-
-
-
-
-
-
\ No newline at end of file
diff --git a/clientV/scripts/rtc.js b/clientV/scripts/rtc.js
deleted file mode 100644
index 9d74201..0000000
--- a/clientV/scripts/rtc.js
+++ /dev/null
@@ -1,220 +0,0 @@
-var name;
-var connectedUser;
-
-var loginInput = document.querySelector('#loginInput');
-var loginBt = document.querySelector('#loginBt');
-
-var callToUsernameInput = document.querySelector('#callInput');
-var callBtn = document.querySelector('#callBt');
-
-var hangUpBtn = document.querySelector('#disconnectBt');
-
-const remoteVideo = document.querySelector('#video');
-const remoteVideo2 = document.querySelector('#video2');
-
-var videoInput = document.querySelector('#videoInput');
-
-var yourConn;
-var stream;
-
-var reader;
-
-const configuration = {
- iceServers: [{ urls: "stun:stun.l.google.com:19302" }]
-};
-
-const offerOptions = {
- offerToReceiveAudio: 1,
- offerToReceiveVideo: 1
-};
-
-videoInput.addEventListener("change", function (event) {
- var file = this.files[0]
- var type = file.type
- var videoNode = remoteVideo2
- var canPlay = videoNode.canPlayType(type)
- if (canPlay === '') canPlay = 'no'
- var message = 'Can play type "' + type + '": ' + canPlay
- var isError = canPlay === 'no'
- //displayMessage(message, isError)
-
- if (isError) {
- return
- }
-
- var fileURL = URL.createObjectURL(file)
- videoNode.src = fileURL
-
-
-});
-
-remoteVideo.onplay = function() {
- if(remoteVideo.mozCaptureStream()) stream = remoteVideo.mozCaptureStream();
- else stream = remoteVideo.captureStream();
-
- //remoteVideo2.srcObject = stream;
-
- stream.getTracks().forEach((track) => {
- console.log("ADDED TRACK");
- yourConn.addTrack(track, stream);
- });
-
- //yourConn.addStream(stream);
-}
-
-// Login when the user clicks the button
-loginBt.addEventListener("click", function (event) {
- name = loginInput.value;
-
- if (name.length > 0) {
- send({
- type: "login",
- name: name
- });
- }
-
-});
-
-function handleLogin(success) {
- if (success === false) {
- alert("Ooops...try a different username");
- } else {
- //**********************
- //Starting a peer connection
- //**********************
-
- yourConn = new RTCPeerConnection(configuration);
-
- yourConn.ontrack = function (event) {
- console.log("GOT TRACK");
- remoteVideo.srcObject = event.streams[0];
- }
-
- if (stream) {
- remoteVideo2.srcObject = stream;
-
- stream.getTracks().forEach((track) => {
- yourConn.addTrack(track, stream);
- });
- }
-
- // setup stream listening
- //yourConn.addStream(stream);
-
-
- // Setup ice handling
- yourConn.onicecandidate = function (event) {
- if (event.candidate) {
- send({
- type: "candidate",
- candidate: event.candidate
- });
- }
- };
-
- yourConn.onnegotiationneeded = handleNegotiationNeededEvent;
-
- window.setInterval(getConnectionStats, 1000);
-
- }
-};
-
-function getConnectionStats() {
- /*yourConn.getStats().then(stats => {
- var statsOutput = "";
- stats.forEach(report => {
- statsOutput += report;
- });
- console.log(statsOutput);
- });*/
- console.log(yourConn.connectionState);
-
-}
-
-//initiating a call
-callBtn.addEventListener("click", function () {
- var callToUsername = callToUsernameInput.value;
-
- if (callToUsername.length > 0) {
-
- connectedUser = callToUsername;
-
- // create an offer
- yourConn.createOffer(function (offer) {
- send({
- type: "offer",
- offer: offer
- });
-
- yourConn.setLocalDescription(offer);
- }, function (error) {
- alert("Error when creating an offer");
- },
- offerOptions);
-
- }
-});
-
-//when somebody sends us an offer
-function handleOffer(offer, name) {
- console.log("GOT OFFER");
- connectedUser = name;
- yourConn.setRemoteDescription(new RTCSessionDescription(offer));
-
- //create an answer to an offer
- yourConn.createAnswer(function (answer) {
- yourConn.setLocalDescription(answer);
-
- send({
- type: "answer",
- answer: answer
- });
-
- }, function (error) {
- alert("Error when creating an answer");
- });
-};
-
-//when we got an answer from a remote user
-function handleAnswer(answer) {
- console.log("GOT ANSWER");
- yourConn.setRemoteDescription(new RTCSessionDescription(answer));
-};
-
-//when we got an ice candidate from a remote user
-function handleCandidate(candidate) {
- console.log("GOT CANDIDATE");
- yourConn.addIceCandidate(new RTCIceCandidate(candidate));
-};
-
-function handleNegotiationNeededEvent() {
- console.log("NEGOTIATION NEEDED");
- yourConn.createOffer().then(function(offer) {
- return yourConn.setLocalDescription(offer);
