@relaymessenger/livekit 0.1.0-staging.1 → 0.1.0-staging.11

package/dist/livekit.d.ts

@@ -2,7 +2,18 @@ import { AudioInput, AudioOutput, type AgentSession } from "@livekit/agents";
 import { AudioFrame } from "@livekit/rtc-node";
 import type Relay from "@relaymessenger/sdk";
 import type { CallRoomOptions } from "@relaymessenger/sdk";
-import { RelayCallTransport, type RelayCallIceDiagnostics, type RelayIceServer, type RelayIceTransportPolicy } from "./transport.js";
+import { RelayCallTransport, type RelayCallIceDiagnostics, type RelayIceServer, type RelayIceServersProvider, type RelayInboundAudioFormat, type RelayIceTransportPolicy } from "./transport.js";
+/**
+ * The format LiveKit's own room input hands an AgentSession:
+ * `@livekit/agents` dist/voice/room_io/room_io.js:46-48
+ * `DEFAULT_ROOM_INPUT_OPTIONS = { audioSampleRate: 24e3, audioNumChannels: 1 }`,
+ * passed to the participant `AudioStream` (room_io.js:373-374). Plugins
+ * resample assuming the channel count they construct with
+ * (`@livekit/agents-plugin-google` 1.9.0 realtime_api.js:1307-1329
+ * `new AudioResampler(frame.sampleRate, 16000, 1)`; rtc-node's resampler reads
+ * raw bytes and ignores `frame.channels`), so the session must receive mono.
+ */
+export declare const LIVEKIT_ROOM_INPUT_AUDIO: Readonly<RelayInboundAudioFormat>;
 /** LiveKit Agents input backed by the remote Relay participant's WebRTC audio. */
 export declare class RelayAudioInput extends AudioInput {
     #private;
@@ -10,11 +21,21 @@ export declare class RelayAudioInput extends AudioInput {
     setAttached(attached: boolean): void;
     close(): Promise<void>;
 }
-/** LiveKit Agents output that publishes TTS PCM to the Relay participant. */
+/**
+ * LiveKit Agents output that publishes TTS PCM to the Relay participant.
+ *
+ * Shape copied from `@livekit/agents` `ParticipantAudioOutput`
+ * (dist/voice/room_io/_output.js): `captureFrame` hands the frame to the
+ * transport and returns without waiting for playout, `flush()` starts a playout
+ * task that reports `onPlaybackFinished` only once the transport has drained,
+ * and `clearBuffer()` resolves the interruption future so that task reports
+ * `interrupted: true` with the position actually played.
+ */
 export declare class RelayAudioOutput extends AudioOutput {
     #private;
     constructor(transport: RelayCallTransport, sampleRate?: number);
     captureFrame(frame: AudioFrame): Promise<void>;
+    /** Mark the segment complete; `onPlaybackFinished` fires once the transport has drained. */
     flush(): void;
     clearBuffer(): void;
     close(): void;
@@ -32,14 +53,24 @@ export interface RelayLiveKitConnectOptions extends RelayLiveKitAudioOptions {
     relay: Relay;
     callId: string;
     room?: CallRoomOptions;
-    /** STUN and TURN servers for the agent's WebRTC peer. Defaults to none. */
-    iceServers?: RelayIceServer[];
+    /**
+     * STUN and TURN servers for the agent's WebRTC peer. Defaults to none. A
+     * function is called again before every restart, to mint fresh TURN credentials.
+     */
+    iceServers?: RelayIceServer[] | RelayIceServersProvider;
     /** `"relay"` forces TURN. Defaults to `"all"`. */
     iceTransportPolicy?: RelayIceTransportPolicy;
-    /** Maximum wait for WebRTC media to connect. Defaults to 15 seconds. */
+    /**
+     * How long one SFU session has to connect before the transport restarts onto
+     * a new one. Defaults to 5 seconds. `connect()` has no overall deadline.
