QuickBlox Video Calling API is built on top of WebRTC. It allows adding real-time video communication features into your app similar to Skype using API easily. The communication is happening between peers representing camera devices. There are two peer types:
  • Local peer is a device running the app right now.
  • Remote peer is an opponent device.
Establishing real-time video communication between two peers involves 3 phases:
  1. Signaling. At this phase, the peers’ local IPs and ports where they can be reached (ICE candidates) are exchanged as well their media capabilities and call session control messages.
  2. Discovery. At this phase, the public IPs and ports at which endpoints can be reached are discovered by STUN/TURN server.
  3. Establishing a connection. At this phase, the data are sent directly to each party of the communication process.
In order to start using Video Calling Module, you need to connect to QuickBlox Chat first. The signaling in the QuickBox WebRTC module is implemented over the XMPP protocol using QuickBlox Chat Module. It acts as a signaling transport for Video Calling API.
Please use this WebRTC Video Calling to make the Group Calls with 4 or fewer users. Because of Mesh architecture we use for multi-point where every participant sends and receives its media to all other participants, the current solution supports group calls with up to 4 people.
Visit Key Concepts page to learn the most important QuickBlox concepts.

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Before you begin

  1. Register a QuickBlox account. This is a matter of a few minutes and you will be able to use this account to build your apps.
  2. Configure QuickBlox SDK for your app. Check out Setup page for more details.
  3. Create a user session to be able to use QuickBlox functionality. See Authentication page to learn how to do it.
  4. Connect to the Chat server to provide a signaling mechanism for Video Calling API. Follow our Chat page to learn about chat connection settings and configuration.

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Initiate a call

To initiate a call, proceed with the steps below.

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Create session

First, you need to create a call session. This is done by using the createNewSession() method. Pass opponent ID/IDs and a call type (video or audio) to it. The session object can be received either as a result of the createNewSession() method or in each event listener callback. Learn more about the event listener in this section.
JavaScript
var calleesIds = [56, 76, 34]; // Users' ids
var sessionType = QB.webrtc.CallType.VIDEO; // AUDIO is also possible
var additionalOptions = {};

var session = QB.webrtc.createNewSession(calleesIds, sessionType, null, additionalOptions);
ArgumentRequiredDescription
calleesIdsyesIDs of opponents.
sessionTypeyesCall type: AUDIO or VIDEO.

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Access local media stream

Once a call session is created, you need to get access to the user’s devices (webcam/microphone) in order to make a call.
JavaScript
var mediaParams = {
  audio: true,
  video: true,
  options: {
    muted: true,
    mirror: true,
  },
  elemId: "localVideoElem",
};

session.getUserMedia(mediaParams, function (error, stream) {
  if (error) {
  } else {
		//run call function here
  }
});
ParametersRequiredDescription
mediaParamsyesMedia stream constrains.
function()yesSpecifies the callback function which receives a local media stream once it is ready.
The mediaParams object has the following fields:
FieldsRequiredDescription
audioyesEnable audio. Boolean parameter. true is enabled, false is disabled.
videoyesEnable video. Boolean parameter. true is enabled, false is disabled.
optionsnoSpecifies two media options:- mute. Mute audio. true is muted, false is unmuted.- mirror. Enable the mirror. true is enabled, false is disabled.
elemIdnoID attribute of the video element to attach a local video stream to.
This method lets the browser ask the user for permission to use devices. You should allow this dialog to access the stream. Otherwise, the browser cannot obtain access and will throw an error for getUserMedia callback function. For more information about possible audio/video constraints, here is a good code sample from WebRTC team on how to work with getUserMedia constraints.

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Attach local media stream

You also need to attach your local media stream to HTML video element. The method below attaches the Media Stream Object to your DOM element and autoplays it.
JavaScript
session.attachMediaStream('myVideoElementId', stream);
The same method is used for attaching the remote video stream that is received from your opponents.