- })
- .then(function() {
- send({
- type: "video-offer",
- sdp: yourConn.localDescription
- });
- });
-}
-
-//hang up
-hangUpBtn.addEventListener("click", function () {
-
- send({
- type: "leave"
- });
-
- handleLeave();
-});
-
-function handleLeave() {
- connectedUser = null;
- remoteVideo.src = null;
-
- yourConn.close();
- yourConn.onicecandidate = null;
- //yourConn.onaddstream = null;
-};
diff --git a/clientV/scripts/rtc2.js b/clientV/scripts/rtc2.js
deleted file mode 100644
index ebe9a5a..0000000
--- a/clientV/scripts/rtc2.js
+++ /dev/null
@@ -1,658 +0,0 @@
-"use strict";
-
-// Get our hostname
-
-var myHostname = window.location.hostname;
-if (!myHostname) {
- myHostname = "localhost";
-}
-log("Hostname: " + myHostname);
-
-// WebSocket chat/signaling channel variables.
-
-var connection = null;
-var clientID = 0;
-
-// The media constraints object describes what sort of stream we want
-// to request from the local A/V hardware (typically a webcam and
-// microphone). Here, we specify only that we want both audio and
-// video; however, you can be more specific. It's possible to state
-// that you would prefer (or require) specific resolutions of video,
-// whether to prefer the user-facing or rear-facing camera (if available),
-// and so on.
-//
-// See also:
-// https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamConstraints
-// https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
-//
-
-var mediaConstraints = {
- audio: true, // We want an audio track
- video: {
- aspectRatio: {
- ideal: 1.333333 // 3:2 aspect is preferred
- }
- }
-};
-
-var myUsername = null;
-var targetUsername = null; // To store username of other peer
-var myPeerConnection = null; // RTCPeerConnection
-var transceiver = null; // RTCRtpTransceiver
-var webcamStream = null; // MediaStream from webcam
-
-// Output logging information to console.
-
-function log(text) {
- var time = new Date();
-
- console.log("[" + time.toLocaleTimeString() + "] " + text);
-}
-
-// Output an error message to console.
-
-function log_error(text) {
- var time = new Date();
-
- console.trace("[" + time.toLocaleTimeString() + "] " + text);
-}
-
-// Send a JavaScript object by converting it to JSON and sending
-// it as a message on the WebSocket connection.
-
-function sendToServer(msg) {
- var msgJSON = JSON.stringify(msg);
-
- log("Sending '" + msg.type + "' message: " + msgJSON);
- connection.send(msgJSON);
-}
-
-// Called when the "id" message is received; this message is sent by the
-// server to assign this login session a unique ID number; in response,
-// this function sends a "username" message to set our username for this
-// session.
-function setUsername() {
- myUsername = document.getElementById("name").value;
-
- sendToServer({
- name: myUsername,
- date: Date.now(),
- id: clientID,
- type: "username"
- });
-}
-
-// Open and configure the connection to the WebSocket server.
-
-function connect() {
- var serverUrl;
- var scheme = "ws";
-
- // If this is an HTTPS connection, we have to use a secure WebSocket
- // connection too, so add another "s" to the scheme.
-
- if (document.location.protocol === "https:") {
- scheme += "s";
- }
- serverUrl = scheme + "://" + myHostname + ":6503";
-
- log(`Connecting to server: ${serverUrl}`);
- connection = new WebSocket(serverUrl, "json");
-
- connection.onopen = function(evt) {
- document.getElementById("text").disabled = false;
- document.getElementById("send").disabled = false;
- };
-
- connection.onerror = function(evt) {
- console.dir(evt);
- }
-
- connection.onmessage = function(evt) {
- var chatBox = document.querySelector(".chatbox");
- var text = "";
- var msg = JSON.parse(evt.data);
- log("Message received: ");
- console.dir(msg);
- var time = new Date(msg.date);
- var timeStr = time.toLocaleTimeString();
-
- switch(msg.type) {
- case "id":
- clientID = msg.id;
- setUsername();
- break;
-
- case "username":
- text = "User " + msg.name + " signed in at " + timeStr + " ";
- break;
-
- case "message":
- text = "(" + timeStr + ") " + msg.name + ": " + msg.text + " ";
- break;
-
- case "rejectusername":
- myUsername = msg.name;
- text = "Your username has been set to " + myUsername +
- " because the name you chose is in use. ";
- break;
-
- case "userlist": // Received an updated user list
- handleUserlistMsg(msg);
- break;
-
- // Signaling messages: these messages are used to trade WebRTC
- // signaling information during negotiations leading up to a video
- // call.
