@rapidmx/meet-plugin 0.13.1 → 0.14.0

package/dist/apps/shared/relay/RelayClient.d.ts

@@ -9,7 +9,12 @@ export declare const MAX_NEGOTIATED_MESSAGE_BYTES: number;
 /** The most senders the server lets one socket `want`. */
 export declare const MAX_WANTED_PEERS = 32;
 /** A media send is dropped, not queued, while more than this many bytes are still waiting to leave - the
- * connection is slower than the media, and queueing would only add latency to the frames that are still fresh. */
+ * connection is slower than the media, and queueing would only add latency to the frames that are still fresh.
+ * 64 KiB is a little under 1.4 seconds of `VideoSender.VIDEO_BITRATE` (350 kbps) plus `AudioSender.AUDIO_BITRATE`
+ * (24 kbps) combined (≈46.75 KB/s) - tighter than the server's own `RelayHub.RELAY_WS_MAX_BACKPRESSURE_BYTES`
+ * (128 KiB) on purpose, so a sender that is itself the slow end of the connection starts dropping its own stale
+ * frames before the server's larger budget would. This was previously 256 KiB (≈5.6 seconds), which defeated the
+ * "drop, don't queue" intent on exactly the constrained uplink that routes a call to this fallback tier at all. */
 export declare const MAX_BUFFERED_BYTES: number;
 /** Reconnect delay: about a second after the first failure, doubling to a ceiling, jittered so a server restart
  * doesn't bring every client back in the same instant. */

package/dist/apps/shared/relay/RelayClient.js

@@ -32,8 +32,13 @@ export const MAX_NEGOTIATED_MESSAGE_BYTES = 1024 * 1024;
 /** The most senders the server lets one socket `want`. */
 export const MAX_WANTED_PEERS = 32;
 /** A media send is dropped, not queued, while more than this many bytes are still waiting to leave - the
- * connection is slower than the media, and queueing would only add latency to the frames that are still fresh. */
-export const MAX_BUFFERED_BYTES = 256 * 1024;
+ * connection is slower than the media, and queueing would only add latency to the frames that are still fresh.
+ * 64 KiB is a little under 1.4 seconds of `VideoSender.VIDEO_BITRATE` (350 kbps) plus `AudioSender.AUDIO_BITRATE`
+ * (24 kbps) combined (≈46.75 KB/s) - tighter than the server's own `RelayHub.RELAY_WS_MAX_BACKPRESSURE_BYTES`
+ * (128 KiB) on purpose, so a sender that is itself the slow end of the connection starts dropping its own stale
+ * frames before the server's larger budget would. This was previously 256 KiB (≈5.6 seconds), which defeated the
+ * "drop, don't queue" intent on exactly the constrained uplink that routes a call to this fallback tier at all. */
+export const MAX_BUFFERED_BYTES = 64 * 1024;
 /** Reconnect delay: about a second after the first failure, doubling to a ceiling, jittered so a server restart
  * doesn't bring every client back in the same instant. */
 export const RELAY_BACKOFF_BASE_MS = 1000;

package/dist/apps/shared/webrtc/MeshConnectionManager.d.ts

@@ -99,9 +99,37 @@
  * unavailable (turned off by the operator, or a browser without WebCodecs) the pair is marked `"failed"` and stays
  * in the roster, rather than the participant vanishing with no explanation as it used to.
  *
- * A pair never moves back up: once on the relay it stays there for the rest of the call.
+ * A pair never moves back up: once on the relay it stays there for the rest of the call - except that the whole
+ * waterfall is itself only what `"auto"` mode does; see the next section.
+ *
+ * ## Forcing a transport (`setTransportMode()`)
+ *
+ * `TransportMode` (`types.ts`) lets a participant override the waterfall above for their own tab: `"p2p"` offers
+ * ICE no TURN servers at all and never falls back past a failed direct attempt; `"relay"` sets
+ * `RTCPeerConnectionLike`'s `iceTransportPolicy: "relay"` (still real WebRTC, just never direct) and likewise never
+ * falls back further; `"websocket"` skips WebRTC negotiation entirely and jumps straight to the server relay for
+ * every peer, via the very same `fallBack()` the `"auto"` waterfall uses once it gives up (same cooperative
+ * `relay-fallback` notification, same "no relay available -> `failed`" handling - `"websocket"` mode just calls it
+ * immediately instead of after a timeout). `"p2p"`/`"relay"` skip the waterfall in the other direction:
+ * `giveUpOnWebRTC()` is what `createPeer()`'s connect timeout and `handleConnectionState()`'s `disconnected`/
+ * `failed` branches call instead of `fallBack()` directly, and it marks the pair `"failed"` outright in forced
+ * `"p2p"`/`"relay"` mode rather than degrading to the relay - a participant who explicitly chose "peer-to-peer
+ * only" or "TURN relay only" to troubleshoot a connection does not want it quietly becoming something else.
