585 lines
24 KiB
TypeScript
585 lines
24 KiB
TypeScript
import { Call } from "@/app/components/Call/Call";
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import { useConsoleLogger } from "@/app/hooks/useConsoleLogger";
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import { createContext, useEffect, useRef, useState } from "react";
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import nacl from 'tweetnacl';
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import { useSender } from "../ProtocolProvider/useSender";
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import { PacketSignalPeer, SignalType } from "../ProtocolProvider/protocol/packets/packet.signal.peer";
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import { usePacket } from "../ProtocolProvider/usePacket";
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import { usePublicKey } from "../AccountProvider/usePublicKey";
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import { PacketWebRTC, WebRTCSignalType } from "../ProtocolProvider/protocol/packets/packet.webrtc";
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import { PacketIceServers } from "../ProtocolProvider/protocol/packets/packet.ice.servers";
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import { modals } from "@mantine/modals";
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import { Button, Flex, Text } from "@mantine/core";
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import { useSound } from "@/app/hooks/useSound";
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import useWindow from "@/app/hooks/useWindow";
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import { attachReceiverE2EE, attachSenderE2EE } from "./audioE2EE";
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import { useDeattachedSender } from "../DialogProvider/useDeattachedSender";
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import { AttachmentType } from "../ProtocolProvider/protocol/packets/packet.message";
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import { generateRandomKey } from "@/app/utils/utils";
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export interface CallContextValue {
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call: (callable: string) => void;
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close: () => void;
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activeCall: string;
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callState: CallState;
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muted: boolean;
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sound: boolean;
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setMuted: (muted: boolean) => void;
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setSound: (sound: boolean) => void;
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duration: number;
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setShowCallView: (show: boolean) => void;
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getKeyCast: () => string;
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accept: () => void;
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}
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export enum CallState {
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CONNECTING,
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KEY_EXCHANGE,
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/**
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* Финальная стадия сигналинга, на которой обе стороны обменялись ключами и теперь устанавливают защищенный канал связи для звонка,
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* через WebRTC, и готовятся к активному звонку.
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*/
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WEB_RTC_EXCHANGE,
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ACTIVE,
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ENDED,
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INCOMING
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}
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export enum CallRole {
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/**
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* Вызывающая сторона, которая инициирует звонок
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*/
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CALLER,
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/**
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* Принимающая сторона, которая отвечает на звонок и принимает его
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*/
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CALLEE
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}
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export const CallContext = createContext<CallContextValue | null>(null);
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export interface CallProviderProps {
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children: React.ReactNode;
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}
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export function CallProvider(props : CallProviderProps) {
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const [activeCall, setActiveCall] = useState<string>("");
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const [callState, setCallState] = useState<CallState>(CallState.ENDED);
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const [muted, setMutedState] = useState<boolean>(false);
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const [sound, setSoundState] = useState<boolean>(true);
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const durationIntervalRef = useRef<NodeJS.Timeout | null>(null);
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const [duration, setDuration] = useState<number>(0);
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const [showCallView, setShowCallView] = useState<boolean>(callState == CallState.INCOMING);
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const {info} = useConsoleLogger("CallProvider");
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const [sessionKeys, setSessionKeys] = useState<nacl.BoxKeyPair | null>(null);
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const send = useSender();
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const publicKey = usePublicKey();
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const peerConnectionRef = useRef<RTCPeerConnection | null>(null);
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const roomIdRef = useRef<string>("");
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const roleRef = useRef<CallRole | null>(null);
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const sharedSecretRef = useRef<string>("");
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const iceServersRef = useRef<RTCIceServer[]>([]);
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const remoteAudioRef = useRef<HTMLAudioElement | null>(null);
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const iceCandidatesBufferRef = useRef<RTCIceCandidate[]>([]);
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const mutedRef = useRef<boolean>(false);
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const soundRef = useRef<boolean>(true);
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const {sendMessage} = useDeattachedSender();
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const {playSound, stopSound, stopLoopSound} = useSound();
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const {setWindowPriority} = useWindow();
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useEffect(() => {
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if(callState == CallState.ACTIVE){
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stopLoopSound();
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stopSound();
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playSound("connected.mp3");
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setWindowPriority(false);
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durationIntervalRef.current = setInterval(() => {
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setDuration(prev => prev + 1);
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}, 1000);
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}
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}, [callState]);
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useEffect(() => {
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/**
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* Нам нужно получить ICE серверы для установки соединения из разных сетей
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* Получаем их от сервера
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*/
