blob: 42ac308e04ee39aaf5a75315f1d8de5a5ed32fa0 [file] [log] [blame]
/*
* Copyright (C) 2022 Savoir-faire Linux Inc.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU Affero General Public License as
* published by the Free Software Foundation; either version 3 of the
* License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Affero General Public License for more details.
*
* You should have received a copy of the GNU Affero General Public
* License along with this program. If not, see
* <https://www.gnu.org/licenses/>.
*/
import { WebRtcIceCandidate, WebRtcSdp, WebSocketMessageType } from 'jami-web-common';
import { createContext, useCallback, useContext, useEffect, useMemo, useState } from 'react';
import LoadingPage from '../components/Loading';
import { WithChildren } from '../utils/utils';
import { useAuthContext } from './AuthProvider';
import { ConversationContext } from './ConversationProvider';
import { IWebSocketContext, WebSocketContext } from './WebSocketProvider';
interface IWebRtcContext {
iceConnectionState: RTCIceConnectionState | undefined;
mediaDevices: Record<MediaDeviceKind, MediaDeviceInfo[]>;
localStream: MediaStream | undefined;
remoteStreams: readonly MediaStream[] | undefined;
getUserMedia: () => Promise<void>;
sendWebRtcOffer: () => Promise<void>;
closeConnection: () => void;
}
const defaultWebRtcContext: IWebRtcContext = {
iceConnectionState: undefined,
mediaDevices: {
audioinput: [],
audiooutput: [],
videoinput: [],
},
localStream: undefined,
remoteStreams: undefined,
getUserMedia: async () => {},
sendWebRtcOffer: async () => {},
closeConnection: () => {},
};
export const WebRtcContext = createContext<IWebRtcContext>(defaultWebRtcContext);
export default ({ children }: WithChildren) => {
const { account } = useAuthContext();
const [webRtcConnection, setWebRtcConnection] = useState<RTCPeerConnection | undefined>();
const webSocket = useContext(WebSocketContext);
useEffect(() => {
if (!webRtcConnection && account) {
const iceServers: RTCIceServer[] = [];
if (account.getDetails()['TURN.enable'] === 'true') {
iceServers.push({
urls: 'turn:' + account.getDetails()['TURN.server'],
username: account.getDetails()['TURN.username'],
credential: account.getDetails()['TURN.password'],
});
}
if (account.getDetails()['STUN.enable'] === 'true') {
iceServers.push({
urls: 'stun:' + account.getDetails()['STUN.server'],
});
}
setWebRtcConnection(new RTCPeerConnection({ iceServers }));
}
}, [account, webRtcConnection]);
if (!webRtcConnection || !webSocket) {
return <LoadingPage />;
}
return (
<WebRtcProvider webRtcConnection={webRtcConnection} webSocket={webSocket}>
{children}
</WebRtcProvider>
);
};
const WebRtcProvider = ({
children,
webRtcConnection,
webSocket,
}: WithChildren & {
webRtcConnection: RTCPeerConnection;
webSocket: IWebSocketContext;
}) => {
const { conversation, conversationId } = useContext(ConversationContext);
const [localStream, setLocalStream] = useState<MediaStream>();
const [remoteStreams, setRemoteStreams] = useState<readonly MediaStream[]>();
const [iceConnectionState, setIceConnectionState] = useState<RTCIceConnectionState | undefined>();
const [mediaDevices, setMediaDevices] = useState<Record<MediaDeviceKind, MediaDeviceInfo[]>>(
defaultWebRtcContext.mediaDevices
);
// TODO: This logic will have to change to support multiple people in a call
const contactUri = useMemo(() => conversation.getFirstMember().contact.getUri(), [conversation]);
const getMediaDevices = useCallback(async () => {
try {
const devices = await navigator.mediaDevices.enumerateDevices();
const newMediaDevices: Record<MediaDeviceKind, MediaDeviceInfo[]> = {
audioinput: [],
audiooutput: [],
videoinput: [],
};
for (const device of devices) {
newMediaDevices[device.kind].push(device);
}
return newMediaDevices;
} catch (e) {
throw new Error('Could not get media devices', { cause: e });
}
}, []);
useEffect(() => {
if (iceConnectionState !== 'connected' && iceConnectionState !== 'completed') {
return;
}
const updateMediaDevices = async () => {
try {
const newMediaDevices = await getMediaDevices();
setMediaDevices(newMediaDevices);
} catch (e) {
console.error('Could not update media devices:', e);
}
};
navigator.mediaDevices.addEventListener('devicechange', updateMediaDevices);
updateMediaDevices();
return () => {
navigator.mediaDevices.removeEventListener('devicechange', updateMediaDevices);
};
}, [getMediaDevices, iceConnectionState]);
const getUserMedia = useCallback(async () => {
const devices = await getMediaDevices();
const shouldGetAudio = devices.audioinput.length !== 0;
const shouldGetVideo = devices.videoinput.length !== 0;
if (!shouldGetAudio && !shouldGetVideo) {
return;
}
try {
const stream = await navigator.mediaDevices.getUserMedia({