+ *
+ * Switching modes mid-call (not just at join) rebuilds every current peer connection under the new policy
+ * (`restartPeer()`): the old `RTCPeerConnection` is closed and a fresh one created in its place, exactly as
+ * `createPeer()` builds one for a brand-new peer, but keeping the existing roster entry (name/audioOn/videoOn/
+ * handRaised) and emitting `participant-updated` rather than a `participant-left`/`participant-joined` pair - from
+ * the UI's perspective this looks like the existing `"connecting"` state any reconnect already uses, not someone
+ * leaving and rejoining. This can't be done unilaterally: a participant cannot rebuild their own half of a pair
+ * without the other side rebuilding its matching half at (almost) the same moment, or the two ends negotiate
+ * against two different objects that have no idea about each other. `setTransportMode()` therefore also broadcasts
+ * `restart-connection` (no payload beyond the envelope - `types.ts`'s `SignalMessage` doc comment), and every
+ * recipient calls `restartPeer()` for just that one sender's connection in response - each side always rebuilds
+ * using whatever mode *it* currently has configured, never the sender's, since this is a personal, per-tab choice,
+ * not something a meeting agrees on together.
  */
-import { type MeshEvent, type MeshParticipant, type ParticipantState, type RTCPeerConnectionFactory, type RelayTransportLike, type SignalingChannel } from "./types.js";
+import { type MeshEvent, type MeshParticipant, type ParticipantState, type RTCPeerConnectionFactory, type RelayTransportLike, type SignalingChannel, type TransportMode } from "./types.js";
 /** The lexicographically smaller uid is always the offerer for that pair - see this module's doc comment. */
 export declare function isOfferer(selfId: string, peerId: string): boolean;
 /** How long a connection may stay in `connecting` before the pair gives up on WebRTC. ICE with an unreachable TURN
@@ -143,6 +171,8 @@ export interface MeshConnectionManagerOptions {
     disconnectedGraceMs?: number;
     /** Defaults to `DEFAULT_DIAGNOSTICS_POLL_MS`. */
     diagnosticsPollMs?: number;
+    /** Defaults to `"auto"` - see this module's doc comment on `setTransportMode()`. */
+    transportMode?: TransportMode;
 }
 export declare class MeshConnectionManager {
     private readonly options;
@@ -155,10 +185,12 @@ export declare class MeshConnectionManager {
     private started;
     private stopped;
     private currentPresenterUid;
+    private currentTransportMode;
     private readonly selfId;
     constructor(options: MeshConnectionManagerOptions);
     get participants(): MeshParticipant[];
     get presenterUid(): string | undefined;
+    get transportMode(): TransportMode;
     onEvent(listener: (event: MeshEvent) => void): () => void;
     /** Subscribes to the channel and announces this participant. Idempotent - a second call is a no-op. */
     start(): void;
@@ -195,6 +227,10 @@ export declare class MeshConnectionManager {
      * there is only ever one sender; the caller (`_CallView.tsx`) is responsible for only offering this to the
      * host. */
     setTalkingStick(active: boolean, holder?: string): void;
+    /** Switches this tab's own transport policy - see this module's doc comment on `TransportMode`. A no-op for the
+     * mode it already has. Rebuilds every current peer connection under the new policy and, once `start()` has
+     * run, tells every peer to rebuild their matching half too (`restart-connection`) - see `restartPeer()`. */
+    setTransportMode(mode: TransportMode): void;
     private sendHello;
     private send;
     private emit;
@@ -207,6 +243,11 @@ export declare class MeshConnectionManager {
     private updatePeer;
     private handleBye;
     private createPeer;
+    /** The `RTCPeerConnection` config for a peer created right now - `this.options.iceServers` as given for `"auto"`
+     * and `"websocket"` (the latter never actually uses them - see `handleHello()`/`handleOffer()`), STUN-only (no
+     * TURN offered to ICE at all) for `"p2p"`, and the same servers with `iceTransportPolicy: "relay"` for
+     * `"relay"` - see this module's doc comment on `TransportMode`. */
+    private pcConfig;
     /** Reacts to the peer connection's state: notes which path a connected pair got, and gives up on WebRTC for a
      * pair that failed or stays disconnected - see `fallBack()`. Ignores a pair that already left WebRTC. */
     private handleConnectionState;
@@ -220,6 +261,12 @@ export declare class MeshConnectionManager {
     private pollDiagnostics;
     private setTransport;
     private clearTimers;
+    /** What a connect timeout or a `"disconnected"`/`"failed"` connection state means for `peer`: `fallBack()` to
+     * the relay for `"auto"` mode (unchanged from before `TransportMode` existed), but `"failed"` outright for a
+     * forced `"p2p"`/`"relay"` mode - see this module's doc comment on why those two never degrade further.