+     */
+    sessionConnectTimeoutMs?: number;
+    /** @deprecated Use `sessionConnectTimeoutMs`; now the per-session wait, not a `connect()` deadline. */
     mediaConnectTimeoutMs?: number;
-    /** @deprecated Use `mediaConnectTimeoutMs`. */
+    /** @deprecated Use `sessionConnectTimeoutMs`. */
     connectionTimeoutMs?: number;
+    /** Aborting stops `connect()` and closes the transport; the Call itself is not ended. */
+    signal?: AbortSignal;
 }
 type AgentSessionAudioTarget = Pick<AgentSession, "input" | "output">;
 /**
@@ -56,6 +87,12 @@ export declare class RelayLiveKitCall {
     static connect(options: RelayLiveKitConnectOptions): Promise<RelayLiveKitCall>;
     attach(session: AgentSessionAudioTarget): void;
     detach(): void;
+    /**
+     * Resolves once the person's audio has arrived and the room shows them
+     * connected (the transport's `peerAudio`). Await it before starting the
+     * AgentSession so the greeting is heard.
+     */
+    waitForPeerAudio(timeoutMs: number): Promise<void>;
     /** ICE candidates and state transitions for this call, with a one-line summary for logs. */
     diagnostics(): RelayCallIceDiagnostics;
     setMuted(muted: boolean): void;

package/dist/livekit.d.ts.map

@@ -1 +1 @@
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package/dist/livekit.js

@@ -1,12 +1,28 @@
-import { AudioInput, AudioOutput } from "@livekit/agents";
+import { AudioInput, AudioOutput, Future } from "@livekit/agents";
 import { AudioFrame } from "@livekit/rtc-node";
 import { TransformStream } from "node:stream/web";
 import { RelayCallTransport, } from "./transport.js";
+/**
+ * The format LiveKit's own room input hands an AgentSession:
+ * `@livekit/agents` dist/voice/room_io/room_io.js:46-48
+ * `DEFAULT_ROOM_INPUT_OPTIONS = { audioSampleRate: 24e3, audioNumChannels: 1 }`,
+ * passed to the participant `AudioStream` (room_io.js:373-374). Plugins
+ * resample assuming the channel count they construct with
+ * (`@livekit/agents-plugin-google` 1.9.0 realtime_api.js:1307-1329
+ * `new AudioResampler(frame.sampleRate, 16000, 1)`; rtc-node's resampler reads
+ * raw bytes and ignores `frame.channels`), so the session must receive mono.
+ */
+export const LIVEKIT_ROOM_INPUT_AUDIO = Object.freeze({
+    sampleRate: 24_000,
+    channelCount: 1,
+});
 /** LiveKit Agents input backed by the remote Relay participant's WebRTC audio. */
 export class RelayAudioInput extends AudioInput {
     #transport;
     #writer;
     #onAudio;
+    /** Serial write chain: each frame waits for the writer, as `pipeTo` does in LiveKit's input. */
+    #writes = Promise.resolve();
     #attached = false;
     #closed = false;
     constructor(transport) {
@@ -19,7 +35,16 @@ export class RelayAudioInput extends AudioInput {
             if (this.#closed || !this.#attached)
                 return;
             const audio = new AudioFrame(frame.samples, frame.sampleRate, frame.channelCount, frame.samples.length / frame.channelCount);
-            void this.#writer.write(audio).catch(() => undefined);
+            // LiveKit's ParticipantAudioInputStream pipes its AudioStream into the
+            // session with `pipeTo(output.writable)` (room_io/_input.js:161-162),
+            // which awaits the writer before each chunk. The transport pushes and
+            // cannot be paused, so the frames wait here in order instead.
+            this.#writes = this.#writes
+                .then(async () => {
+                await this.#writer.ready;
+                await this.#writer.write(audio);
+            })
+                .catch(() => undefined);
         };
         this.#transport.on("audio", this.#onAudio);
     }
@@ -31,16 +56,30 @@ export class RelayAudioInput extends AudioInput {
             return;
         this.#closed = true;
         this.#transport.off("audio", this.#onAudio);
+        await this.#writes;
         await this.#writer.close().catch(() => undefined);
         await super.close();
     }
 }
-/** LiveKit Agents output that publishes TTS PCM to the Relay participant. */
+/**
+ * LiveKit Agents output that publishes TTS PCM to the Relay participant.
