Releases · QuickBlox/quickblox-javascript-sdk

Dependencies updated:
"lodash": ">=4.17.13",
"lodash.template": ">=4.5.0",
"handlebars": ">=4.1.0",
"js-yaml": ">=3.13.1",
"marked": "^0.6.0",
"ws": ">=3.3.1",
"node.extend": ">=1.1.7",
"cached-path-relative": ">=1.0.2",
"karma": "^4.2.0"

Assets 2

27 Nov 17:56

sshaforenkoqb

2.12.2

84204d0

2.12.2

An issue with video calls in Safari 11 fixed.

Assets 2

04 Nov 09:26

AndriiPovelychenko

2.12.1

09309bb

2.12.1

Fix of security vulnerabilities in dependencies

Assets 2

24 May 13:15

dimaspirit

2.12.0

ffe19a4

2.12.0

Removed:

Remove unused samples (users, roster). All cases will be shown at chat, data, webrtc samples;
The window.navigator.onLine check was remove (the check was move to the WebRTC Sample); To make sure that a client has internet connecteion before make a call;

Fixed:

during the call (WebRTC session) the WebRTC Sample will be able to switch to other camera, if it is possible and would stop call if all media devices were unplugged;
bugs were fixed when user doesn't allow permanent permissions for getUserMedia;

Added:

Ability to restore connect to chat by time interval after disconnect, if it wasn't voluntary (Added for node-xmpp-client and nativescript-xmpp-client, updated for Strophe.js client).
Add a property chatReconnectionTimeInterval to config, by default is 5 sec;
QB.chat.disconnect() stops reconnection actions;
The call of QB.chat.connect(params, callback) when chat is in connecting state (the connection one by one) was blocked, the callback funtion will return an error ('Status.REJECT - The connection is still in the Status.CONNECTING state');
QB.webrtc.onDevicesChangeListener() was added - the listener that is called when a media device has been plugged or unplugged;
An ability to change audio and video tracks was added (switch cameras), use the method webRTCSession.switchMediaTracks(deviceIds, cb).
Supported and tested on Firefox from v.60.

Here is code snippet how to replace audio/video tracks:

var switchMediaTracksBtn = document.getElementById('confirmSwitchMediaTracks');

var webRTCSession = QB.webrtc.createNewSession(params);

QB.webrtc.getMediaDevices('videoinput').then(function(devices) {
    var selectVideoInput = document.createElement('select'),
        selectVideoInput.id = 'videoInput',
        someDocumentElement.appendChild(selectVideoInput);

    if (devices.length > 1) {
        for (var i = 0; i !== devices.length; ++i) {
            var device = devices[i],
                option = document.createElement('option');

            if (device.kind === 'videoinput') {
                option.value = device.deviceId;
                option.text = device.label;
                selectVideoInput.appendChild(option);
            }
        }
    }
}).catch(function(error) {
    console.error(error);
});

QB.webrtc.getMediaDevices('audioinput').then(function(devices) {
    var selectAudioInput = document.createElement('select'),
        selectAudioInput.id = 'audioInput',
        someDocumentElement.appendChild(selectAudioInput);

    if (devices.length > 1) {
        for (var i = 0; i !== devices.length; ++i) {
            var device = devices[i],
                option = document.createElement('option');

            if (device.kind === 'audioinput') {
                option.value = device.deviceId;
                option.text = device.label;
                selectAudioInput.appendChild(option);
            }
        }
    }
}).catch(function(error) {
    console.error(error);
});

switchMediaTracksBtn.onclick = function(event) {
    var audioDeviceId = document.getElementById('audioInput').value || undefined,
        videoDeviceId = document.getElementById('videoInput').value || undefined,
        deviceIds = {
            audio: audioDeviceId,
            video: videoDeviceId,
        };

    var callback = function(error, stream) {
            if (err) {
                console.error(error);
            } else {
                console.log(stream);
            }
         };