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let packet = new PacketIceServers();
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send(packet);
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return () => {
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stopSound();
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if (remoteAudioRef.current) {
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remoteAudioRef.current.pause();
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remoteAudioRef.current.srcObject = null;
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}
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peerConnectionRef.current?.close();
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peerConnectionRef.current = null;
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};
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}, []);
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usePacket(28, async (packet: PacketIceServers) => {
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let iceServers = packet.getIceServers();
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/**
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* ICE серверы получены, теперь нужно привести их к форматку клиента и добавить udp и tcp варианты
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*/
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let formattedIceServers: RTCIceServer[] = [];
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for(let i = 0; i < iceServers.length; i++){
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let server = iceServers[i];
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formattedIceServers.push({
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urls: "turn:" + server.url + "?transport=" + server.transport,
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username: server.username,
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credential: server.credential
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});
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}
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iceServersRef.current = formattedIceServers;
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info("Received ICE servers from server, count: " + formattedIceServers.length);
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}, []);
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usePacket(27, async (packet: PacketWebRTC) => {
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if(callState != CallState.WEB_RTC_EXCHANGE && callState != CallState.ACTIVE){
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/**
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* Нет активного звонка или мы не на стадии обмена WebRTC сигналами, игнорируем
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*/
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return;
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}
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const signalType = packet.getSignalType();
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if(signalType == WebRTCSignalType.ANSWER){
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/**
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* Другая сторона (сервер SFU) отправил нам SDP ответ на наш оффер
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*/
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const sdp = JSON.parse(packet.getSdpOrCandidate());
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await peerConnectionRef.current?.setRemoteDescription(new RTCSessionDescription(sdp));
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if(iceCandidatesBufferRef.current.length > 0){
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/**
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* У нас есть буферизированные ICE кандидаты, которые мы получили до установки удаленного описания, теперь мы можем их добавить в PeerConnection
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*/
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for(let i = 0; i < iceCandidatesBufferRef.current.length; i++){
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await peerConnectionRef.current?.addIceCandidate(iceCandidatesBufferRef.current[i]);
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}
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iceCandidatesBufferRef.current = [];
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}
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info("Received WebRTC answer and set remote description");
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return;
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}
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if(signalType == WebRTCSignalType.ICE_CANDIDATE){
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/**
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* Другая сторона отправила нам ICE кандидата для установления WebRTC соединения
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*/
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const candidate = JSON.parse(packet.getSdpOrCandidate());
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console.info(candidate);
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if(peerConnectionRef.current?.remoteDescription == null){
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/**
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* Удаленное описание еще не установлено, буферизуем кандидата, чтобы добавить его после установки удаленного описания
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*/
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iceCandidatesBufferRef.current.push(new RTCIceCandidate(candidate));
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info("Received WebRTC ICE candidate but remote description is not set yet, buffering candidate");
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return;
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}
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await peerConnectionRef.current?.addIceCandidate(new RTCIceCandidate(candidate));
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info("Received WebRTC ICE candidate and added to peer connection");
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return;
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}
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if(signalType == WebRTCSignalType.OFFER && peerConnectionRef.current){
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/**
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* SFU сервер отправил нам оффер, например при renegotiation, нам нужно его принять и
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* отправить ответ (ANSWER)
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*/
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const sdp = JSON.parse(packet.getSdpOrCandidate());
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await peerConnectionRef.current?.setRemoteDescription(new RTCSessionDescription(sdp));
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let answer = await peerConnectionRef.current?.createAnswer();
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await peerConnectionRef.current?.setLocalDescription(answer);
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let answerSignal = new PacketWebRTC();
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answerSignal.setSignalType(WebRTCSignalType.ANSWER);
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answerSignal.setSdpOrCandidate(JSON.stringify(answer));
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send(answerSignal);
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info("Received WebRTC offer, set remote description and sent answer");
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return;
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}
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}, [activeCall, sessionKeys, callState, roomIdRef]);
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usePacket(26, async (packet: PacketSignalPeer) => {
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const signalType = packet.getSignalType();
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if(signalType == SignalType.END_CALL_BECAUSE_BUSY) {
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openCallsModal("Line is busy, the user is currently on another call. Please try again later.");
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end();
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}
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if(signalType == SignalType.END_CALL_BECAUSE_PEER_DISCONNECTED) {
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openCallsModal("The connection with the user was lost. The call has ended.")