+ *
+ * Shape copied from `@livekit/agents` `ParticipantAudioOutput`
+ * (dist/voice/room_io/_output.js): `captureFrame` hands the frame to the
+ * transport and returns without waiting for playout, `flush()` starts a playout
+ * task that reports `onPlaybackFinished` only once the transport has drained,
+ * and `clearBuffer()` resolves the interruption future so that task reports
+ * `interrupted: true` with the position actually played.
+ */
 export class RelayAudioOutput extends AudioOutput {
     #transport;
-    #segmentDuration = 0;
-    #segmentStarted = false;
-    #segmentStartedAt = 0;
+    /** Seconds pushed to the transport in the open segment. */
+    #pushedDuration = 0;
+    #firstFrameEmitted = false;
+    #flushTask;
+    #flushDone = true;
+    /** Resolved by `clearBuffer()` with the milliseconds still queued at that moment. */
+    #interruptedFuture = new Future();
     #closed = false;
     constructor(transport, sampleRate = 48_000) {
         super(sampleRate, undefined, { pause: false });
@@ -49,38 +88,47 @@ export class RelayAudioOutput extends AudioOutput {
     async captureFrame(frame) {
         if (this.#closed)
             throw new Error("Relay LiveKit audio output is closed.");
+        if (this.#flushTask && !this.#flushDone) {
+            this.logger.error("captureFrame called while flush is in progress");
+            await this.#flushTask;
+        }
         await super.captureFrame(frame);
-        if (!this.#segmentStarted) {
-            this.#segmentStarted = true;
-            this.#segmentStartedAt = Date.now();
-            this.onPlaybackStarted(this.#segmentStartedAt);
+        if (!this.#firstFrameEmitted) {
+            this.#firstFrameEmitted = true;
+            this.onPlaybackStarted(Date.now());
         }
-        this.#segmentDuration += frame.samplesPerChannel / frame.sampleRate;
+        this.#pushedDuration += frame.samplesPerChannel / frame.sampleRate;
+        // Resolves once the slices are queued; the engine's pump paces the wire.
         await this.#transport.writeAudio({
             samples: frame.data,
             sampleRate: frame.sampleRate,
             channelCount: frame.channels,
         });
     }
+    /** Mark the segment complete; `onPlaybackFinished` fires once the transport has drained. */
     flush() {
         super.flush();
-        if (!this.#segmentStarted)
+        if (!this.#pushedDuration)
             return;
-        this.onPlaybackFinished({
-            playbackPosition: this.#segmentDuration,
-            interrupted: false,
+        if (this.#flushTask && !this.#flushDone)
+            return;
+        this.#flushDone = false;
+        this.#flushTask = this.#waitForPlayoutTask().finally(() => {
+            this.#flushDone = true;
         });
-        this.#resetSegment();
+        void this.#flushTask.catch(() => undefined);
     }
     clearBuffer() {
+        const queuedMs = this.#transport.queuedAudioMs();
         this.#transport.clearAudio();
-        if (!this.#segmentStarted)
+        if (this.#interruptedFuture.done)
             return;
-        this.onPlaybackFinished({
-            playbackPosition: Math.min(this.#segmentDuration, Math.max(0, (Date.now() - this.#segmentStartedAt) / 1_000)),
-            interrupted: true,
-        });
-        this.#resetSegment();
+        if (this.#pushedDuration === 0 && this.pendingPlayoutSegments === 0)
+            return;
+        if (!this.#flushTask || this.#flushDone)
+            this.flush();
+        if (!this.#interruptedFuture.done)
+            this.#interruptedFuture.resolve(queuedMs);
     }
     close() {
         if (this.#closed)
