BasicUsage.md

Basic Usage

For starters we're going to import everything ready to use.
Most of the included functionality is similar to how you would deal with WebRTC in your browser.
We support a lot of the official WebRTC APIs, see this document for more details.
If you see functions that are listed in the document above but not listed below then they are likely not supported by this module yet and will most likely be supported in the near future, we're open to contributions.

import {
	ScreenCapturePickerView,
	RTCPeerConnection,
	RTCIceCandidate,
	RTCSessionDescription,
	RTCView,
	MediaStream,
	MediaStreamTrack,
	mediaDevices,
	registerGlobals
} from 'react-native-webrtc';

Registering Globals

You'll only really need to use this function if you are mixing project development with libraries that use browser based WebRTC functions. Also applies if you are making your project compatible with react-native-web.

registerGlobals();

Here is a list of everything that will be linked up.
You can also find a shim for react-native-web over here.

navigator.mediaDevices.getUserMedia()
navigator.mediaDevices.getDisplayMedia()
navigator.mediaDevices.enumerateDevices()
window.RTCPeerConnection
window.RTCIceCandidate
window.RTCSessionDescription
window.MediaStream
window.MediaStreamTrack

Get Available Media Devices

Some devices might not have more than 1 camera. The following will allow you to know how many cameras the device has. You can ofcourse use enumerateDevices to list other media device information too.

let cameraCount = 0;

try {
	const devices = await mediaDevices.enumerateDevices();

	devices.map( device => {
		if ( device.kind != 'videoinput' ) { return; };

		cameraCount = cameraCount + 1;
	} );
} catch( err ) {
	// Handle Error
};

Defining Media Constraints

By default we're sending both audio and video.
This will allow us to toggle the video stream during a call.

let mediaConstraints = {
	audio: true,
	video: {
		frameRate: 30,
		facingMode: 'user'
	}
};

Getting a Media Stream using getUserMedia

If you only want a voice call then you can flip isVoiceOnly over to true.
You can then cycle and enable or disable the video tracks on demand during a call.

let localMediaStream;
let isVoiceOnly = false;

try {
	const mediaStream = await mediaDevices.getUserMedia( mediaConstraints );

	if ( isVoiceOnly ) {
		let videoTrack = await mediaStream.getVideoTracks()[ 0 ];
		videoTrack.enabled = false;
	};

	localMediaStream = mediaStream;
} catch( err ) {
	// Handle Error
};

Getting a Media Stream using getDisplayMedia

This will allow capturing the device screen, also requests permission on execution.
Android 10+ requires that a foreground service is running otherwise capturing won't work, follow this solution.

try {
	const mediaStream = await mediaDevices.getDisplayMedia();

	localMediaStream = mediaStream;
} catch( err ) {
	// Handle Error
};

Destroying the Media Stream

Cycling all of the tracks and stopping them is more than enough to clean up after a call has finished.
You won't usually need to do this for remote tracks, only local.

localMediaStream.getTracks().forEach(
	track => track.stop()
);

localMediaStream = null;

Defining Peer Constraints

We're only specifying a STUN server but you should look at also using a TURN server.
If you want to improve call reliability then check this guide.

let peerConstraints = {
	iceServers: [
		{
			urls: 'stun:stun.l.google.com:19302'
		}
	]
};

Creating a Peer Connection

Here we're creating a peer connection required to get a call started.
You can also hook up events by directly overwriting functions instead of using event listeners.

let peerConnection = new RTCPeerConnection( peerConstraints );

peerConnection.addEventListener( 'connectionstatechange', event => {} );
peerConnection.addEventListener( 'icecandidate', event => {} );
peerConnection.addEventListener( 'icecandidateerror', event => {} );
peerConnection.addEventListener( 'iceconnectionstatechange', event => {} );
peerConnection.addEventListener( 'icegatheringstatechange', event => {} );
peerConnection.addEventListener( 'negotiationneeded', event => {} );
peerConnection.addEventListener( 'signalingstatechange', event => {} );
peerConnection.addEventListener( 'track', event => {} );

Destroying the Peer Connection

When ending a call you should always make sure to dispose of everything ready for another call.
The following should dispose of everything related to the peer connection.

peerConnection.close();
peerConnection = null;

Adding the Media Stream

After using one of the media functions above you can then add the media stream to the peer.
The negotiation needed event will be triggered on the peer connection afterwords.

