@relaymessenger/livekit 0.1.0-staging.1 → 0.1.0-staging.10

package/README.md

@@ -39,10 +39,21 @@ async function answerCall(callId: string, session: AgentSession) {
 ```
 
 `RelayLiveKitCall.connect()` does not resolve until the WebRTC media peer has
-reached `connected`. `RelayAudioInput` converts the remote participant's PCM16
-audio to LiveKit frames. `RelayAudioOutput` publishes LiveKit TTS frames to
-Relay, pacing larger frames into 10 ms WebRTC source slices. `clearBuffer()`
-drops queued outbound audio that has not yet reached WebRTC.
+reached `connected`. `RelayAudioInput` hands the AgentSession the remote
+participant's audio as 24 kHz mono PCM16 frames, the format of LiveKit's own
+room input; the Opus decoder produces that format directly.
+
+`RelayAudioOutput` has the shape of LiveKit's own `ParticipantAudioOutput`:
+`captureFrame()` hands the frame to the transport and returns at once, so the
+AgentSession may push a whole reply faster than real time; the engine's 20 ms
+pump paces the wire. `flush()` closes the segment and reports
+`playbackFinished` only after the transport has drained. `clearBuffer()` drops
+audio that has not reached the wire and reports the segment as interrupted at
+the position that actually played.
+
+On the transport, `writeAudio()` resolves once its 10 ms slices are queued,
+`queuedAudioMs()` is what has not left yet, and `waitForPlayout()` resolves when
+the queue is empty and the pump is idle (early on `clearAudio()`).
 
 Use `setMuted(true)` to publish participant mute state, `end()` to end the Relay
 Call, and `close()` for local cleanup. If the signaling connection is replaced,
@@ -63,6 +74,9 @@ const transport = new RelayCallTransport({ room });
 transport.on("audio", ({ samples, sampleRate, channelCount }) => {
   // Interleaved signed PCM16 from the remote Relay participant.
 });
+// Inbound audio is 48 kHz stereo unless you pass
+// `inboundAudio: { sampleRate: 8000 | 12000 | 16000 | 24000 | 48000, channelCount: 1 | 2 }`;
+// the `"wrtc"` engine accepts only the default.
 
 await transport.connect();
 await transport.writeAudio({
@@ -70,6 +84,7 @@ await transport.writeAudio({
   sampleRate: 48_000,
   channelCount: 1,
 });
+await transport.waitForPlayout();
 ```
 
 ## ICE servers, TURN, and diagnostics
@@ -96,19 +111,56 @@ const call = await RelayLiveKitCall.connect({
     },
   ],
   iceTransportPolicy: "all",
-  mediaConnectTimeoutMs: 15_000,
 });
 ```
 
-`mediaConnectTimeoutMs` bounds the wait for the media peer to reach
-`connected` (15 seconds by default). When that wait expires, the error message
-carries what ICE saw, for example
-`Timed out connecting Relay WebRTC media (local: host 2, srflx 0, relay 0; remote: udp 1473; states: new→complete 0.2s, connecting 0.3s, no connected)`.
+`iceServers` may also be a function; it is called before every peer
+connection, so it can mint fresh TURN credentials for each restart.
+
+`connect()` resolves when media first reaches `connected`. It has no overall
+deadline: when an SFU session is not `connected` within
+`sessionConnectTimeoutMs` (5 seconds by default) of its answer, becomes
+`failed`, or stays `disconnected` for 7 seconds, the transport closes that
+peer, waits 250 ms (x1.1 per further attempt, at most 10 s), and publishes
+from a new peer on a new session, for as long as the Call is ringing or in
+progress. Outgoing audio keeps flowing into the new peer. `connect()` rejects
+only when the Call ends, the room or transport closes, the room reports an
+error, or the `signal` passed to it aborts. Each replacement emits
+`restarted` with the reason and the replaced session's summary, for example
+`local: host 2, srflx 0, relay 0; remote: udp 1473; states: new→complete 0.2s, connecting 0.3s, no connected; …`,
+and `diagnostics().restarts` counts them. `mediaConnectTimeoutMs` and
+`connectionTimeoutMs` are deprecated names for `sessionConnectTimeoutMs`.