@@ -88,10 +136,19 @@ export class RelayAudioOutput extends AudioOutput {
         this.#closed = true;
         this.clearBuffer();
     }
-    #resetSegment() {
-        this.#segmentDuration = 0;
-        this.#segmentStarted = false;
-        this.#segmentStartedAt = 0;
+    async #waitForPlayoutTask() {
+        const interruptedFuture = this.#interruptedFuture;
+        await Promise.race([this.#transport.waitForPlayout(), interruptedFuture.await]);
+        const interrupted = interruptedFuture.done;
+        let playbackPosition = this.#pushedDuration;
+        if (interrupted) {
+            playbackPosition = Math.max(0, playbackPosition - interruptedFuture.result / 1_000);
+        }
+        this.#pushedDuration = 0;
+        this.#firstFrameEmitted = false;
+        if (this.#interruptedFuture === interruptedFuture)
+            this.#interruptedFuture = new Future();
+        this.onPlaybackFinished({ playbackPosition, interrupted });
     }
 }
 /** Build LiveKit Agents audio IO around an already-connected Relay transport. */
@@ -119,6 +176,7 @@ export class RelayLiveKitCall {
         const transportOptions = {
             relay: options.relay,
             callId: options.callId,
+            inboundAudio: LIVEKIT_ROOM_INPUT_AUDIO,
             ...(options.room ? { room: options.room } : {}),
             ...(options.roomClient ? { roomClient: options.roomClient } : {}),
             ...(options.webRTC ? { webRTC: options.webRTC } : {}),
@@ -127,6 +185,9 @@ export class RelayLiveKitCall {
             ...(options.iceGatheringTimeoutMs === undefined
                 ? {}
                 : { iceGatheringTimeoutMs: options.iceGatheringTimeoutMs }),
+            ...(options.sessionConnectTimeoutMs === undefined
+                ? {}
+                : { sessionConnectTimeoutMs: options.sessionConnectTimeoutMs }),
             ...(options.mediaConnectTimeoutMs === undefined
                 ? {}
                 : { mediaConnectTimeoutMs: options.mediaConnectTimeoutMs }),
@@ -135,7 +196,7 @@ export class RelayLiveKitCall {
                 : { connectionTimeoutMs: options.connectionTimeoutMs }),
         };
         const transport = new RelayCallTransport(transportOptions);
-        await transport.connect();
+        await transport.connect(options.signal ? { signal: options.signal } : {});
         return new RelayLiveKitCall(transport, createRelayLiveKitAudio(transport, options));
     }
     attach(session) {
@@ -156,6 +217,14 @@ export class RelayLiveKitCall {
             this.#session.output.audio = null;
         this.#session = undefined;
     }
+    /**
+     * Resolves once the person's audio has arrived and the room shows them
+     * connected (the transport's `peerAudio`). Await it before starting the
+     * AgentSession so the greeting is heard.
+     */
+    waitForPeerAudio(timeoutMs) {
+        return this.transport.waitForPeerAudio(timeoutMs);
+    }
     /** ICE candidates and state transitions for this call, with a one-line summary for logs. */
     diagnostics() {
         return this.transport.diagnostics();

package/dist/livekit.js.map

@@ -1 +1 @@
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package/dist/transport.d.ts

@@ -12,6 +12,39 @@ export interface RelayIceServer {
     credential?: string;
 }
 export type RelayIceTransportPolicy = "all" | "relay";
+/**
+ * Returns the ICE servers for one peer connection. Called before the first
+ * peer and again before every restart, so short-lived TURN credentials can be
+ * minted per attempt (PartyTracks: a reconnect "will trigger new sessionId,
+ * new ice server credentials and a new peerConnection").