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end();
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}
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if(activeCall){
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/**
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* У нас уже есть активный звонок, игнорируем все сигналы, кроме сигналов от текущего звонка
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*/
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if(packet.getSrc() != activeCall && packet.getSrc() != publicKey){
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console.info("Received signal from " + packet.getSrc() + " but active call is with " + activeCall + ", ignoring");
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info("Received signal for another call, ignoring");
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return;
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}
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}
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if(signalType == SignalType.END_CALL){
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/**
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* Сбросили звонок
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*/
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end();
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return;
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}
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if(signalType == SignalType.CALL){
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/**
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* Нам поступает звонок
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*/
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setWindowPriority(true);
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playSound("ringtone.mp3", true);
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setActiveCall(packet.getSrc());
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setCallState(CallState.INCOMING);
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setShowCallView(true);
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}
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if(signalType == SignalType.KEY_EXCHANGE && roleRef.current == CallRole.CALLER){
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console.info("EXCHANGE SIGNAL RECEIVED, CALLER ROLE");
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/**
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* Другая сторона сгенерировала ключи для сессии и отправила нам публичную часть,
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* теперь мы можем создать общую секретную сессию для шифрования звонка
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*/
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const sharedPublic = packet.getSharedPublic();
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if(!sharedPublic){
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info("Received key exchange signal without shared public key");
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return;
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}
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const sessionKeys = generateSessionKeys();
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const computedSharedSecret = nacl.box.before(Buffer.from(sharedPublic, 'hex'), sessionKeys.secretKey);
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sharedSecretRef.current = Buffer.from(computedSharedSecret).toString('hex');
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info("Generated shared secret for call session: " + sharedSecretRef.current);
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/**
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* Нам нужно отправить свой публичный ключ другой стороне, чтобы она тоже могла создать общую секретную сессию
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*/
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const signalPacket = new PacketSignalPeer();
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signalPacket.setSrc(publicKey);
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signalPacket.setDst(packet.getSrc());
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signalPacket.setSignalType(SignalType.KEY_EXCHANGE);
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signalPacket.setSharedPublic(Buffer.from(sessionKeys.publicKey).toString('hex'));
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send(signalPacket);
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setCallState(CallState.WEB_RTC_EXCHANGE);
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/**
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* Создаем комнату на сервере SFU, комнату создает звонящий
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*/
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let webRtcSignal = new PacketSignalPeer();
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webRtcSignal.setSignalType(SignalType.CREATE_ROOM);
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webRtcSignal.setSrc(publicKey);
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webRtcSignal.setDst(packet.getSrc());
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send(webRtcSignal);
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}
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if(signalType == SignalType.KEY_EXCHANGE && roleRef.current == CallRole.CALLEE){
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console.info("EXCHANGE SIGNAL RECEIVED, CALLEE ROLE");
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/**
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* Мы отправили свою публичную часть ключа другой стороне,
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* теперь мы получили ее публичную часть и можем создать общую
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* секретную сессию для шифрования звонка
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*/
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const sharedPublic = packet.getSharedPublic();
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if(!sharedPublic){
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info("Received key exchange signal without shared public key");
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return;
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}
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if(!sessionKeys){
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info("Received key exchange signal but session keys are not generated");
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return;
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}
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const computedSharedSecret = nacl.box.before(Buffer.from(sharedPublic, 'hex'), sessionKeys.secretKey);
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sharedSecretRef.current = Buffer.from(computedSharedSecret).toString('hex');
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info("Generated shared secret for call session: " + sharedSecretRef.current);
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setCallState(CallState.WEB_RTC_EXCHANGE);
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}
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if(signalType == SignalType.CREATE_ROOM) {
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/**
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* Создана комната для обмена WebRTC потоками
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*/
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roomIdRef.current = packet.getRoomId();