+     * `"websocket"` mode never reaches here at all: `handleHello()`/`handleOffer()` call `fallBack()` immediately
+     * on peer creation, well before a connect timeout could ever fire. */
+    private giveUpOnWebRTC;
     /**
      * Gives up on WebRTC for `peer` and moves the pair to the WebSocket relay (the third path - see this module's doc
      * comment), or marks it `"failed"` when there is no relay to move to. `notify` tells the other side, which may not
@@ -229,6 +276,12 @@ export declare class MeshConnectionManager {
     private handleRelayFallback;
     /** The relay only encodes and uploads the local media while at least one pair is actually using it. */
     private syncRelaySending;
+    /** Rebuilds `uid`'s connection from scratch under the current `TransportMode` - see this module's doc comment
+     * on `setTransportMode()`. A no-op if `uid` isn't a known peer (already left). Keeps the roster entry (name and
+     * announced state) and emits `participant-updated`, not a `participant-left`/`participant-joined` pair - from
+     * the UI's perspective this is the same `"connecting"` state any other reconnect already shows, not someone
+     * leaving and rejoining. */
+    private restartPeer;
     private initiateOffer;
     private handleOffer;
     private handleAnswer;

package/dist/apps/shared/webrtc/MeshConnectionManager.js

@@ -103,7 +103,35 @@
  * unavailable (turned off by the operator, or a browser without WebCodecs) the pair is marked `"failed"` and stays
  * in the roster, rather than the participant vanishing with no explanation as it used to.
  *
- * A pair never moves back up: once on the relay it stays there for the rest of the call.
+ * A pair never moves back up: once on the relay it stays there for the rest of the call - except that the whole
+ * waterfall is itself only what `"auto"` mode does; see the next section.
+ *
+ * ## Forcing a transport (`setTransportMode()`)
+ *
+ * `TransportMode` (`types.ts`) lets a participant override the waterfall above for their own tab: `"p2p"` offers
+ * ICE no TURN servers at all and never falls back past a failed direct attempt; `"relay"` sets
+ * `RTCPeerConnectionLike`'s `iceTransportPolicy: "relay"` (still real WebRTC, just never direct) and likewise never
+ * falls back further; `"websocket"` skips WebRTC negotiation entirely and jumps straight to the server relay for
+ * every peer, via the very same `fallBack()` the `"auto"` waterfall uses once it gives up (same cooperative
+ * `relay-fallback` notification, same "no relay available -> `failed`" handling - `"websocket"` mode just calls it
+ * immediately instead of after a timeout). `"p2p"`/`"relay"` skip the waterfall in the other direction:
+ * `giveUpOnWebRTC()` is what `createPeer()`'s connect timeout and `handleConnectionState()`'s `disconnected`/
+ * `failed` branches call instead of `fallBack()` directly, and it marks the pair `"failed"` outright in forced
+ * `"p2p"`/`"relay"` mode rather than degrading to the relay - a participant who explicitly chose "peer-to-peer
+ * only" or "TURN relay only" to troubleshoot a connection does not want it quietly becoming something else.
+ *
+ * Switching modes mid-call (not just at join) rebuilds every current peer connection under the new policy
+ * (`restartPeer()`): the old `RTCPeerConnection` is closed and a fresh one created in its place, exactly as
+ * `createPeer()` builds one for a brand-new peer, but keeping the existing roster entry (name/audioOn/videoOn/
+ * handRaised) and emitting `participant-updated` rather than a `participant-left`/`participant-joined` pair - from
+ * the UI's perspective this looks like the existing `"connecting"` state any reconnect already uses, not someone
+ * leaving and rejoining. This can't be done unilaterally: a participant cannot rebuild their own half of a pair
+ * without the other side rebuilding its matching half at (almost) the same moment, or the two ends negotiate
+ * against two different objects that have no idea about each other. `setTransportMode()` therefore also broadcasts
+ * `restart-connection` (no payload beyond the envelope - `types.ts`'s `SignalMessage` doc comment), and every
+ * recipient calls `restartPeer()` for just that one sender's connection in response - each side always rebuilds
+ * using whatever mode *it* currently has configured, never the sender's, since this is a personal, per-tab choice,
+ * not something a meeting agrees on together.