    // Switch media tracks in audio/video HTML's element (the local stream)
    // replace media tracks in peers (will change media tracks for each user in WebRTC session)
    webRTCSession.switchMediaTracks(deviceIds, callback);
}

Assets 2

12 Apr 12:43

dimaspirit

2.11.0

af86e55

2.11.0

New:

Add header 'QB-OS' in all API requests for improving analytics;

Updated:

Encode body of all API requests;
Encode based on this https://en.wikipedia.org/wiki/Percent-encoding#Percent-encoding_reserved_characters;
Rework muc.join method:
- Rework putting parameters of the muc.join method. Now could pass jid or id of a dialog;
- Now uses Node.js callbacks approach instead of a returned stanza if you pass 2 arguments to the callback function. If you pass 1 arguments you will get stanza element as before;

QB.chat.muc.join(dialogId, function(error, response) {
   if(error) {
      console.log('Error is null when all is Ok');
  }

  console.log(`response.dialogId` is always contains in response);
});

QB.webrtc.onCallStatsReport(session, userId, stats, error) will return new (uptaded) stats:

stats = {
    local: {
        audio: {
            bitrate: 71, // kilobits per second (kbps)
            bytesSent: 226410,
            packetsSent: 1250, 
            timestamp: 1522680935736
        },
        video: {
            frameHeight: 480,
            frameWidth: 640,
            framesPerSecond: 25.8,
            bitrate: 498,  // kilobits per second (kbps)
            bytesSent: 1438862, 
            packetsSent: 1498,
            timestamp: 1522680935736
        },
        candidate: {
            protocol: "udp"
            ip: "192.168.1.179"
            port: 51038
        }
    },
    remote: {
        audio: {
            bitrate: 47, // kilobits per second (kbps)
            bytesReceived: 148211,
            packetsReceived: 1250,
            timestamp: 1522680935736
        },
        video: {
            frameHeight: 480,
            frameWidth: 640,
            framesPerSecond: 30.4,
            bitrate: 533, // kilobits per second (kbps)
            bytesReceived: 1663716,
            packetsReceived: 1498,
            timestamp: 1522680935736
        },
        candidate: {
            protocol: "udp",
            ip: "192.168.1.179",
            port: 51908
        }
    }
}

Assets 2

15 Mar 10:45

dimaspirit

2.10.0

64db7b1

2.10.0

Improvements / Updated:

Add the opportunity to choose crypto standard (sha1, sha256);
By default, uses sha1. If you want to use sha256 add hash property to config;

var CONFIG = {
  debug: {mode: 1},
  hash: 'sha256'
};
QB.init(3477, 'ChRnwEJ3WzxH9O4', 'AS546kpUQ2tfbvv', CONFIG);

Assets 2

01 Mar 09:04

dimaspirit

2.9.0

2c7b71e

2.9.0

Added:

Add possibility to set a bandwidth limit for the video call.
Set the bandwidth at create sssion by QB.webrtc.createNewSession(calleesIds, sessionType, callerID, { bandwidth: Number });.
Add DELETE Custom Object by criteria by updated a method QB.data.delete.
Now the method takes QB.data.delete(className, {string|array|object}, callback) where second parameter an id (String) or a list of ids (Array) or criteria rules (Object) to delete.
Check out docs for more details;

Updated:

Normalize an answer on DELETE Custom Object by criteria for any passed types.
Check out docs for more details;
Update QB.webrtc.onCallStatsReport(session, userId, stats, error) listener;

Samples:

Add ability set credentials and endpoints by search param of URL in webRTC sample:
https//:www.host.com/?appId={Number}&authKey={String}&authSecret={String}&endpoints.api={String}&endpoints.chat={String}.
Also remove production configuration/app and leave stage app configuration;
Add a bandwidth configuration to WebRTC sample;
Remove code for "Hack for Firefox" (https://bugzilla.mozilla.org/show_bug.cgi?id=852665) cause we are support FF52+ now;