+
+`call.waitForPeerAudio(timeoutMs)` resolves once the person's audio has
+arrived and the room shows them connected (the transport's `peerAudio` event);
+start the agent's session after it so the first words are heard.
+
 The same facts are available at any time from `call.diagnostics()` or
 `transport.diagnostics()`: local candidate counts by type, the remote
 candidates' transport and port (never their address), the ICE gathering, ICE
 connection and peer connection state changes with their offsets from
-`connect()`, and the one-line `summary`.
+`connect()`, packet counts in both directions (`inbound`: RTP received, Opus
+decode failures, PCM frames delivered, first and last packet offsets, packets
+in the last 5 s; `outbound`: PCM frames accepted, Opus packets, RTP written
+with the caller's audio, RTP written with silence, first and last packet
+offsets, packets in the last 5 s, paced queue size, pacer state), the room frames seen (`roomState` count, pull `offer` count,
+`ended` reason, `error` messages), and the one-line `summary`, for example
+`…; in: 1234 rtp, 0 bad, 1234 frames, first 0.9s last 41.2s, 250/5s; out: 2600 frames, 1300 opus, 1300 rtp, silence 700, first 1.1s last 41.0s, 250/5s, queue 0, pacer alive; room: 3 roomState, 1 offer`.
+Packet counts come from the `werift` engine; `wrtc` reports zero packets and
+`pacer n/a`.
+
+Like a live microphone, the `werift` engine's published track sends one Opus
+packet every 20 ms from the moment media connects until the transport closes:
+the caller's audio when some is queued, Opus silence otherwise. Cloudflare's
+SFU will not let the person's side pull a track that has carried no RTP, so a
+silent agent track would never be heard. Silence never counts toward
+`queuedAudioMs()` or `waitForPlayout()`. The `wrtc` engine sends only the
+audio written to it; write silence yourself if you use it.
+
+`onWarning` is called once per call, with the summary, when outbound audio is
+queued but no RTP packet has been written for 2 s while media is connected.
+Nothing is restarted; the callback exists so the failing direction is named in
+the agent's logs.
 
 `RelayCallTransport` consumes the SDK's `CallRoom`; it does not duplicate the
 room protocol. The default engine is `werift` (pure TypeScript WebRTC) with

package/dist/engine-werift.d.ts

@@ -6,6 +6,8 @@ export declare const WERIFT_CHANNEL_COUNT = 2;
 /** One RTP packet carries 20 ms: 960 frames at 48 kHz, timestamp advances by 960. */
 export declare const WERIFT_PACKET_MS = 20;
 export type WeriftAudioSourceData = Parameters<RelayAudioSourceLike["onData"]>[0];
+/** Packets counted for `diagnostics()`; the recent window is the last 5 s. */
+export declare const STATS_WINDOW_MS = 5000;
 /** Convert any PCM16 frame to interleaved 48 kHz stereo by linear interpolation. */
 export declare const toWireFormat: (data: {
     samples: Int16Array;

package/dist/engine-werift.d.ts.map

@@ -1 +1 @@