  */
 import { REACTION_EMOJIS, } from "./types.js";
 /** The lexicographically smaller uid is always the offerer for that pair - see this module's doc comment. */
@@ -133,6 +161,7 @@ export class MeshConnectionManager {
         this.started = false;
         this.stopped = false;
         this.selfId = options.peerId ?? options.selfUid;
+        this.currentTransportMode = options.transportMode ?? "auto";
         this.localTracks = { audio: options.localAudioTrack ?? null, video: options.localVideoTrack ?? null };
         this.localState = {
             audioOn: !!options.localAudioTrack,
@@ -149,6 +178,9 @@ export class MeshConnectionManager {
     get presenterUid() {
         return this.currentPresenterUid;
     }
+    get transportMode() {
+        return this.currentTransportMode;
+    }
     onEvent(listener) {
         this.listeners.add(listener);
         return () => this.listeners.delete(listener);
@@ -257,6 +289,21 @@ export class MeshConnectionManager {
         this.send({ kind: "talking-stick", active, holder });
         this.emit({ type: "talking-stick-changed", active, holder });
     }
+    /** Switches this tab's own transport policy - see this module's doc comment on `TransportMode`. A no-op for the
+     * mode it already has. Rebuilds every current peer connection under the new policy and, once `start()` has
+     * run, tells every peer to rebuild their matching half too (`restart-connection`) - see `restartPeer()`. */
+    setTransportMode(mode) {
+        if (this.currentTransportMode === mode) {
+            return;
+        }
+        this.currentTransportMode = mode;
+        if (this.started && !this.stopped) {
+            this.send({ kind: "restart-connection" });
+        }
+        for (const uid of [...this.peers.keys()]) {
+            this.restartPeer(uid);
+        }
+    }
     sendHello() {
         this.send({ kind: "hello", name: this.options.selfName, state: this.localState });
     }
@@ -323,6 +370,9 @@ export class MeshConnectionManager {
             case "talking-stick":
                 this.emit({ type: "talking-stick-changed", active: message.active === true, holder: typeof message.holder === "string" ? message.holder : undefined });
                 return;
+            case "restart-connection":
+                this.restartPeer(message.from);
+                return;
         }
     }
     handleHello(message) {
@@ -338,6 +388,11 @@ export class MeshConnectionManager {
         // Let a newcomer who couldn't have seen our own original `hello` learn about us too - see this module's
         // doc comment on roster discovery.
         this.sendHello();
+        if (this.currentTransportMode === "websocket") {
+            // Forced relay-only: don't negotiate WebRTC at all - see this module's doc comment on `setTransportMode()`.
+            this.fallBack(peer, true);
+            return;
+        }
         if (offerer) {
             void this.initiateOffer(peer);
         }
@@ -395,7 +450,7 @@ export class MeshConnectionManager {
         }
     }
     createPeer(uid, name, state, offerer) {
-        const pc = this.options.createPeerConnection({ iceServers: this.options.iceServers });
+        const pc = this.options.createPeerConnection(this.pcConfig());
         const peer = {
             uid,
             name,
@@ -430,9 +485,23 @@ export class MeshConnectionManager {
         };
         pc.onconnectionstatechange = () => this.handleConnectionState(peer);
         // Cleared the moment the connection comes up, so this only ever fires for a pair still waiting on WebRTC.
-        peer.connectTimer = setTimeout(() => this.fallBack(peer), this.options.connectTimeoutMs ?? DEFAULT_CONNECT_TIMEOUT_MS);
+        peer.connectTimer = setTimeout(() => this.giveUpOnWebRTC(peer), this.options.connectTimeoutMs ?? DEFAULT_CONNECT_TIMEOUT_MS);
         return peer;
     }
+    /** The `RTCPeerConnection` config for a peer created right now - `this.options.iceServers` as given for `"auto"`
+     * and `"websocket"` (the latter never actually uses them - see `handleHello()`/`handleOffer()`), STUN-only (no
+     * TURN offered to ICE at all) for `"p2p"`, and the same servers with `iceTransportPolicy: "relay"` for
+     * `"relay"` - see this module's doc comment on `TransportMode`. */
+    pcConfig() {
+        switch (this.currentTransportMode) {
+            case "p2p":
+                return { iceServers: stunOnlyServers(this.options.iceServers) };
+            case "relay":
+                return { iceServers: this.options.iceServers, iceTransportPolicy: "relay" };
+            default:
+                return { iceServers: this.options.iceServers };
+        }
+    }
     /** Reacts to the peer connection's state: notes which path a connected pair got, and gives up on WebRTC for a
      * pair that failed or stays disconnected - see `fallBack()`. Ignores a pair that already left WebRTC. */
     handleConnectionState(peer) {
@@ -446,10 +515,10 @@ export class MeshConnectionManager {
                 return;
             case "disconnected":
                 // ICE often recovers from this on its own, so it is only given up on after a grace period.
-                peer.disconnectTimer ?? (peer.disconnectTimer = setTimeout(() => this.fallBack(peer), this.options.disconnectedGraceMs ?? DEFAULT_DISCONNECTED_GRACE_MS));
+                peer.disconnectTimer ?? (peer.disconnectTimer = setTimeout(() => this.giveUpOnWebRTC(peer), this.options.disconnectedGraceMs ?? DEFAULT_DISCONNECTED_GRACE_MS));
                 return;
             case "failed":
-                this.fallBack(peer);
+                this.giveUpOnWebRTC(peer);
                 return;
             case "closed":
                 this.handleBye(peer.uid);
@@ -513,6 +582,31 @@ export class MeshConnectionManager {
         peer.connectTimer = undefined;
         peer.disconnectTimer = undefined;
     }
+    /** What a connect timeout or a `"disconnected"`/`"failed"` connection state means for `peer`: `fallBack()` to
+     * the relay for `"auto"` mode (unchanged from before `TransportMode` existed), but `"failed"` outright for a
+     * forced `"p2p"`/`"relay"` mode - see this module's doc comment on why those two never degrade further.