+ */
+export type RelayIceServersProvider = (attempt: {
+    /** 0 for the first peer, then 1, 2, ... for each restart. */
+    restarts: number;
+}) => RelayIceServer[] | Promise<RelayIceServer[]>;
+/** Why the transport replaced its peer connection with a new SFU session. */
+export type RelayCallRestartReason = "timeout" | "failed" | "disconnected" | "error";
+export interface RelayCallRestartEvent {
+    reason: RelayCallRestartReason;
+    /** `diagnostics().summary` of the session being replaced, taken as it was given up. */
+    summary: string;
+    /** Restarts since `connect()`, this one included. */
+    restarts: number;
+    /** Backoff waited before the new peer was built. */
+    delayMs: number;
+}
+/**
+ * PCM format the engine decodes the remote participant's Opus into. The values
+ * are the ones libopus decodes to (`@evan/opus` lib.d.ts: `channels?: 1 | 2`,
+ * `sample_rate?: 8000 | 12000 | 16000 | 24000 | 48000`); the decoder itself
+ * resamples and downmixes, so no PCM is converted by hand.
+ */
+export interface RelayInboundAudioFormat {
+    sampleRate: 8000 | 12000 | 16000 | 24000 | 48000;
+    channelCount: 1 | 2;
+}
+/** Opus's native rate and Relay's wire channel count. */
+export declare const DEFAULT_INBOUND_AUDIO: Readonly<RelayInboundAudioFormat>;
 export type RelayCallEngine = "werift" | "wrtc";
 export interface RelayCallTransportOptions {
     relay: Relay;
@@ -24,16 +57,79 @@ export interface RelayCallTransportOptions {
     engine?: RelayCallEngine;
     /**
      * STUN and TURN servers handed to the engine's `RTCPeerConnection`. Defaults
-     * to none: Cloudflare's SFU answers with its own host candidates. Set TURN
-     * servers when the agent runs behind a NAT or firewall that blocks UDP.
+     * to Cloudflare's STUN server, the configuration of Cloudflare's own Realtime
+     * echo example. Set TURN servers when the agent runs behind a NAT or firewall
+     * that blocks UDP. A function is called before every peer connection,
+     * restarts included, so it can mint fresh TURN credentials each time.
      */
-    iceServers?: RelayIceServer[];
+    iceServers?: RelayIceServer[] | RelayIceServersProvider;
     /** `"relay"` forces every candidate through TURN. Defaults to `"all"`. */
     iceTransportPolicy?: RelayIceTransportPolicy;
-    /** Maximum wait for the WebRTC peer to become connected. Defaults to 15 seconds. */
+    /**
+     * How long one SFU session has, after its answer is applied, to reach
+     * `connected` before the transport restarts onto a new session. Defaults to
+     * 5 seconds. `connect()` itself has no deadline: it waits through restarts
+     * until media connects, the Call ends, `close()` is called, or its signal aborts.
+     */
+    sessionConnectTimeoutMs?: number;
+    /** @deprecated Use `sessionConnectTimeoutMs`; now the per-session wait, not a `connect()` deadline. */
     mediaConnectTimeoutMs?: number;
-    /** @deprecated Use `mediaConnectTimeoutMs`. */
+    /** @deprecated Use `sessionConnectTimeoutMs`. */
     connectionTimeoutMs?: number;
+    /**
+     * PCM format of the `audio` events: the Opus decoder decodes the remote
+     * track straight to this rate and channel count. Defaults to 48 kHz stereo.
+     * The `"wrtc"` engine delivers libwebrtc's own format and accepts only the default.
+     */
+    inboundAudio?: RelayInboundAudioFormat;
+    /**
+     * Called once per call when outbound audio is queued but no RTP packet has
+     * been written for 2 s while media is connected. Receives the diagnostics
+     * summary. Defaults to a no-op; nothing is restarted.
