Added rtcmulticonnection-v3@3.2.88 - Also, v2.2.2 is updated to suppo…

…rt remote audio+video recording in Canary.

RecordRTC can be used with v3.2.88 to record remote or local
audio+video streams.

Gruntfile.js

@@ -39,7 +39,7 @@ module.exports = function(grunt) {
                     'dev/StreamsHandler.js',
 
                     'dev/DetectRTC.js',
-                    'dev/getScreenId.js', // or Screen-Capturing.js
+                    'dev/getScreenId.js', // or getScreenId.js or Screen-Capturing.js
 
                     'dev/TextSenderReceiver.js',
                     'dev/FileProgressBarHandler.js',

LICENSE.md

@@ -0,0 +1,23 @@
+
+The MIT License
+================
+
+Copyright (c) 2014-2018 Muaz Khan \<muazkh@gmail.com\>
+
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is
+furnished to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in
+all copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+THE SOFTWARE.

README.md

@@ -64,7 +64,7 @@ All files from `/dist` directory are available on CDN: `https://cdn.webrtc-exper
 <script src="https://cdn.webrtc-experiment.com:443/rmc3.min.js"></script>
 
 <!-- or specific version -->
-<script src="https://github.com/muaz-khan/RTCMultiConnection/releases/download/3.2.87/rmc3.min.js"></script>
+<script src="https://github.com/muaz-khan/RTCMultiConnection/releases/download/3.2.88/rmc3.min.js"></script>
 ```
 
 If you're sharing files, you also need to link:
@@ -76,7 +76,7 @@ If you're sharing files, you also need to link:
 <script src="https://cdn.webrtc-experiment.com:443/rmc3.fbr.min.js"></script>
 
 <!-- or specific version -->
-<script src="https://github.com/muaz-khan/RTCMultiConnection/releases/download/3.2.87/rmc3.fbr.min.js"></script>
+<script src="https://github.com/muaz-khan/RTCMultiConnection/releases/download/3.2.88/rmc3.fbr.min.js"></script>
 ```
 
 > You can link multiple files from `dev` directory. Order doesn't matters.
@@ -282,14 +282,34 @@ connection.resetTrack();
 
 ### `onUserStatusChanged`
 
-This even allows you show online/offline statuses of the user:
+This event allows you show online/offline statuses of the user:
 
 ```javascript
 connection.onUserStatusChanged = function(status) {
 	document.getElementById(event.userid).src = status === 'online' ? 'online.gif' : 'offline.gif';
 };
 ```
 
+You can even manually call above method from `onopen`, `onstream` and similar events to get the most accurate result possible:
+
+```javascript
+connection.onopen = connection.stream = function(event) {
+    connection.onUserStatusChanged({
+        userid: event.userid,
+        extra: event.extra,
+        status: 'online'
+    });
+};
+
+connection.onleave = connection.streamended = connection.onclose = function(event) {
+    connection.onUserStatusChanged({
+        userid: event.userid,
+        extra: event.extra,
+        status: 'offline'
+    });
+};
+```
+
 ### `getSocket`
 
 This method allows you get the `socket` object used for signaling (handshake/presence-detection/etc.):
@@ -347,7 +367,7 @@ connection.getUserMediaHandler(options);
 
 Its defined here:
 
-* [getUserMedia.js#L20](https://github.com/muaz-khan/RTCMultiConnection/blob/master/RTCMultiConnection-v3.0/dev/getUserMedia.js#L20)
+* [getUserMedia.js#L20](https://github.com/muaz-khan/RTCMultiConnection/tree/master/devgetUserMedia.js#L20)
 
 ## `becomePublicModerator`
 
@@ -459,7 +479,7 @@ connection.checkPresence('roomid', function(isRoomEists, roomid) {
 
 ## `onReadyForOffer`
 
-This even is fired as soon as callee says "I am ready for offer. I enabled camera. Please create offer and share.".
+This event is fired as soon as callee says "I am ready for offer. I enabled camera. Please create offer and share.".
 
 ```javascript
 connection.onReadyForOffer = function(remoteUserId, userPreferences) {
@@ -538,8 +558,8 @@ connection.getAllParticipants().forEach(function(remoteUserId) {
 	var user = connection.peers[remoteUserId];
 	console.log(user.extra);
 
-	user.peer.connection.close();
-	alert(user.peer.connection === webkitRTCPeerConnection);
+	user.peer.close();
+	alert(user.peer === webkitRTCPeerConnection);
 });
 ```
 
@@ -561,7 +581,7 @@ This method allows you skip Socket.io and force Firebase or PubNub or WebSockets
 connection.setCustomSocketHandler(FirebaseConnection);
 ```
 
-Please check [`FirebaseConnection`](https://github.com/muaz-khan/RTCMultiConnection/blob/master/RTCMultiConnection-v3.0/dev/FirebaseConnection.js) or [`PubNubConnection.js`](https://github.com/muaz-khan/RTCMultiConnection/blob/master/RTCMultiConnection-v3.0/dev/PubNubConnection.js) to understand how it works.
+Please check [`FirebaseConnection`](https://github.com/muaz-khan/RTCMultiConnection/tree/master/devFirebaseConnection.js) or [`PubNubConnection.js`](https://github.com/muaz-khan/RTCMultiConnection/tree/master/devPubNubConnection.js) to understand how it works.
 
 ## `enableLogs`
 
@@ -1055,6 +1075,55 @@ connection.onunmute = function(e) {
 };
 ```
 
+## HD Streaming
+
+```javascript
+connection.bandwidth = {
+    audio: 128,
+    video: 1024,
+    screen: 1024
+};
+
+var videoConstraints = {
+    mandatory: {
+        maxWidth: 1920,
+        maxHeight: 1080,
+        minAspectRatio: 1.77,
+        minFrameRate: 3,
+        maxFrameRate: 64
+    },
+    optional: []
+};
+
+connection.mediaConstraints.video = videoConstraints;
+```
+
+## Default devices?
+
+By default, RTCMultiConnection tries to use last available microphone and camera. However you can disable this behavior and ask to use default devices instead:
+
+```javascript
+// pass second parameter to force options
+var connection = new RTCMultiConnection(roomId, {
+    useDefaultDevices: true
+});
+```
+
+## Auto open or join?
+
+By default, you always have to call `open` or `join` or `openOrJoin` methods manually. However you can force RTCMultiConnection to auto open/join room as soon as constructor is initialized.
+
+```javascript
+// pass second parameter to force options
+var connection = new RTCMultiConnection(roomId, {
+    autoOpenOrJoin: true
+});
+```
+
+For low-latency audio:
+
+* https://twitter.com/WebRTCWeb/status/499102787733450753
+
 ## RTCMultiConnection v2.2.2 Demos
 
 | Experiment Name        | Demo           | Source Code |
@@ -1108,6 +1177,10 @@ v2.2.2 is available here:
 
 * https://github.com/muaz-khan/RTCMultiConnection/tree/master/v2.2.2
 
+## Twitter
+
+* https://twitter.com/WebRTCWeb i.e. @WebRTCWeb
+
 ## License
 
 [RTCMultiConnection](https://github.com/muaz-khan/RTCMultiConnection) is released under [MIT licence](https://github.com/muaz-khan/RTCMultiConnection/blob/master/LICENSE.md) . Copyright (c) [Muaz Khan](http://www.MuazKhan.com/).