+     * `"websocket"` mode never reaches here at all: `handleHello()`/`handleOffer()` call `fallBack()` immediately
+     * on peer creation, well before a connect timeout could ever fire. */
+    giveUpOnWebRTC(peer) {
+        if (this.currentTransportMode !== "p2p" && this.currentTransportMode !== "relay") {
+            this.fallBack(peer);
+            return;
+        }
+        /* v8 ignore if -- unreachable via real usage: unlike `fallBack()` (also callable directly from
+           `handleRelayFallback()`, which is how its own identical-looking guard is actually exercised),
+           `giveUpOnWebRTC()`'s only callers are `createPeer()`'s connect timer and `handleConnectionState()`'s
+           disconnect timer/`"failed"` case - and every transition that could make either side of this true
+           (`fallBack()` moving a peer to `"websocket"`, `restartPeer()` replacing the map entry) clears this
+           peer's pending timers first, so a stale timer referencing this exact peer object never fires. Kept for
+           the same reason `fallBack()`'s guard is - a future caller that doesn't pre-clear timers should fail
+           safe, not corrupt an already-superseded peer. */
+        if (peer.transport === "websocket" || this.peers.get(peer.uid) !== peer) {
+            return;
+        }
+        this.clearTimers(peer);
+        this.stopDiagnosticsPolling(peer);
+        this.setTransport(peer, "failed");
+    }
     /**
      * Gives up on WebRTC for `peer` and moves the pair to the WebSocket relay (the third path - see this module's doc
      * comment), or marks it `"failed"` when there is no relay to move to. `notify` tells the other side, which may not
@@ -556,6 +650,40 @@ export class MeshConnectionManager {
     syncRelaySending() {
         this.options.relay?.setSending([...this.peers.values()].some((peer) => peer.transport === "websocket"));
     }
+    /** Rebuilds `uid`'s connection from scratch under the current `TransportMode` - see this module's doc comment
+     * on `setTransportMode()`. A no-op if `uid` isn't a known peer (already left). Keeps the roster entry (name and
+     * announced state) and emits `participant-updated`, not a `participant-left`/`participant-joined` pair - from
+     * the UI's perspective this is the same `"connecting"` state any other reconnect already shows, not someone
+     * leaving and rejoining. */
+    restartPeer(uid) {
+        const old = this.peers.get(uid);
+        if (!old) {
+            return;
+        }
+        this.clearTimers(old);
+        this.stopDiagnosticsPolling(old);
+        // Stop listening to the dying connection before closing it - same ordering `fallBack()` uses, and for the
+        // same reason: closing it must never be mistaken for the peer leaving.
+        old.pc.onconnectionstatechange = null;
+        old.pc.onicecandidate = null;
+        old.pc.ontrack = null;
+        old.pc.close();
+        if (old.transport === "websocket") {
+            this.options.relay?.stopReceivingFrom(uid);
+        }
+        const offerer = isOfferer(this.selfId, uid);
+        const fresh = this.createPeer(uid, old.name, { audioOn: old.audioOn, videoOn: old.videoOn, handRaised: old.handRaised }, offerer);
+        if (this.currentTransportMode === "websocket") {
+            this.fallBack(fresh, true);
+            return;
+        }
+        // In case `old` was the relay's only remaining user - `fresh` starts on WebRTC, not the relay.
+        this.syncRelaySending();
+        this.emit({ type: "participant-updated", participant: toParticipant(fresh) });
+        if (offerer) {
+            void this.initiateOffer(fresh);
+        }
+    }
     async initiateOffer(peer) {
         const offer = await peer.pc.createOffer();
         await peer.pc.setLocalDescription(offer);
@@ -570,6 +698,17 @@ export class MeshConnectionManager {
         if (!peer) {
             peer = this.createPeer(message.from, message.from, undefined, false);
         }
+        if (isNew) {
+            this.emit({ type: "participant-joined", participant: toParticipant(peer) });
+            this.sendHello();
+        }
+        if (this.currentTransportMode === "websocket") {
+            // Forced relay-only: don't bother negotiating this offer at all - see this module's doc comment on
+            // `setTransportMode()`. Also tells the offerer (`relay-fallback`), so they don't wait out their own
+            // connect timeout for an answer that was never coming.