+     */
+    onWarning?: (message: string) => void;
+}
+/** Packet counts for the other participant's track, as seen by this participant. */
+export interface RelayCallInboundDiagnostics {
+    /** RTP packets received on the subscribed track. */
+    rtpPackets: number;
+    /** Opus packets the decoder rejected. */
+    decodeFailures: number;
+    /** PCM frames handed to the `audio` listeners. */
+    frames: number;
+    /** ms since `connect()` for the first and last RTP packet; undefined until one arrives. */
+    firstPacketAtMs: number | undefined;
+    lastPacketAtMs: number | undefined;
+    /** RTP packets received in the 5 s before `diagnostics()` was called. */
+    recentRtpPackets: number;
+}
+/** Packet counts for this participant's published track. */
+export interface RelayCallOutboundDiagnostics {
+    /** PCM slices accepted from the caller and handed to the engine source. */
+    frames: number;
+    opusPackets: number;
+    /** RTP packets carrying the caller's audio. */
+    rtpPackets: number;
+    /** RTP packets carrying Opus silence, written while nothing was queued. */
+    silencePackets: number;
+    /** ms since `connect()` for the first and last RTP packet of either kind. */
+    firstPacketAtMs: number | undefined;
+    lastPacketAtMs: number | undefined;
+    /** RTP packets of either kind written in the 5 s before `diagnostics()` was called. */
+    recentRtpPackets: number;
+    /** Encoded packets waiting for the 20 ms pacer. */
+    queued: number;
+    /** `undefined` when the engine does not expose its pacer (`wrtc`). */
+    pacerAlive: boolean | undefined;
+}
+/** Room signaling frames counted since `connect()`. */
+export interface RelayCallRoomDiagnostics {
+    roomStates: number;
+    /** Subscription (pull) offers received from the room. */
+    offers: number;
+    endedReason: string | undefined;
+    /** `error` frame messages, in order. */
+    errors: string[];
 }
 /** ICE facts recorded for one call, for logs and for the connect timeout error. */
 export interface RelayCallIceDiagnostics {
@@ -55,7 +151,18 @@ export interface RelayCallIceDiagnostics {
         state: string;
         atMs: number;
     }>;
+    /**
+     * The local side of the candidate pair media flows on, as `"<type> <protocol>"`
+     * (for example `"relay udp"`), read from the W3C stats once the peer connects;
+     * undefined until then or when the engine has no `getStats()`.
+     */
+    selectedPair: string | undefined;
     connected: boolean;
+    inbound: RelayCallInboundDiagnostics;
+    outbound: RelayCallOutboundDiagnostics;
+    room: RelayCallRoomDiagnostics;
+    /** Peer connections replaced by a new SFU session since `connect()`. */
+    restarts: number;
     /** One-line rendering of the fields above. */
     summary: string;
 }
@@ -67,13 +174,47 @@ export interface RelayCallTransportCloseEvent {
 type TransportEventMap = {
     audio: [RelayAudioFrame];
     connected: [];
+    /** A new peer connection published on a new SFU session; audio continues on it. */
+    restarted: [RelayCallRestartEvent];
     roomState: [CallRoomStateFrame];
+    /** The person's camera started (`true`) or stopped (`false`) sending, from `roomState`. */
+    remoteVideo: [boolean];
+    /**
+     * Once per call: the person's audio has reached this peer (first inbound
+     * frame on the pulled track) and `roomState` shows the person connected. The
+     * room pulls both directions together, so from here the person hears what is
+     * written; start speaking after this, as LiveKit Agents start a session once
+     * the participant is in the room.
+     */
+    peerAudio: [];
     ended: [CallRoomEndedFrame];
     error: [Error];
     close: [RelayCallTransportCloseEvent];
 };
 type TransportEvent = keyof TransportEventMap;
 type TransportListener<K extends TransportEvent> = (...args: TransportEventMap[K]) => void;
+/**
+ * Restart rule, copied from PartyTracks and Cloudflare (PROTOCOL.md section 4):
+ * not `connected` 5 s after the SFU answer (Cloudflare's echo example waits
+ * 5000 ms; SFU operations block up to 5 s awaiting `connected`), `failed`, or
+ * `disconnected` for 7 s (PartyTracks.ts `timeoutSeconds = 7`). Backoff 250 ms
+ * x1.1 per attempt, capped at 10 s (PartyTracks `retryWithBackoff`
+ * `backoffFactor: 1.1`, rxjs-helpers.ts defaults). werift never reports
+ * `failed` on a session whose checks go unanswered (ice.js:983-986), so the
+ * 5 s timer is the trigger that fires in practice.