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+{"version":3,"file":"engine-werift.d.ts","sourceRoot":"","sources":["../src/engine-werift.ts"],"names":[],"mappings":"AAqCA,OAAO,EAGL,iBAAiB,EAIlB,MAAM,QAAQ,CAAC;AAChB,OAAO,KAAK,EAGV,oBAAoB,EAIpB,yBAAyB,EAEzB,kBAAkB,EACnB,MAAM,gBAAgB,CAAC;AAExB,sGAAsG;AACtG,eAAO,MAAM,kBAAkB,QAAS,CAAC;AACzC,eAAO,MAAM,oBAAoB,IAAI,CAAC;AACtC,qFAAqF;AACrF,eAAO,MAAM,gBAAgB,KAAK,CAAC;AAMnC,MAAM,MAAM,qBAAqB,GAAG,UAAU,CAAC,oBAAoB,CAAC,QAAQ,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC;AAElF,8EAA8E;AAC9E,eAAO,MAAM,eAAe,OAAQ,CAAC;AA8BrC,oFAAoF;AACpF,eAAO,MAAM,YAAY,GAAI,MAAM;IACjC,OAAO,EAAE,UAAU,CAAC;IACpB,UAAU,EAAE,MAAM,CAAC;IACnB,YAAY,EAAE,MAAM,CAAC;CACtB,KAAG,UAwBH,CAAC;AAwPF,eAAO,MAAM,0BAA0B,GACrC,SAAQ,yBAAyE,KAChF,iBAeC,CAAC;AAEL,eAAO,MAAM,yBAAyB,QAAO,kBAI3C,CAAC"}

package/dist/engine-werift.js

@@ -21,7 +21,11 @@
  *   offsets sequenceNumber/timestamp, so the local track starts both at 0.
  * - ../../rtp/src/codec/opus.d.ts: `OpusRtpPayload.deSerialize(buffer)`.
  * - node_modules/@evan/opus/lib.d.ts: `Encoder.encode(pcm16)` returns one Opus
- *   packet; `Decoder.decode(packet)` returns interleaved PCM16 bytes.
+ *   packet; `Decoder.decode(packet)` returns interleaved PCM16 bytes;
+ *   `new Decoder({ channels?: 1 | 2, sample_rate?: 8000 | 12000 | 16000 |
+ *   24000 | 48000 })` decodes any Opus stream to that count and rate (libopus
+ *   `opus_decoder_create(Fs, channels)`), which is how the sink honours
+ *   `RelayInboundAudioFormat`.
  * Participant shapes copied from the headless phone that joined Relay's room
  * through Cloudflare's SFU on 2026-09-21 (_runtime/bin/headless-phone-webrtc.mjs
  * lines 93-160): `bundlePolicy: "max-bundle", iceServers: []` (Cloudflare
@@ -41,6 +45,34 @@ const PACKET_FRAMES = (WERIFT_SAMPLE_RATE * WERIFT_PACKET_MS) / 1000;
 const PACKET_SAMPLES = PACKET_FRAMES * WERIFT_CHANNEL_COUNT;
 /** Dynamic payload type; werift's sender replaces it with the negotiated one. */
 const LOCAL_PAYLOAD_TYPE = 111;
+/** Packets counted for `diagnostics()`; the recent window is the last 5 s. */
+export const STATS_WINDOW_MS = 5_000;
+/** Monotone packet counter with first/last epoch timestamps and a 5 s window. */
+class PacketClock {
+    count = 0;
+    firstAt;
+    lastAt;
+    #recent = [];
+    mark(now = Date.now()) {
+        this.count += 1;
+        this.firstAt ??= now;
+        this.lastAt = now;
+        this.#recent.push(now);
+        this.#prune(now);
+    }
+    recent(now = Date.now()) {
+        this.#prune(now);
+        return this.#recent.length;
+    }
+    #prune(now) {
+        const floor = now - STATS_WINDOW_MS;
+        let drop = 0;
+        while (drop < this.#recent.length && this.#recent[drop] < floor)
+            drop += 1;
+        if (drop)
+            this.#recent.splice(0, drop);
+    }
+}
 /** Convert any PCM16 frame to interleaved 48 kHz stereo by linear interpolation. */
 export const toWireFormat = (data) => {
     const { samples, sampleRate, channelCount } = data;
@@ -69,8 +101,21 @@ export const toWireFormat = (data) => {
 };
 /**
  * PCM16 in, paced 20 ms Opus RTP out. Frames are accumulated into whole
- * packets; a timer drains one packet every 20 ms so the wire sees a steady
- * cadence regardless of how the adapter chunks its writes.