+            this.fallBack(peer, true);
+            return;
+        }
         await peer.pc.setRemoteDescription(message.sdp);
         this.flushPendingCandidates(peer);
         if (!peer.senders.audio && !peer.senders.video) {
@@ -582,10 +721,6 @@ export class MeshConnectionManager {
                 }
             }
         }
-        if (isNew) {
-            this.emit({ type: "participant-joined", participant: toParticipant(peer) });
-            this.sendHello();
-        }
         const answer = await peer.pc.createAnswer();
         await peer.pc.setLocalDescription(answer);
         this.send({ kind: "answer", to: peer.uid, sdp: answer });
@@ -681,3 +816,12 @@ function sanitizeState(state) {
 function isReactionEmoji(emoji) {
     return REACTION_EMOJIS.includes(emoji);
 }
+/** `servers` with every entry that names so much as one `turn:`/`turns:` url dropped - what `"p2p"` mode gives ICE,
+ * so it has no TURN relay candidate to ever produce. A `stun:`/`stuns:`-only entry (the normal shape this app's own
+ * `buildIceServers()` produces - one STUN entry, one TURN entry) is kept as-is. */
+function stunOnlyServers(servers) {
+    return servers.filter((server) => {
+        const urls = Array.isArray(server.urls) ? server.urls : [server.urls];
+        return urls.every((url) => url.startsWith("stun:") || url.startsWith("stuns:"));
+    });
+}

package/dist/apps/shared/webrtc/types.d.ts

@@ -14,6 +14,20 @@ export interface ParticipantState {
 /** The emoji a participant can send during a call. A received `reaction` outside this list is ignored, so an
  * arbitrary string from another participant is never rendered. */
 export declare const REACTION_EMOJIS: readonly ["👍", "👏", "❤️", "🎉", "😂", "😮", "😢", "👋"];
+/** Which media path this participant's own tab currently insists on, for every peer - see
+ * `MeshConnectionManager.setTransportMode()`'s doc comment for what each one actually does:
+ *
+ * - `"auto"`: the default waterfall (direct, then TURN, then the server's own WebSocket relay) - unchanged behavior.
+ * - `"p2p"`: direct connections only - no TURN server offered to ICE, and a pair that can't connect directly stays
+ * `"failed"` rather than falling back to the WebSocket relay.
+ * - `"relay"`: TURN-relayed WebRTC only (`RTCPeerConnection`'s own `iceTransportPolicy: "relay"`) - still real
+ * WebRTC, just never direct; a pair with no reachable TURN server stays `"failed"`, same "no further fallback" rule
+ * as `"p2p"`.
+ * - `"websocket"`: skips WebRTC negotiation entirely for every pair and goes straight to the server's own relay.
+ *
+ * A purely personal, per-tab troubleshooting choice - never transmitted as a setting, and independent of whatever
+ * every other participant has chosen for their own tab. */
+export type TransportMode = "auto" | "p2p" | "relay" | "websocket";
 /**
  * One WebRTC signaling message, published as the JSON body of `POST /push/:meetingUid` (`BasePushRoute.send()`)
  * and received back over `/push`'s WebSocket, exactly as `BaseVideoMeetingRoute`'s doc comment describes ("SDP
@@ -33,10 +47,16 @@ export declare const REACTION_EMOJIS: readonly ["👍", "👏", "❤️", "🎉"
  * revoking the recipient's server-side channel grant (`BaseVideoMeetingRoute.revokeChannelGrant()`), which is
  * enforced and does not depend on the recipient's client cooperating; `talking-stick` has no server-side
  * counterpart at all (see this module's doc comment on why it is purely an in-call runtime signal).
+ *
+ * `restart-connection` is broadcast whenever a participant changes their own `TransportMode`
+ * (`MeshConnectionManager.setTransportMode()`): every connection of theirs needs rebuilding under the new policy,
+ * and a recipient can't rebuild its own half of a pair unilaterally without the other side doing the same at
+ * (almost) the same moment, or the two sides end up negotiating against two different `RTCPeerConnection`s. No
+ * payload beyond the envelope - `from` already says whose connection to rebuild.
