</body>
</html>

今からハッカソンエントリーしてくる

webrtc 縦横比とCSSグレースケール

アプリケーションによって制約があるかと思うが、基本はwidth, heightはvideo constraintsに沿って4:3にする
cssのfilter: saturate(0.0x); でvideoをグレースケール化できる

WebRTC Video Resolutions 2 – the Constraints Fight Back


https://webrtchacks.com/

<script>
		navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || window.navigator.mozGetUserMedia;
		var constraints = {audio: false, video: {
				mandatory: {
					maxWidth: 640,
					maxHeight: 480
				}
			}
		};
		var videoArea = document.querySelector("video");
		navigator.getUserMedia(constraints, onSuccess, onError);

		function onSuccess(stream){
			console.log("Success! we have a stream!");
			// videoArea.src = window.URL.createObjectURL(stream);
                        videoArea.className = "grayscale_filter";
			videoArea.srcObject = stream;
		}

		function onError(error){
			console.log("Error with getUserMedia: ", error);
		}
	</script>
.grayscale_filter{
	-webkit-filter: saturate(0.02);
	filter: saturate(0.02);
}

$ vendor/bin/hyper-run -s 192.168.33.10:8000
whala

あれ、ちょっと待てよ、webrtcそのまま画像認識に使える👺 あれ?

coturn

firewallを超える場合に、turnサーバーでドメインを指定する

$ sudo cp /usr/local/etc/turnserver.conf.default /usr/local/etc/turnserver.conf
$ sudo vi /usr/local/etc/turnserver.conf

# The default realm to be used for the users when no explicit 
# origin/realm relationship was found in the database, or if the TURN
# server is not using any database (just the commands-line settings
# and the userdb file). Must be used with long-term credentials 
# mechanism or with TURN REST API.
#
#realm=mycompany.org
...

# Uncomment if no UDP client listener is desired.
# By default UDP client listener is always started.
#
#no-udp

# Uncomment if no TCP client listener is desired.
# By default TCP client listener is always started.
#
#no-tcp

# Uncomment if no TLS client listener is desired.
# By default TLS client listener is always started.
#
#no-tls

# Uncomment if no DTLS client listener is desired.
# By default DTLS client listener is always started.
#
#no-dtls

ってことはSTUNは外部で調達して、TURNはインスタンスで立てるって理解でOK?

NATとSTUN・TURNサーバー

WebRTC通信
– Peer to Peer、ブラウザ間で直接通信
– UDP/IPを使用、オーバーヘッドが少ない
– 鍵交換で暗号化通信を行う
-> 相手のIPアドレス、動的なUDPポート番号を知る必要がある
-> 通信経路の仕組みがInteractive Connectivity Establishment、その候補がICE Candidate
–> NATを通過するためのSTUNサーバから取得したポートマッピング
–> Firewallを越えるための、TURNによるリレーサーバーを介した中継通信
–> このやりとりをシグナリングと言う。WebSocketなど複数の方法がある
-> 複数人通信の場合には、それぞれのユーザとSDP/IPのconnectionをつくる必要がある

NATとは
 グローバルIPとローカルのネットワークIPとの変換
 複数のPC/デバイスが同時に通信できるよう、ポートマッピングによるポート変換
 →ブラウザはローカルのIP、UDPポートはわかるが、グローバルのIP、UDPはわからない
 →→Peer to Peerはグローバルの情報を交換する必要がある

STUN(Session Traversal Utilities for NATs)
NATで変換されたIP/UDPを外のSTUNサーバーから教えてもらう
→グローバル情報をシグナリングサーバーけいゆうで相手に渡す
STUNサーバーはGoogleのstun.l.google.com:19302など

TURN(Traversal Using Relays around NAT)
ストリームデータの受け渡しにリレーする
TURNサーバが入ると厳密にはPeer to Peerではなくなる
データのデコード、エンコードは行わないので、ネットワーク負荷が高くなる