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Make a call

Now that you have attached the local media stream, you can initiate a call by calling the call() method.
JavaScript
var extension = {};
session.call(extension, function(error) {

});
ArgumentRequiredDescription
extensionyesCustom user data. Specify any key-value pairs. In each pair, the key and value are both string values.
fuction()yesSpecifies the callback function. The error is received in case of unsuccessful callback.
As a result, your opponent will receive an onCallListener() callback specifying that a new call session has been received.
JavaScript
QB.webrtc.onCallListener = function(session, extension) {

	// if you are going to take a call
	session.getUserMedia(mediaParams, function (error, stream) {
  if (error) {
    } else {
      //run accept function here
    }
  });

};
Your opponent receives an incoming call through either the established connection with the Chat server or a VoIP Push Notifications when the app is in the background.

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Accept a call

To accept a call request, call accept() method.
JavaScript
var extension = {};
session.accept(extension);
After this, your opponents will get a confirmation in the following callback.
JavaScript
QB.webrtc.onAcceptCallListener = function(session, userId, extension) {

	//here you can stop the ringtone

};
Also, both the caller and opponents will get a special callback with the remote stream.
JavaScript
QB.webrtc.onRemoteStreamListener = function(session, userID, remoteStream) {
  // attach the remote stream to DOM element
  session.attachMediaStream("remoteOpponentVideoElementId", remoteStream);
};
From this point, you and your opponents can see each other.

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Reject a call

To reject a call request, use reject() method.
JavaScript
var extension = {};
session.reject(extension);
After this, the caller will get a confirmation in the following callback.
JavaScript
QB.webrtc.onRejectCallListener = function(session, userId, extension) {

};

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End a call

To end a call, use stop() method and pass sessionId parameter to tell SDK which call session to end.
JavaScript
var extension = {};
session.stop(extension);
After this, the opponents will get a confirmation in the following callback.
JavaScript
QB.webrtc.onStopCallListener = function(session, userId, extension) {

};

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Call changes

Sometimes, you make changes during an active call (for example, applying some CSS filter to a video element) that needs to be fed to the opponents and updated. To allow for this, you can use update() method to send instructions to the opponents on how to update the UI.
JavaScript
var extension = {
  userInfo: {
    videoFilter: "sepica"
  }
};
session.update(extension);
Your opponents will receive the following callback.
JavaScript
QB.webrtc.onUpdateCallListener = function(session, userId, extension) {

};

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Event listener

To process events such as incoming call, call reject, hang up, etc. you need to set up the event listener. The event listener processes various events that happen with the call session or peer connection in your app. Using the callbacks provided by the event delegate, you can implement and execute the event-related processing code. For example, the onAcceptCallListener callback is received when your call has been accepted by the user. This callback receives information about the call session, user ID who accepted the call, and additional key-value data about the user. QuickBlox JavaScript SDK persistently interacts with the server via XMPP connection that works as a signaling transport for establishing a call between two or more peers. It receives the callbacks of the asynchronous events which happen with the call and peer connection. This allows you to track these events and build your own video calling features around them. To track call session and peer connection events, you should use the event listener. The supported event callbacks for a call session and peer connection are listed in the table below.
MethodInvoked when
onCallListenerA new call session has been received.
onAcceptCallListenerA call has been accepted.
onRejectCallListenerA call has been rejected.
onStopCallListenerAn accepted call has been ended by the peer by pressing the hang-up button.
onUserNotAnswerListenerA user did not respond to your call within the dedicated time interval.
onSessionCloseListenerA call session has been closed.
onCallStatsReportAn updated stats report, which is called by timeout, has been received for the user ID.
onSessionConnectionStateChangedListenerA peer connection state has been changed. View Peer connection states section for more details.
onUpdateCallListenerCalled when updates to the active call are made.
onInvalidEventsListenerCalled when some operation cannot be performed (for example, accept() method is called on the closed call session).
onRemoteStreamListenerA user received a stream from the opponent.
The following code lists all supported event callbacks.
JavaScript
QB.webrtc.onCallListener = function(session, extension) { }