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info("WebRTC room created with id: " + packet.getRoomId());
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/**
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* Нужно отправить свой SDP оффер другой стороне, чтобы установить WebRTC соединение
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*/
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peerConnectionRef.current = new RTCPeerConnection({
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iceServers: iceServersRef.current,
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// @ts-ignore
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encodedInsertableStreams: true
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});
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/**
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* Подписываемся на ICE кандидат
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*/
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peerConnectionRef.current.onicecandidate = (event) => {
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if(event.candidate){
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let candidateSignal = new PacketWebRTC();
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candidateSignal.setSignalType(WebRTCSignalType.ICE_CANDIDATE);
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candidateSignal.setSdpOrCandidate(JSON.stringify(event.candidate));
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send(candidateSignal);
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}
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}
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/**
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* Соединение установлено, можно начинать звонок, переходим в активное состояние звонка
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*/
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peerConnectionRef.current.onconnectionstatechange = () => {
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console.info("Peer connection state changed: " + peerConnectionRef.current?.connectionState);
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if(peerConnectionRef.current?.connectionState == "connected"){
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setCallState(CallState.ACTIVE);
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info("WebRTC connection established, call is active");
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}
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}
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peerConnectionRef.current.ontrack = async (event) => {
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try {
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await attachReceiverE2EE(event.receiver, Buffer.from(sharedSecretRef.current, "hex"));
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} catch (e) {
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console.error("attachReceiverE2EE failed:", e);
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}
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/**
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* При получении медиа-трека с другой стороны
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*/
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if(remoteAudioRef.current && event.streams[0]){
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console.info(event.streams);
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remoteAudioRef.current.srcObject = event.streams[0];
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remoteAudioRef.current.muted = !soundRef.current;
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void remoteAudioRef.current.play().catch((e) => {
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console.error("Failed to play remote audio:", e);
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});
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}
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}
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/**
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* Запрашиваем Аудио поток с микрофона и добавляем его в PeerConnection, чтобы другая сторона могла его получить и воспроизвести,
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* когда мы установим WebRTC соединение
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*/
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const localStream = await navigator.mediaDevices.getUserMedia({ audio: true });
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const audioTrack = localStream.getAudioTracks()[0];
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const tx = peerConnectionRef.current.addTransceiver(audioTrack, {
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direction: "sendrecv",
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streams: [localStream]
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});
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await attachSenderE2EE(tx.sender, Buffer.from(sharedSecretRef.current, "hex"));
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/**
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* Отправляем свой оффер другой стороне
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*/
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let offer = await peerConnectionRef.current.createOffer();
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await peerConnectionRef.current.setLocalDescription(offer);
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let offerSignal = new PacketWebRTC();
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offerSignal.setSignalType(WebRTCSignalType.OFFER);
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offerSignal.setSdpOrCandidate(JSON.stringify(offer));
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send(offerSignal);
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return;
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}
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}, [activeCall, sessionKeys, duration]);
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const openCallsModal = (text : string) => {
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modals.open({
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centered: true,
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children: (
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<>
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<Text size="sm">
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{text}
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</Text>
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<Flex align={'center'} justify={'flex-end'}>
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<Button color={'red'} variant={'subtle'} onClick={() => modals.closeAll()} mt="md">
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Close
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</Button>
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</Flex>
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</>
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),
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withCloseButton: false
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});
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}
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const generateSessionKeys = () => {
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const sessionKeys = nacl.box.keyPair();