localMediaStream.getTracks().forEach( 
	track => peerConnection.addTrack( track, localMediaStream );
);

Creating a Data Channel

Usually the call initialiser would create the data channel but it can be done on both sides.
The negotiation needed event will be triggered on the peer connection afterwords.

let datachannel = peerConnection.createDataChannel( 'my_channel' );

datachannel.addEventListener( 'open', event => {} );
datachannel.addEventListener( 'close', event => {} );
datachannel.addEventListener( 'message', message => {} );

Handling Data Channels

The following event is for the second client, not the client which created the data channel.
Unless ofcourse you want both sides to create separate data channels.

peerConnection.addEventListener( 'datachannel', event => {
	let datachannel = event.channel;

	// Now you've got the datachannel.
	// You can hookup and use the same events as above ^
} );

Sending a Message via the Data Channel

You can send a range of different data types over data channels, we're gong to send a simple string.
Bare in mind there are limits so sending large amounts of data isn't usually advised.

datachannel.send( 'Hey There!' );

Destroying the Data Channel

When the peer connection is destroyed, data channels should also be destroyed automatically.
But as good practice, you can always close them yourself.

datachannel.close();
datachannel = null;

Defining Session Constraints

As mentioned above by default we're going for the approach of offering both video and voice.
That will allow you to enable and disable video streams on demand while a call is active.

let sessionConstraints = {
	mandatory: {
		OfferToReceiveAudio: true,
		OfferToReceiveVideo: true,
		VoiceActivityDetection: true
	}
};

Creating an Offer

Executed by the call initialiser after media streams have been added to the peer connection.
ICE Candidate creation and gathering will start as soon as an offer has been created.

try {
	const offerDescription = await peerConnection.createOffer( sessionConstraints );
	await peerConnection.setLocalDescription( offerDescription );

	// Send the offerDescription to the other participant.
} catch( err ) {
	// Handle Errors
};

Creating an Answer

All parties will need to ensure they are handling ICE Candidates correctly.
Otherwise the offer and answer handshake stage will go a little wonky.

try {
	// Use the received offerDescription
	const offerDescription = new RTCSessionDescription( offerDescription );
	await peerConnection.setRemoteDescription( offerDescription );

	const answerDescription = await peerConnection.createAnswer();
	await peerConnection.setLocalDescription( answerDescription );

	// Send the answerDescription back as a response to the offerDescription.
} catch( err ) {
	// Handle Errors
};

Toggle the Active Microphone

During an active call you might want to mute your microphone.
Easy to accomplish by flipping the track enabled value to false, also possible on remote tracks.

let isMuted = false;

try {
	const audioTrack = await localMediaStream.getAudioTracks()[ 0 ];
	audioTrack.enabled = !audioTrack.enabled;

	isMuted = !isMuted;
} catch( err ) {
	// Handle Error
};

Switching the Active Camera

Naturally we assume you'll be using the front camera by default when starting a call.
So we set isFrontCam as true and let the value flip on execution.

let isFrontCam = true;

try {
	// Taken from above, we don't want to flip if we don't have another camera.
	if ( cameraCount < 2 ) { return; };

	const videoTrack = await localMediaStream.getVideoTracks()[ 0 ];
	videoTrack._switchCamera();

	isFrontCam = !isFrontCam;
} catch( err ) {
	// Handle Error
};

Rendering the Media Stream

Once you've gained a local and/or remote stream then rendering it is as follows.
Don't forget, the user facing camera is usually mirrored.

<RTCView
	mirror={true}
	objectFit={'cover'}
	streamURL={localMediaStream.toURL()}
	zOrder={0}
/>

Param	Type	Default	Description
mirror	boolean	false	Indicates whether the video specified by streamURL should be mirrored.
objectFit	string	'contain'	Can be 'contain' or 'cover' nothing more or less.
streamURL	string	'streamURL'	Required to have an actual video stream rendering.
zOrder	number	0	Similar to zIndex.

Controlling remote audio tracks

Remote audio tracks are automatically handled and played through your audio output. The volume of individual tracks can be set with the _setVolume function. It takes in a number between 0 to 10, defaults to 1.

const audioTrack = remoteMediaStream.getAudioTracks()[0];
audioTrack._setVolume(0.5);