あれ、Bitcoinって、P2Pだけど、STAN, TURN使ってるんだっけ?
否、BitcoinはTCP😂

Vagrant AmazonLinuxでWebRTC

まず、WebRTCとは?
-WebReal-Time Communicationsの略
-ウェブでシンプルなAPI経由でリアルタイム通信を提供する
-P2P通信
-オープンソース

アーキテクチャ

セッションでやりとりしてるんか。。

-getUserMedia
ブラウザから端末に取り付けられているカメラやマイクにアクセスしてストリームデータを取得
-RTCPeerConnection
マルチメディアセッションを確立するAPI
-RTCDataChannel
テキストデータ、バイナリデータのP2Pデータ通信API

vagrant でvideoのテスト

<!doctype html>
<html>
<head>
 <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
 <title>Wrap old and new getUserMedia</title>
</head>
<body>
  Wrap old and new getUserMedia<br />
  <button type="button" onclick="startVideo();">Start</button>
  <button type="button" onclick="stopVideo();">Stop</button>
  <br />
  <video id="local_video" autoplay style="width: 320px; height: 240px; border: 1px solid black;"></video>
</body>
<script type="text/javascript">
  var localVideo = document.getElementById('local_video');
  var localStream = null;

  // --- prefix -----
  navigator.getUserMedia  = navigator.getUserMedia    || navigator.webkitGetUserMedia ||
                            navigator.mozGetUserMedia || navigator.msGetUserMedia;

  // ---------------------- video handling ----------------------- 
  // start local video
  function startVideo() {
    getDeviceStream({video: true, audio: false})
    .then(function (stream) { // success
      localStream = stream;
      playVideo(localVideo, stream);
    }).catch(function (error) { // error
      console.error('getUserMedia error:', error);
      return;
    });
  }

  // stop local video
  function stopVideo() {
    pauseVideo(localVideo);
    stopLocalStream(localStream);
  }

  function stopLocalStream(stream) {
    let tracks = stream.getTracks();
    if (! tracks) {
      console.warn('NO tracks');
      return;
    }
    
    for (let track of tracks) {
      track.stop();
    }
  }
  
  function getDeviceStream(option) {
    if ('getUserMedia' in navigator.mediaDevices) {
      console.log('navigator.mediaDevices.getUserMadia');
      return navigator.mediaDevices.getUserMedia(option);
    }
    else {
      console.log('wrap navigator.getUserMadia with Promise');
      return new Promise(function(resolve, reject){    
        navigator.getUserMedia(option,
          resolve,
          reject
        );
      });      
    }
  }

  function playVideo(element, stream) {
    if ('srcObject' in element) {
      element.srcObject = stream;
    }
    else {
      element.src = window.URL.createObjectURL(stream);
    }
    element.play();
    element.volume = 0;
  }

  function pauseVideo(element) {
    element.pause();
    if ('srcObject' in element) {
      element.srcObject = null;
    }
    else {
      if (element.src && (element.src !== '') ) {
        window.URL.revokeObjectURL(element.src);
      }
      element.src = '';
    }
  }
</script>
</html>

built in
[vagrant@localhost webrtc]$ php -S 192.168.33.10:8000
あれ?
なに、みれないぞ。。

ソースコードがおかしいか?
同じソースコードで、*.github.ioにcommitして確認
=> 見れる

なにいいいいいいいいいいいいいいいいいいいいい
server側の設定か?
そんなばかな。。。 
=> 2時間くらい調査
=> 少し疲れたので休憩

=> あれ、videoタグの設定か?
=> chromeのカメラの設定を確認

httpだと、カメラのアクセスがブロックされるのね。。。
amazon linuxにmod-sslを入れます。

[vagrant@localhost webrtc]$ sudo yum install -y mod_ssl
–> Finished Dependency Resolution
Error: httpd24 conflicts with httpd-2.2.34-1.15.amzn1.x86_64
Error: httpd24-tools conflicts with httpd-tools-2.2.34-1.15.amzn1.x86_64

ぎゃああああああああああああああああああああああ

[vagrant@localhost webrtc]$ sudo yum install mod24_ssl
Complete!

もうやだ。