  */
 export interface SignalMessage {
     type: "video-meeting-signal";
-    kind: "hello" | "bye" | "offer" | "answer" | "ice-candidate" | "presenter-claim" | "presenter-release" | "state" | "reaction" | "relay-fallback" | "mute-request" | "kicked" | "talking-stick";
+    kind: "hello" | "bye" | "offer" | "answer" | "ice-candidate" | "presenter-claim" | "presenter-release" | "state" | "reaction" | "relay-fallback" | "mute-request" | "kicked" | "talking-stick" | "restart-connection";
     /** The sender's authenticated uid - the real caller's own uid when `join()` returned `authenticated: true`, else
      * the `guest:<random>` uid `BaseVideoMeetingRoute.join()` minted. The server refuses a published message whose
      * `from` isn't the authenticated caller's uid (it stops one participant speaking as another), so this must be
@@ -165,8 +185,12 @@ export interface RelayTransportLike {
     setLocalTrack(kind: "audio" | "video", track: MediaStreamTrack | null): void;
     close(): void;
 }
+/** `iceTransportPolicy` mirrors the real `RTCConfiguration` field of the same name - `"relay"` forces ICE to use
+ * only a TURN relay candidate, never a direct one (`TransportMode.relay`); omitted (or `"all"`) is the ordinary,
+ * unconstrained default. */
 export type RTCPeerConnectionFactory = (config: {
     iceServers: RTCIceServer[];
+    iceTransportPolicy?: "all" | "relay";
 }) => RTCPeerConnectionLike;
 /** One other participant currently known to be in the call - `MeshConnectionManager.participants` never includes
  * the local participant themselves (the UI already knows its own name/uid without asking the manager). */

package/dist/lib/util/RedisRelayBus.js

@@ -11,6 +11,9 @@ export const REDIS_RELAY_ORIGIN_BYTES = 16;
 /** The least time between two `warn` messages about failing Redis commands; the failures in between are logged at
  * `debug`, so a Redis outage cannot flood the log at the rate media frames arrive. */
 export const REDIS_RELAY_WARN_INTERVAL_MS = 30000;
+/** How often the bus re-checks, via one batched `PUBSUB NUMSUB`, whether any other replica still has a listener for
+ * each of its active rooms - see the class doc comment's "Skipping Redis for a single-replica room" section. */
+export const REDIS_RELAY_NUMSUB_POLL_MS = 2000;
 /** The channel a room's frames are published to. */
 export function redisRelayChannel(roomId) {
     return REDIS_RELAY_CHANNEL_PREFIX + roomId;
@@ -66,13 +69,33 @@ export function redisRelayChannel(roomId) {
  * server's client-output-buffer limit for pub/sub, after which Redis drops that connection (the client then
  * reconnects and resubscribes).
  *
+ * ## Skipping Redis for a single-replica room
+ *
+ * Every room still SUBSCRIBEs the moment it has its first local listener (so a remote replica's room gets this
+ * replica's frames the instant it starts caring, with no detection delay), but whether `publish()` also bothers
+ * PUBLISHing a frame to Redis in the first place is a separate decision, re-checked on a timer
+ * (`pollLocalOnly()`, every `numSubPollMs`/`REDIS_RELAY_NUMSUB_POLL_MS`): one batched `PUBSUB NUMSUB` across every
+ * active room's channel tells this bus, authoritatively, how many subscribers across the *whole* Redis deployment
+ * each channel has right now. A channel with at most one (this bus's own subscription, and only this bus's own)
+ * has no listener on any other replica - there is nobody `publishToRedis()` could possibly be publishing *for* -
+ * so a room's `localOnly` flag is set and every frame for it skips the Redis publish entirely until the next poll
+ * says otherwise, which is by far the common case for a small/single-instance deployment and for most rooms even in
+ * a larger one (most calls never span replicas). A room `localOnly` has not yet been determined for (just created,
+ * or the last poll failed) keeps publishing - the safe default, since under-publishing (an unnoticed remote
+ * listener goes briefly silent) is a real glitch and over-publishing (a few redundant PUBLISHes with no
+ * subscriber) costs only a little Redis traffic. The gap between a remote replica's listener actually arriving and
+ * this bus's next poll noticing is therefore bounded by `numSubPollMs` - frames to that listener may be briefly
+ * missing right at that boundary, which is the same order of magnitude as, and no worse than, media this transport
+ * already tolerates losing under ordinary backpressure (see `RelayHub.RELAY_WS_MAX_BACKPRESSURE_BYTES`'s own doc
+ * comment).
+ *
  * @author Jean-Philippe Steinmetz
  */
 export class RedisRelayBus {
     constructor(options) {
         this.options = options;
         /** What the bus has done so far. */
-        this.stats = { published: 0, publishSkipped: 0, failures: 0, received: 0, receivedDropped: 0 };
+        this.stats = { published: 0, publishSkipped: 0, publishSkippedLocalOnly: 0, failures: 0, received: 0, receivedDropped: 0 };
         this.id = crypto.randomBytes(REDIS_RELAY_ORIGIN_BYTES);
         this.rooms = new Map();
         this.lastWarn = Number.NEGATIVE_INFINITY;
@@ -108,17 +131,20 @@ export class RedisRelayBus {
         this.subscriber = subscriber;
         this.run("connect the publisher", () => publisher.connect());
         this.run("connect the subscriber", () => subscriber.connect());
+        this.pollTimer = setInterval(() => void this.pollLocalOnly(), this.options.numSubPollMs ?? REDIS_RELAY_NUMSUB_POLL_MS);
     }
     /**
-     * Shuts the bus down: destroys both clients (which ends their connections, and with them every subscription) and
-     * stops publishing. Local delivery keeps working, as after a failed `connect()`. Safe to call repeatedly.
-     * Nothing is unsubscribed one by one: the connection going away does that server-side, and an UNSUBSCRIBE could
-     * only stall behind a Redis that is down.