-
- case "video-offer": // Invitation and offer to chat
- handleVideoOfferMsg(msg);
- break;
-
- case "video-answer": // Callee has answered our offer
- handleVideoAnswerMsg(msg);
- break;
-
- case "new-ice-candidate": // A new ICE candidate has been received
- handleNewICECandidateMsg(msg);
- break;
-
- case "hang-up": // The other peer has hung up the call
- handleHangUpMsg(msg);
- break;
-
- // Unknown message; output to console for debugging.
-
- default:
- log_error("Unknown message received:");
- log_error(msg);
- }
-
- // If there's text to insert into the chat buffer, do so now, then
- // scroll the chat panel so that the new text is visible.
-
- if (text.length) {
- chatBox.innerHTML += text;
- chatBox.scrollTop = chatBox.scrollHeight - chatBox.clientHeight;
- }
- };
-}
-
-// Handles a click on the Send button (or pressing return/enter) by
-// building a "message" object and sending it to the server.
-function handleSendButton() {
- var msg = {
- text: document.getElementById("text").value,
- type: "message",
- id: clientID,
- date: Date.now()
- };
- sendToServer(msg);
- document.getElementById("text").value = "";
-}
-
-// Handler for keyboard events. This is used to intercept the return and
-// enter keys so that we can call send() to transmit the entered text
-// to the server.
-function handleKey(evt) {
- if (evt.keyCode === 13 || evt.keyCode === 14) {
- if (!document.getElementById("send").disabled) {
- handleSendButton();
- }
- }
-}
-
-// Create the RTCPeerConnection which knows how to talk to our
-// selected STUN/TURN server and then uses getUserMedia() to find
-// our camera and microphone and add that stream to the connection for
-// use in our video call. Then we configure event handlers to get
-// needed notifications on the call.
-
-async function createPeerConnection() {
- log("Setting up a connection...");
-
- // Create an RTCPeerConnection which knows to use our chosen
- // STUN server.
-
- myPeerConnection = new RTCPeerConnection({
- iceServers: [ // Information about ICE servers - Use your own!
- {
- urls: "turn:" + myHostname, // A TURN server
- username: "webrtc",
- credential: "turnserver"
- }
- ]
- });
-
- // Set up event handlers for the ICE negotiation process.
-
- myPeerConnection.onicecandidate = handleICECandidateEvent;
- myPeerConnection.oniceconnectionstatechange = handleICEConnectionStateChangeEvent;
- myPeerConnection.onicegatheringstatechange = handleICEGatheringStateChangeEvent;
- myPeerConnection.onsignalingstatechange = handleSignalingStateChangeEvent;
- myPeerConnection.onnegotiationneeded = handleNegotiationNeededEvent;
- myPeerConnection.ontrack = handleTrackEvent;
-}
-
-// Called by the WebRTC layer to let us know when it's time to
-// begin, resume, or restart ICE negotiation.
-
-async function handleNegotiationNeededEvent() {
- log("*** Negotiation needed");
-
- try {
- log("---> Creating offer");
- const offer = await myPeerConnection.createOffer();
-
- // If the connection hasn't yet achieved the "stable" state,
- // return to the caller. Another negotiationneeded event
- // will be fired when the state stabilizes.
-
- if (myPeerConnection.signalingState != "stable") {
- log(" -- The connection isn't stable yet; postponing...")
- return;
- }
-
- // Establish the offer as the local peer's current
- // description.
-
- log("---> Setting local description to the offer");
- await myPeerConnection.setLocalDescription(offer);
-
- // Send the offer to the remote peer.
-
- log("---> Sending the offer to the remote peer");
- sendToServer({
- name: myUsername,
- target: targetUsername,
- type: "video-offer",
- sdp: myPeerConnection.localDescription
- });
- } catch(err) {
- log("*** The following error occurred while handling the negotiationneeded event:");
- reportError(err);
- };
-}
-
-// Called by the WebRTC layer when events occur on the media tracks
-// on our WebRTC call. This includes when streams are added to and
-// removed from the call.
-//
-// track events include the following fields:
-//
-// RTCRtpReceiver receiver
-// MediaStreamTrack track
-// MediaStream[] streams
-// RTCRtpTransceiver transceiver
-//
-// In our case, we're just taking the first stream found and attaching
-// it to the