audio: shouldGetAudio,
video: shouldGetVideo,
});
for (const track of stream.getTracks()) {
track.enabled = false;
webRtcConnection.addTrack(track, stream);
}
setLocalStream(stream);
} catch (e) {
throw new Error('Could not get media devices', { cause: e });
}
}, [webRtcConnection, getMediaDevices]);
const sendWebRtcOffer = useCallback(async () => {
const sdp = await webRtcConnection.createOffer({
offerToReceiveAudio: true,
offerToReceiveVideo: true,
});
const webRtcOffer: WebRtcSdp = {
contactId: contactUri,
conversationId: conversationId,
sdp,
};
await webRtcConnection.setLocalDescription(new RTCSessionDescription(sdp));
console.info('Sending WebRtcOffer', webRtcOffer);
webSocket.send(WebSocketMessageType.WebRtcOffer, webRtcOffer);
}, [webRtcConnection, webSocket, conversationId, contactUri]);
const sendWebRtcAnswer = useCallback(async () => {
const sdp = await webRtcConnection.createAnswer({
offerToReceiveAudio: true,
offerToReceiveVideo: true,
});
const webRtcAnswer: WebRtcSdp = {
contactId: contactUri,
conversationId: conversationId,
sdp,
};
await webRtcConnection.setLocalDescription(new RTCSessionDescription(sdp));
console.info('Sending WebRtcAnswer', webRtcAnswer);
webSocket.send(WebSocketMessageType.WebRtcAnswer, webRtcAnswer);
}, [contactUri, conversationId, webRtcConnection, webSocket]);
/* WebSocket Listeners */
useEffect(() => {
const webRtcOfferListener = async (data: WebRtcSdp) => {
console.info('Received event on WebRtcOffer', data);
if (data.conversationId !== conversationId) {
console.warn('Wrong incoming conversationId, ignoring action');
return;
}
await webRtcConnection.setRemoteDescription(new RTCSessionDescription(data.sdp));
await sendWebRtcAnswer();
};
const webRtcAnswerListener = async (data: WebRtcSdp) => {
console.info('Received event on WebRtcAnswer', data);
if (data.conversationId !== conversationId) {
console.warn('Wrong incoming conversationId, ignoring action');
return;
}
await webRtcConnection.setRemoteDescription(new RTCSessionDescription(data.sdp));
};
webSocket.bind(WebSocketMessageType.WebRtcOffer, webRtcOfferListener);
webSocket.bind(WebSocketMessageType.WebRtcAnswer, webRtcAnswerListener);
return () => {
webSocket.unbind(WebSocketMessageType.WebRtcOffer, webRtcOfferListener);
webSocket.unbind(WebSocketMessageType.WebRtcAnswer, webRtcAnswerListener);
};
}, [webSocket, webRtcConnection, sendWebRtcAnswer, conversationId]);
useEffect(() => {
const webRtcIceCandidateListener = async (data: WebRtcIceCandidate) => {
if (data.conversationId !== conversationId) {
console.warn('Wrong incoming conversationId, ignoring action');
return;
}
await webRtcConnection.addIceCandidate(data.candidate);
};
webSocket.bind(WebSocketMessageType.WebRtcIceCandidate, webRtcIceCandidateListener);
return () => {
webSocket.unbind(WebSocketMessageType.WebRtcIceCandidate, webRtcIceCandidateListener);
};
}, [webRtcConnection, webSocket, conversationId]);
/* WebRTC Listeners */
useEffect(() => {
const iceCandidateEventListener = (event: RTCPeerConnectionIceEvent) => {
if (event.candidate) {
const webRtcIceCandidate: WebRtcIceCandidate = {
contactId: contactUri,
conversationId: conversationId,
candidate: event.candidate,
};
webSocket.send(WebSocketMessageType.WebRtcIceCandidate, webRtcIceCandidate);
}
};
webRtcConnection.addEventListener('icecandidate', iceCandidateEventListener);
return () => {
webRtcConnection.removeEventListener('icecandidate', iceCandidateEventListener);
};
}, [webRtcConnection, webSocket, contactUri, conversationId]);
useEffect(() => {
const trackEventListener = (event: RTCTrackEvent) => {
console.info('Received WebRTC event on track', event);
setRemoteStreams(event.streams);
};
const iceConnectionStateChangeEventListener = (event: Event) => {
console.info(`Received WebRTC event on iceconnectionstatechange: ${webRtcConnection.iceConnectionState}`, event);
setIceConnectionState(webRtcConnection.iceConnectionState);
};
webRtcConnection.addEventListener('track', trackEventListener);
webRtcConnection.addEventListener('iceconnectionstatechange', iceConnectionStateChangeEventListener);
return () => {
webRtcConnection.removeEventListener('track', trackEventListener);
webRtcConnection.removeEventListener('iceconnectionstatechange', iceConnectionStateChangeEventListener);
};
}, [webRtcConnection]);
const closeConnection = useCallback(() => {
const localTracks = localStream?.getTracks();
if (localTracks) {
for (const track of localTracks) {
track.stop();
}
}
webRtcConnection.close();
}, [webRtcConnection, localStream]);
return (
<WebRtcContext.Provider
value={{
iceConnectionState,
mediaDevices,
localStream,
remoteStreams,
getUserMedia,
sendWebRtcOffer,
closeConnection,
}}
>
{children}
</WebRtcContext.Provider>
);
};