+     * Shuts the bus down: destroys both clients (which ends their connections, and with them every subscription),
+     * stops the `pollLocalOnly()` timer and stops publishing. Local delivery keeps working, as after a failed
+     * `connect()`. Safe to call repeatedly. Nothing is unsubscribed one by one: the connection going away does that
+     * server-side, and an UNSUBSCRIBE could only stall behind a Redis that is down.
      */
     close() {
         const clients = [this.publisher, this.subscriber];
         this.publisher = undefined;
         this.subscriber = undefined;
+        clearInterval(this.pollTimer);
+        this.pollTimer = undefined;
         for (const client of clients) {
             try {
                 client?.destroy();
@@ -152,6 +178,10 @@ export class RedisRelayBus {
         }
     }
     publishToRedis(roomId, sender, frame) {
+        if (this.rooms.get(roomId)?.localOnly) {
+            this.stats.publishSkippedLocalOnly++;
+            return;
+        }
         const publisher = this.publisher;
         const senderBytes = Buffer.from(sender, "utf8");
         if (!publisher?.isReady || senderBytes.length === 0 || senderBytes.length > 255) {
@@ -166,6 +196,32 @@ export class RedisRelayBus {
         this.stats.published++;
         this.run("publish a frame", () => publisher.publish(redisRelayChannel(roomId), message));
     }
+    /** Refreshes every active room's `localOnly` flag with one batched `PUBSUB NUMSUB`, on the publisher client (the
+     * subscriber is reserved for buffer-mode subscribe/unsubscribe) - see the class doc comment. Does nothing while
+     * there are no rooms or the publisher isn't ready; a failed command is counted and logged, and leaves every
+     * room's existing flag untouched, so a transient failure never flips a room's safe-default `undefined`/`false`
+     * to a stale `true` it didn't earn. Applies results by walking `this.rooms` fresh *after* the command resolves,
+     * not the snapshot queried - a room that left while this was in flight is simply no longer in the map to visit,
+     * and one that arrived gets no count back (`undefined`, falling back to the safe default below) and waits for
+     * the next poll, with no race to special-case either way. */
+    async pollLocalOnly() {
+        const publisher = this.publisher;
+        const roomIds = [...this.rooms.keys()];
+        if (!publisher?.isReady || roomIds.length === 0) {
+            return;
+        }
+        let counts;
+        try {
+            counts = await publisher.pubSubNumSub(roomIds.map(redisRelayChannel));
+        }
+        catch (err) {
+            this.report("check room subscriber counts", err);
+            return;
+        }
+        for (const [roomId, room] of this.rooms) {
+            room.localOnly = (counts[redisRelayChannel(roomId)] ?? Number.POSITIVE_INFINITY) <= 1;
+        }
+    }
     redisSubscribe(roomId, room) {
         const subscriber = this.subscriber;
         if (!subscriber) {

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package/dist/lib/util/RelayHub.js

@@ -15,9 +15,16 @@ export const RELAY_MAX_PAYLOAD_BYTES = 16 * 1024;
 export const RELAY_LARGE_PAYLOAD_BYTES = 64 * 1024;
 /** How many bytes the framework may hold unsent for one relay socket before it drops further messages to it, when the
  * installed `@rapidrest/service-core` lets a route set that (its default, 64 KiB, is smaller than one key frame's worth
- * of headroom and made a briefly slow receiver lose frames constantly). One MiB is roughly two seconds of a 350 kbps
- * video stream plus audio. */
-export const RELAY_WS_MAX_BACKPRESSURE_BYTES = 1024 * 1024;
+ * of headroom and made a briefly slow receiver lose frames constantly). 128 KiB is roughly 2.7 seconds of
+ * `VideoSender.VIDEO_BITRATE` (350 kbps) plus `AudioSender.AUDIO_BITRATE` (24 kbps) combined - (350_000 + 24_000) / 8
+ * bytes/s ≈ 46.75 KB/s - comfortably more than one key frame but not so much that a receiver whose downlink is
+ * briefly slower than the stream is left playing several seconds stale: this was previously 1 MiB (≈22 seconds at
+ * this bitrate, not the "roughly two seconds" the comment claimed - a bits-vs-bytes mix-up), which let a receiver's
+ * backlog grow essentially without bound before anything was ever dropped, rather than the "briefly slow" case this
+ * budget is meant to cover. `apps/shared/relay/`'s own senders can't import this constant (`tsconfig.apps.json`
+ * builds `apps/` as its own program - see its own doc comments), so this is a cross-reference, not a shared import:
+ * changing either bitrate should revisit this value too. */
+export const RELAY_WS_MAX_BACKPRESSURE_BYTES = 128 * 1024;
 /** The most sockets (registered or still awaiting their `hello`) one room holds at once. */
 export const RELAY_MAX_SOCKETS_PER_ROOM = 16;
 /** The most sockets one authenticated uid may hold in one room at once (several tabs or devices of one account). */