feat: Add TURN server support
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+139
@@ -0,0 +1,139 @@
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import {
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Injectable,
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inject,
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computed,
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type Signal
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} from '@angular/core';
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import { Store } from '@ngrx/store';
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import { selectCurrentUser, selectOnlineUsers } from '../../../../store/users/users.selectors';
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import { toSignal } from '@angular/core/rxjs-interop';
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import { RealtimeSessionFacade } from '../../../../core/realtime';
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import type { User } from '../../../../shared-kernel';
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/**
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* Connectivity health status for a single peer in voice.
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*/
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export interface PeerConnectivityHealth {
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peerId: string;
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/** Number of voice peers this peer can send/receive audio to/from. */
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connectedPeerCount: number;
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/** Total peers expected in voice. */
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totalVoicePeers: number;
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/** true when this peer has the fewest connections -> warning target. */
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hasDesync: boolean;
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}
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/**
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* Tracks per-peer voice connectivity health by comparing the number
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* of connected audio streams each peer has. Peers with fewest
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* bidirectional audio connections are flagged.
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*
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* Uses peer latency data as proxy for healthy bidirectional connection.
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*/
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@Injectable({ providedIn: 'root' })
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export class VoiceConnectivityHealthService {
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readonly currentUser: Signal<User | null | undefined>;
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readonly onlineUsers: Signal<User[]>;
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readonly desyncPeerIds: Signal<ReadonlySet<string>>;
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readonly localUserHasDesync: Signal<boolean>;
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private readonly webrtc = inject(RealtimeSessionFacade);
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constructor() {
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const store = inject(Store);
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this.currentUser = toSignal(store.select(selectCurrentUser));
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this.onlineUsers = toSignal(store.select(selectOnlineUsers), { initialValue: [] });
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/**
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* Map of peerId -> true for peers that have connectivity issues.
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* A peer is flagged when it has fewer healthy connections than the
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* majority of users in the same voice channel.
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*/
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this.desyncPeerIds = computed<ReadonlySet<string>>(() => {
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const me = this.currentUser();
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const myVoice = me?.voiceState;
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if (!myVoice?.isConnected || !myVoice.roomId || !myVoice.serverId) {
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return new Set<string>();
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}
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// Find all users in same voice room
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const voiceUsers = this.onlineUsers().filter(
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(user) =>
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user.voiceState?.isConnected
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&& user.voiceState.roomId === myVoice.roomId
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&& user.voiceState.serverId === myVoice.serverId
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);
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if (voiceUsers.length < 2) {
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return new Set<string>();
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}
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// Use peer latencies as proxy. A peer we can ping has a working
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// data-channel (= working RTCPeerConnection). Peers without latency
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// measurements are considered unreachable.
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const connectedPeers = this.webrtc.connectedPeers();
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const connectedSet = new Set(connectedPeers);
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const myKey = me?.oderId || me?.id;
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if (!myKey) {
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return new Set<string>();
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}
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// Count how many voice peers each voice user is connected to (from
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// the local perspective). We can only see our own connections - but
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// if WE can't reach peer X while we CAN reach peers Y and Z, peer X
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// is the one with issues.
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const unreachableFromUs = new Set<string>();
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for (const user of voiceUsers) {
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const key = user.oderId || user.id;
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if (key === myKey) {
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continue;
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}
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const hasConnection = connectedSet.has(key)
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|| connectedSet.has(user.id)
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|| connectedSet.has(user.oderId ?? '');
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if (!hasConnection) {
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unreachableFromUs.add(key);
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}
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}
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// If we can reach everyone, no desync
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if (unreachableFromUs.size === 0) {
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return new Set<string>();
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}
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// If we can't reach ANYONE, the problem is likely on our end
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const reachableCount = voiceUsers.length - 1 - unreachableFromUs.size;
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if (reachableCount === 0 && voiceUsers.length > 2) {
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// Everyone unreachable from us -> WE are the problem
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return new Set([myKey]);
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}
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return unreachableFromUs;
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});
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/**
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* Whether the LOCAL user is the one with connectivity issues.
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*/
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this.localUserHasDesync = computed(() => {
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const me = this.currentUser();
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const myKey = me?.oderId || me?.id;
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return !!myKey && this.desyncPeerIds().has(myKey);
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});
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}
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/**
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* Check if a specific peer has a desync warning.
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*/
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hasPeerDesync(peerKey: string): boolean {
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return this.desyncPeerIds().has(peerKey);
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}
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}
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+1
-1
@@ -209,7 +209,7 @@ export class VoicePlaybackService {
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* ↓
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* muted <audio> element (Chrome workaround - primes the stream)
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* ↓
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* MediaStreamSource → GainNode → MediaStreamDestination → output <audio>
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* MediaStreamSource -> GainNode -> MediaStreamDestination -> output <audio>
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*/
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private createPipeline(peerId: string, stream: MediaStream): void {
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// Chromium/Electron needs a muted <audio> element before Web Audio can read the stream.
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