+ * packets; a timer writes one packet every 20 ms so the wire sees a steady
+ * cadence regardless of how the adapter chunks its writes. Adapters may push
+ * far ahead of real time: `queuedMs()` is what has not left yet and
+ * `waitForDrain()` resolves once the queued application audio has been
+ * written (LiveKit's `AudioSource.queuedDuration` / `waitForPlayout` shape).
+ *
+ * From `start()` (the transport calls it when the peer first connects) until
+ * the track stops, the pump never pauses: a tick with no application audio
+ * queued writes an Opus silence frame instead, as a live microphone track
+ * does. Cloudflare's SFU refuses to pull a published track that has carried
+ * no RTP: `tracks/new` for the remote track returned `empty_track_error` "No
+ * track data from remote peer" after about 8.2 s on staging 2026-09-22, and
+ * the same pull returned in about 470 ms with silence flowing. Silence and
+ * application packets share one sequence and timestamp line, so queued audio
+ * replaces silence on the next tick with no gap and no jump.
  */
 class WeriftAudioSource {
     #encoder = new Encoder({
@@ -78,26 +123,69 @@ class WeriftAudioSource {
         sample_rate: WERIFT_SAMPLE_RATE,
         application: "voip",
     });
+    /** 20 ms of digital silence, encoded once by the same encoder and reused. */
+    #silence = Buffer.from(this.#encoder.encode(new Int16Array(PACKET_SAMPLES)));
     #track = new MediaStreamTrack({ kind: "audio", id: "microphone", streamId: "relay-call" });
     #packets = [];
+    /** Every RTP packet written, silence included. */
+    #rtp = new PacketClock();
+    #opusPackets = 0;
+    #applicationRtpPackets = 0;
+    #silencePackets = 0;
     #pending = new Int16Array(0);
     #sequenceNumber = 1;
     #timestamp = 0;
     #first = true;
     #pump;
+    #started = false;
     #stopped = false;
+    #drainWaiters = new Set();
+    stats() {
+        return {
+            opusPackets: this.#opusPackets,
+            rtpPackets: this.#applicationRtpPackets,
+            silencePackets: this.#silencePackets,
+            firstRtpAt: this.#rtp.firstAt,
+            lastRtpAt: this.#rtp.lastAt,
+            recentRtpPackets: this.#rtp.recent(),
+            queued: this.#packets.length,
+            pacerAlive: this.#pump !== undefined,
+        };
+    }
     createTrack() {
         const track = this.#track;
         const stop = track.stop;
         track.stop = () => {
             this.#stopped = true;
-            this.#packets.length = 0;
-            this.#pending = new Int16Array(0);
+            this.clear();
             this.#stopPump();
             stop();
         };
         return track;
     }
+    start() {
+        if (this.#started || this.#stopped)
+            return;
+        this.#started = true;
+        this.#startPump();
+    }
+    queuedMs() {
+        return this.#packets.length * WERIFT_PACKET_MS
+            + (this.#pending.length / WERIFT_CHANNEL_COUNT / WERIFT_SAMPLE_RATE) * 1_000;
+    }
+    waitForDrain() {
+        this.#flushPending();
+        if (this.#drained())
+            return Promise.resolve();
+        return new Promise((resolve) => this.#drainWaiters.add(resolve));
+    }
+    clear() {
+        this.#packets.length = 0;
+        this.#pending = new Int16Array(0);
+        if (!this.#started)
+            this.#stopPump();
+        this.#notifyDrained();
+    }
     onData(data) {
         if (this.#stopped)
             return;
@@ -109,21 +197,56 @@ class WeriftAudioSource {
         while (merged.length - offset >= PACKET_SAMPLES) {
             const frame = merged.subarray(offset, offset + PACKET_SAMPLES);