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info("Generated keys for call session, len: " + sessionKeys.publicKey.length);
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setSessionKeys(sessionKeys);
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return sessionKeys;
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}
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const call = (dialog: string) => {
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if(callState == CallState.ACTIVE
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|| callState == CallState.CONNECTING
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|| callState == CallState.KEY_EXCHANGE
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|| callState == CallState.WEB_RTC_EXCHANGE){
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openCallsModal("You are already on a call, please end the current call before starting a new one.");
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return;
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}
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setWindowPriority(false);
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setActiveCall(dialog);
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setCallState(CallState.CONNECTING);
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setShowCallView(true);
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const signalPacket = new PacketSignalPeer();
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signalPacket.setSrc(publicKey);
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signalPacket.setDst(dialog);
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signalPacket.setSignalType(SignalType.CALL);
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send(signalPacket);
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roleRef.current = CallRole.CALLER;
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playSound("calling.mp3", true);
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}
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const close = () => {
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const packetSignal = new PacketSignalPeer();
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packetSignal.setSrc(publicKey);
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packetSignal.setDst(activeCall);
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packetSignal.setSignalType(SignalType.END_CALL);
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send(packetSignal);
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end();
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}
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const end = () => {
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stopLoopSound();
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stopSound();
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if (remoteAudioRef.current) {
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remoteAudioRef.current.pause();
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remoteAudioRef.current.srcObject = null;
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}
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generateCallAttachment();
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setDuration(0);
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durationIntervalRef.current && clearInterval(durationIntervalRef.current);
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setWindowPriority(false);
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playSound("end_call.mp3");
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peerConnectionRef.current?.close();
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peerConnectionRef.current = null;
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roomIdRef.current = "";
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mutedRef.current = false;
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soundRef.current = true;
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setActiveCall("");
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setCallState(CallState.ENDED);
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setShowCallView(false);
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setSessionKeys(null);
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setDuration(0);
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setMutedState(false);
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setSoundState(true);
|
||
roleRef.current = null;
|
||
}
|
||
|
||
/**
|
||
* Отправляет сообщение в диалог с звонящим с информацией о звонке
|
||
*/
|
||
const generateCallAttachment = () => {
|
||
if(roleRef.current != CallRole.CALLER){
|
||
/**
|
||
* Только звонящий отправляет информацию о звонке в виде вложения, чтобы ее можно было отобразить в UI диалога, например длительность звонка
|
||
*/
|
||
return;
|
||
}
|
||
sendMessage(activeCall, "", [{
|
||
id: generateRandomKey(16),
|
||
preview: duration.toString(),
|
||
type: AttachmentType.CALL,
|
||
transport: {
|
||
transport_server: "",
|
||
transport_tag: ""
|
||
},
|
||
encoding: {
|
||
encoded_for: "",
|
||
encoder: ""
|
||
},
|
||
blob: ""
|
||
}], true);
|
||
}
|
||
|
||
const accept = () => {
|
||
if(callState != CallState.INCOMING){
|
||
/**
|
||
* Нечего принимать
|
||
*/
|
||
return;
|
||
}
|
||
setWindowPriority(false);
|
||
stopLoopSound();
|
||
stopSound();
|
||
/**
|
||
* Звонок принят, генерируем ключи для сессии и отправляем их другой стороне для установления защищенного канала связи
|
||
*/
|
||
const keys = generateSessionKeys();
|
||
const signalPacket = new PacketSignalPeer();
|
||
signalPacket.setSrc(publicKey);
|
||
signalPacket.setDst(activeCall);
|
||
signalPacket.setSignalType(SignalType.KEY_EXCHANGE);
|
||
signalPacket.setSharedPublic(Buffer.from(keys.publicKey).toString('hex'));
|
||
send(signalPacket);
|
||
setCallState(CallState.KEY_EXCHANGE);
|
||
roleRef.current = CallRole.CALLEE;
|
||
}
|
||
|
||
/**
|
||
* Получает слепок ключа для отображения в UI
|
||
* чтобы не показывать настоящий ключ
|
||
* @returns
|
||
*/
|
||
const getKeyCast = () => {
|
||
if(!sharedSecretRef.current){
|
||
return "";
|
||
}
|
||
return sharedSecretRef.current;
|
||
}
|
||
|
||
|
||
const setMuted = (nextMuted: boolean) => {
|
||
if (mutedRef.current === nextMuted) {
|
||
return;
|
||
}
|
||
|
||
mutedRef.current = nextMuted;
|
||
playSound(nextMuted ? "micro_enable.mp3" : "micro_disable.mp3");
|
||
|
||
if(peerConnectionRef.current){
|
||
peerConnectionRef.current.getSenders().forEach(sender => {
|
||
if(sender.track?.kind == "audio"){
|
||
sender.track.enabled = !nextMuted;
|
||
}
|
||
});
|
||
}
|
||
|
||
setMutedState(nextMuted);
|
||
}
|
||
|
||
const setSound = (nextSound: boolean) => {
|
||
if (soundRef.current === nextSound) {
|
||
return;
|
||
}
|
||
|
||
soundRef.current = nextSound;
|
||
playSound(nextSound ? "sound_enable.mp3" : "sound_disable.mp3");
|
||
|
||
if(remoteAudioRef.current){
|
||
remoteAudioRef.current.muted = !nextSound;
|
||
if (nextSound) {
|
||
void remoteAudioRef.current.play().catch((e) => {
|
||
console.error("Failed to resume remote audio:", e);
|
||
});
|
||
}
|
||
}
|
||
|
||
setSoundState(nextSound);
|
||
}
|
||
|
||
const context = {
|
||
call,
|
||
close,
|
||
activeCall,
|
||
callState,
|
||
muted,
|
||
sound,
|
||
setMuted,
|
||
setSound,
|
||
duration,
|
||
setShowCallView,
|
||
getKeyCast,
|
||
accept
|
||
};
|
||
|
||
return (
|
||
<CallContext.Provider value={context}>
|
||
{props.children}
|
||
<audio ref={remoteAudioRef} autoPlay playsInline style={{ display: 'none' }} />
|
||
{showCallView && <Call context={context}></Call>}
|
||
</CallContext.Provider>
|
||
)
|
||
} |