| /* |
| * 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 React, { createContext, useCallback, useRef } from 'react'; |
| import { connect, Socket } from 'socket.io-client'; |
| |
| import { WithChildren } from '../utils/utils'; |
| |
| /* |
| * TODO: This socket is temporary, it will be replaced by the real socket |
| * for communication with webrtc |
| * */ |
| const socket = connect('http://192.168.0.12:8080', { transports: ['websocket'] }); |
| |
| interface IWebRTCContext { |
| localVideoRef: React.RefObject<HTMLVideoElement> | null; |
| remoteVideoRef: React.RefObject<HTMLVideoElement> | null; |
| createWebRTCConnection: () => void; |
| sendWebRTCOffer: () => void; |
| sendWebRTCAnswer: (remoteSdp: RTCSessionDescriptionInit) => void; |
| handleWebRTCAnswer: (remoteSdp: RTCSessionDescriptionInit) => void; |
| addIceCandidate: (candidate: RTCIceCandidateInit) => void; |
| socket: Socket; |
| } |
| |
| const DefaultWebRTCContext: IWebRTCContext = { |
| localVideoRef: null, |
| remoteVideoRef: null, |
| createWebRTCConnection: () => {}, |
| sendWebRTCOffer: () => {}, |
| sendWebRTCAnswer: () => {}, |
| handleWebRTCAnswer: () => {}, |
| addIceCandidate: () => {}, |
| socket: socket, |
| }; |
| |
| export const WebRTCContext = createContext<IWebRTCContext>(DefaultWebRTCContext); |
| |
| export default ({ children }: WithChildren) => { |
| const localVideoRef = useRef<HTMLVideoElement>(null); |
| const remoteVideoRef = useRef<HTMLVideoElement>(null); |
| const webRTCConnectionRef = useRef<RTCPeerConnection>(); |
| |
| const createWebRTCConnection = useCallback(async () => { |
| //TODO use SFL iceServers |
| const iceConfig = { iceServers: [{ urls: 'stun:stun.l.google.com:19302' }] }; |
| webRTCConnectionRef.current = new RTCPeerConnection(iceConfig); |
| const localStream = await navigator.mediaDevices.getUserMedia({ |
| video: true, |
| audio: true, |
| }); |
| |
| if (localVideoRef.current) { |
| localVideoRef.current.srcObject = localStream; |
| } |
| |
| localStream.getTracks().forEach((track) => { |
| if (webRTCConnectionRef.current) { |
| webRTCConnectionRef.current.addTrack(track, localStream); |
| } |
| }); |
| webRTCConnectionRef.current.addEventListener('icecandidate', (event) => { |
| if (event.candidate && socket) { |
| console.log('webRTCConnection : onicecandidate'); |
| socket.emit('candidate', event.candidate); |
| } |
| }); |
| webRTCConnectionRef.current.addEventListener('track', async (event) => { |
| if (remoteVideoRef.current) { |
| remoteVideoRef.current.srcObject = event.streams[0]; |
| console.log('webRTCConnection : add remotetrack success'); |
| } |
| }); |
| }, [webRTCConnectionRef, localVideoRef, remoteVideoRef]); |
| |
| const sendWebRTCOffer = useCallback(async () => { |
| try { |
| if (webRTCConnectionRef.current && socket) { |
| const sdp = await webRTCConnectionRef.current.createOffer({ |
| offerToReceiveAudio: true, |
| offerToReceiveVideo: true, |
| }); |
| await webRTCConnectionRef.current.setLocalDescription(new RTCSessionDescription(sdp)); |
| socket.emit('offer', sdp); |
| } |
| } catch (e) { |
| console.error(e); |
| } |
| }, [webRTCConnectionRef]); |
| |
| const sendWebRTCAnswer = useCallback( |
| async (remoteSdp: RTCSessionDescriptionInit) => { |
| try { |
| if (webRTCConnectionRef.current && socket && remoteSdp) { |
| await webRTCConnectionRef.current.setRemoteDescription(new RTCSessionDescription(remoteSdp)); |
| const mySdp = await webRTCConnectionRef.current.createAnswer({ |
| offerToReceiveAudio: true, |
| offerToReceiveVideo: true, |
| }); |
| await webRTCConnectionRef.current.setLocalDescription(new RTCSessionDescription(mySdp)); |
| socket.emit('answer', mySdp); |
| } |
| } catch (e) { |
| console.error(e); |
| } |
| }, |
| [webRTCConnectionRef] |
| ); |
| |
| const handleWebRTCAnswer = useCallback( |
| async (remoteSdp: RTCSessionDescriptionInit) => { |
| try { |
| if (webRTCConnectionRef.current && remoteSdp) { |
| await webRTCConnectionRef.current.setRemoteDescription(new RTCSessionDescription(remoteSdp)); |
| } |
| } catch (e) { |
| console.error(e); |
| } |
| }, |
| [webRTCConnectionRef] |
| ); |
| |
| const addIceCandidate = useCallback( |
| async (candidate: RTCIceCandidateInit) => { |
| try { |
| if (webRTCConnectionRef.current) { |
| await webRTCConnectionRef.current.addIceCandidate(new RTCIceCandidate(candidate)); |
| } |
| } catch (e) { |
| console.error(e); |
| } |
| }, |
| [webRTCConnectionRef] |
| ); |
| |
| return ( |
| <WebRTCContext.Provider |
| value={{ |
| localVideoRef, |
| remoteVideoRef, |
| createWebRTCConnection, |
| sendWebRTCOffer, |
| sendWebRTCAnswer, |
| handleWebRTCAnswer, |
| addIceCandidate, |
| socket, |
| }} |
| > |
| {children} |
| </WebRTCContext.Provider> |
| ); |
| }; |