             this.#packets.push(Buffer.from(this.#encoder.encode(frame)));
+            this.#opusPackets += 1;
             offset += PACKET_SAMPLES;
         }
         this.#pending = merged.slice(offset);
         this.#startPump();
     }
+    /** Pad a sub-packet remainder with silence so the tail of a segment reaches the wire. */
+    #flushPending() {
+        if (this.#stopped || this.#pending.length === 0)
+            return;
+        const padded = new Int16Array(PACKET_SAMPLES);
+        padded.set(this.#pending);
+        this.#pending = new Int16Array(0);
+        this.#packets.push(Buffer.from(this.#encoder.encode(padded)));
+        this.#opusPackets += 1;
+        this.#startPump();
+    }
+    /** Application audio only: the silence the started pump keeps writing never counts. */
+    #drained() {
+        return this.#packets.length === 0
+            && this.#pending.length === 0
+            && (this.#started || this.#pump === undefined);
+    }
+    #notifyDrained() {
+        if (!this.#drained())
+            return;
+        for (const resolve of [...this.#drainWaiters]) {
+            this.#drainWaiters.delete(resolve);
+            resolve();
+        }
+    }
+    /**
+     * The first packet of a run leaves at once; every later one waits for the
+     * 20 ms tick, so a burst pushed faster than real time reaches the wire at
+     * packet cadence. Before `start()` the interval stops once the queue is
+     * empty; after it, the interval runs until the track stops.
+     */
     #startPump() {
-        if (this.#pump || this.#packets.length === 0)
+        if (this.#pump || this.#stopped)
             return;
-        this.#sendNext();
-        if (this.#packets.length === 0)
+        if (!this.#started && this.#packets.length === 0)
             return;
+        this.#sendNext();
         this.#pump = setInterval(() => {
             this.#sendNext();
-            if (this.#packets.length === 0)
-                this.#stopPump();
+            if (this.#packets.length === 0) {
+                if (!this.#started)
+                    this.#stopPump();
+                this.#notifyDrained();
+            }
         }, WERIFT_PACKET_MS);
         this.#pump.unref?.();
     }
@@ -134,8 +257,10 @@ class WeriftAudioSource {
         this.#pump = undefined;
     }
     #sendNext() {
-        const payload = this.#packets.shift();
-        if (!payload || this.#stopped)
+        if (this.#stopped)
+            return;
+        const application = this.#packets.shift();
+        if (!application && !this.#started)
             return;
         const header = new RtpHeader({
             payloadType: LOCAL_PAYLOAD_TYPE,
@@ -147,16 +272,35 @@ class WeriftAudioSource {
         this.#first = false;
         this.#sequenceNumber = (this.#sequenceNumber + 1) & 0xffff;
         this.#timestamp = (this.#timestamp + PACKET_FRAMES) >>> 0;
-        this.#track.writeRtp(new RtpPacket(header, payload));
+        this.#track.writeRtp(new RtpPacket(header, application ?? this.#silence));
+        if (application)
+            this.#applicationRtpPackets += 1;
+        else
+            this.#silencePackets += 1;
+        this.#rtp.mark();
     }
 }
-/** Opus RTP in, PCM16 (48 kHz stereo) out, one `ondata` per packet. */
+/** Opus RTP in, PCM16 in the requested format (default 48 kHz stereo) out, one `ondata` per packet. */
 class WeriftAudioSink {
     ondata = null;
-    #decoder = new Decoder({ channels: WERIFT_CHANNEL_COUNT, sample_rate: WERIFT_SAMPLE_RATE });
+    #decoder;
+    #rtp = new PacketClock();
+    #decodeFailures = 0;
     #unsubscribe;
-    constructor(track) {
+    stats() {
+        return {
+            rtpPackets: this.#rtp.count,