QB.webrtc.onAcceptCallListener = function(session, userId, extension) { }

QB.webrtc.onRejectCallListener = function(session, userId, extension) { }

QB.webrtc.onStopCallListener = function(session, userId, extension) { }

QB.webrtc.onUpdateCallListener = function(session, userId, extension) { }

QB.webrtc.onInvalidEventsListener = function(eventName, session, userId, userInfo) {}

QB.webrtc.onUserNotAnswerListener = function(session, userId) { }

QB.webrtc.onRemoteStreamListener = function(session, userId, stream) { }

QB.webrtc.onSessionConnectionStateChangedListener = function(session, userId, connectionState) { }

QB.webrtc.onSessionCloseListener = function(session) {}

QB.webrtc.onCallStatsReport = function(session, userId, stats, error) { }
Go to the Resources section to see a sequence diagram for a regular call workflow.

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Call session states

The following table lists all supported call session states.
StateDescription
NEWA new call session has such a state.
ACTIVEA call session gets this state when you initiated or accepted the incoming call.
HUNGUPA call session gets this state when you hang up, before closing the call session.
REJECTEDA call session gets this state when you reject an incoming call before closing the call session.
CLOSEDA call session is closed.

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Call reconnection states

Since version 2.15.0 Quickblox-WebRTC ```JavaScript JavaScript SDK supports reconnection functional.

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Manage reconnection

You could add onReconnectListener to your client code to define session states and manage reconnection states.
JavaScript
QB.webrtc.onReconnectListener = function (session, userId, state) {
    console.group('onReconnectListener');
    console.log('Session: ', session);
    console.log('User Id: ', userId);
    console.log('Reconnection state: ', state);
    console.groupEnd();
};
Doesn’t invoke in conference.
The following table lists all supported call reconnection states:
StateDescription
ReconnectionState.RECONNECTINGA connection with opponent in reconnection progress after ICE connection failed.
ReconnectionState.RECONNECTEDA connection with opponent was successfully established.
ReconnectionState.FAILEDA connection with opponent isn’t restored by disconnect time interval.
You can also increase disconnect timeout. By default the time is set to 30 seconds. Minimal time is 10 seconds. It’s time while users can have availability to reconnect. You could set the disconnectTimeInterval in the CONFIG object and then call the init() method.
JavaScript
var APPLICATION_ID = 41;
var AUTH_KEY = "lkjdueksu7392kj";
var AUTH_SECRET = "iiohfdija792hj";
var ACCOUNT_KEY = "sdjfnksnlk2bk1k34kb";
var CONFIG = {
  // other settings
  webrtc: {
    disconnectTimeInterval: 30,
  }
  // other settings
};
QB.init(APPLICATION_ID, AUTH_KEY, AUTH_SECRET, ACCOUNT_KEY, CONFIG);

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Peer connection states

The following table lists all supported states of peer connection:
StateDescription
QB.webrtc.SessionConnectionState.CONNECTINGThe ICE agent has been given one or more remote candidates and is checking pairs of local and remote candidates against one another to try to find a compatible match, but has not yet found a pair which will allow the peer connection to be made.
QB.webrtc.SessionConnectionState.CONNECTEDA usable pairing of local and remote candidates has been found for all components of the connection, and the connection has been established.
QB.webrtc.SessionConnectionState.CLOSEDA peer connection was closed. But the call session can still be open because there can several peer connections in a single call session.The ICE agent for this RTCPeerConnection has shut down and is no longer handling requests.
QB.webrtc.SessionConnectionState.FAILEDOne or more of the ICE transports on the connection is in the failed state. This can occur in different circumstances, for example, bad network, etc.
QB.webrtc.SessionConnectionState.COMPLETEDThe ICE agent has finished gathering candidates, has checked all pairs against one another, and has found a connection for all components.

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Resources

A regular call workflow.
JS_SDK_initiate_accept_hang_up.png