+ */
+export declare const RESTART_CONNECT_TIMEOUT_MS = 5000;
+export declare const RESTART_DISCONNECTED_MS = 7000;
+export declare const RESTART_INITIAL_DELAY_MS = 250;
+export declare const RESTART_BACKOFF_FACTOR = 1.1;
+export declare const RESTART_MAX_DELAY_MS = 10000;
+/**
+ * Backoff before restart number `attempt` (1-based) counted since the last
+ * session that connected. PartyTracks passes `resetOnSuccess: true` to rxjs
+ * `retry` (rxjs-helpers.ts:18, :39); this transport reads "success" as a
+ * session reaching `connected`.
+ */
+export declare const restartDelayMs: (attempt: number) => number;
 export declare class RelayCallTransportError extends Error {
     readonly code?: string;
     constructor(message: string, code?: string);
@@ -82,25 +223,56 @@ export declare class RelayCallTransportError extends Error {
  * Provider-neutral Node WebRTC bridge for Relay Call rooms.
  *
  * The class owns Relay media negotiation internally. Higher-level adapters only
- * exchange PCM16 frames and call lifecycle events.
+ * exchange PCM16 frames and call lifecycle events. When an SFU session never
+ * connects or dies, the transport builds a new peer connection on a new
+ * session (PROTOCOL.md section 4); the audio source and the `audio` events
+ * carry on across the swap, so adapters only hear silence.
  */
 export declare class RelayCallTransport {
     #private;
     constructor(options: RelayCallTransportOptions);
     on<K extends TransportEvent>(event: K, listener: TransportListener<K>): this;
     off<K extends TransportEvent>(event: K, listener: TransportListener<K>): this;
-    connect(): Promise<void>;
+    /**
+     * Join the room, publish, and resolve on the first `connected`. Dead SFU
+     * sessions are replaced as they are found (PROTOCOL.md section 4), with no
+     * overall deadline: this rejects only when the Call ends, the room or
+     * transport closes, the room reports an error, or `signal` aborts (which
+     * also closes the transport).
+     */
+    connect(options?: {
+        signal?: AbortSignal;
+    }): Promise<void>;
     /** Replace only the room signaling socket and replay the identical publication. */
     reconnect(): Promise<void>;
-    /** ICE candidates and state transitions recorded for this call, with a one-line summary. */
+    /**
+     * ICE candidates, state transitions, packet counts in both directions,
+     * room frame counts and restarts recorded for this call, with a one-line summary.
+     */
     diagnostics(): RelayCallIceDiagnostics;
     /**
      * Feed interleaved PCM16 audio into Relay. Frames are split into 10 ms WebRTC
-     * source slices and paced in real time, so adapters may push larger chunks.
+     * source slices and handed to the engine at once; the engine's 20 ms pump
+     * paces the wire, so adapters may push faster than real time (LiveKit's
+     * `AudioSource.captureFrame` shape). Resolves once the slices are queued;
+     * `waitForPlayout()` tells when they have left.
      */
     writeAudio(frame: RelayAudioFrame): Promise<void>;
-    /** Drop queued outgoing PCM. At most one already-submitted 10 ms slice remains. */
+    /** Milliseconds of audio accepted by `writeAudio` but not yet written to RTP. */
+    queuedAudioMs(): number;
+    /**
+     * Resolves when every accepted slice has been written to RTP and the engine's
+     * pump is idle; immediately when nothing is queued; early on `clearAudio()`
+     * or `close()` (the caller reads `queuedAudioMs()` to learn what was dropped).
+     */
+    waitForPlayout(): Promise<void>;
+    /** Drop outgoing PCM that has not reached RTP and release `waitForPlayout()` callers. */
     clearAudio(): void;
+    /**
+     * Resolves once `peerAudio` has fired (at once if it already has). Rejects
+     * after `timeoutMs`, or when the Call ends or the transport closes first.
+     */
+    waitForPeerAudio(timeoutMs: number): Promise<void>;
     setMuted(muted: boolean): void;
     end(): void;
     close(): void;

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