+            decodeFailures: this.#decodeFailures,
+            firstRtpAt: this.#rtp.firstAt,
+            lastRtpAt: this.#rtp.lastAt,
+            recentRtpPackets: this.#rtp.recent(),
+        };
+    }
+    constructor(track, format = { sampleRate: WERIFT_SAMPLE_RATE, channelCount: WERIFT_CHANNEL_COUNT }) {
+        const { sampleRate, channelCount } = format;
+        this.#decoder = new Decoder({ channels: channelCount, sample_rate: sampleRate });
         const { unSubscribe } = track.onReceiveRtp.subscribe((rtp) => {
+            this.#rtp.mark();
             const handler = this.ondata;
             if (!handler || !rtp.payload?.length)
                 return;
@@ -166,16 +310,17 @@ class WeriftAudioSink {
                 bytes = this.#decoder.decode(packet.payload);
             }
             catch {
+                this.#decodeFailures += 1;
                 return;
             }
             const samples = new Int16Array(bytes.byteLength >> 1);
             new Uint8Array(samples.buffer).set(bytes.subarray(0, samples.byteLength));
             handler({
                 samples,
-                sampleRate: WERIFT_SAMPLE_RATE,
+                sampleRate,
                 bitsPerSample: 16,
-                channelCount: WERIFT_CHANNEL_COUNT,
-                numberOfFrames: samples.length / WERIFT_CHANNEL_COUNT,
+                channelCount,
+                numberOfFrames: samples.length / channelCount,
             });
         });
         this.#unsubscribe = unSubscribe;
@@ -185,6 +330,15 @@ class WeriftAudioSink {
         this.#unsubscribe();
     }
 }
+/**
+ * Video codecs the peer can accept in a pull offer. Nothing decodes video: the
+ * transport only answers the person's `video` m-line receive-only and ignores
+ * the track (PROTOCOL.md section 5). werift matches a remote codec by MIME
+ * type alone (`findCodecByMimeType` in `TransceiverManager.setRemoteRTP`) and
+ * throws "negotiate codecs failed." for a media section with no local codec
+ * of its kind, so an audio-only codec list cannot answer a video m-line.
+ */
+const RECEIVE_VIDEO_MIME_TYPES = ["video/VP8", "video/VP9", "video/H264", "video/AV1"];
 export const createWeriftPeerConnection = (config = { iceServers: [], iceTransportPolicy: "all" }) => new RTCPeerConnection({
     bundlePolicy: "max-bundle",
     iceServers: config.iceServers,
@@ -197,11 +351,12 @@ export const createWeriftPeerConnection = (config = { iceServers: [], iceTranspo
                 channels: WERIFT_CHANNEL_COUNT,
             }),
         ],
+        video: RECEIVE_VIDEO_MIME_TYPES.map((mimeType) => new RTCRtpCodecParameters({ mimeType, clockRate: 90_000 })),
     },
 });
 export const createWeriftWebRTCFactory = () => ({
     createPeerConnection: (config) => createWeriftPeerConnection(config),
     createAudioSource: () => new WeriftAudioSource(),
-    createAudioSink: (track) => new WeriftAudioSink(track),
+    createAudioSink: (track, format) => new WeriftAudioSink(track, format),
 });
 //# sourceMappingURL=engine-werift.js.map

package/dist/engine-werift.js.map

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

@@ -1,4 +1,4 @@
-export { RelayAudioInput, RelayAudioOutput, RelayLiveKitCall, createRelayLiveKitAudio, type RelayLiveKitAudio, type RelayLiveKitAudioOptions, type RelayLiveKitConnectOptions, } from "./livekit.js";
-export { RelayCallTransport, RelayCallTransportError, type RelayAudioFrame, type RelayCallEngine, type RelayCallIceDiagnostics, type RelayCallTransportCloseEvent, type RelayCallTransportOptions, type RelayIceServer, type RelayIceTransportPolicy, } from "./transport.js";
+export { RelayAudioInput, RelayAudioOutput, LIVEKIT_ROOM_INPUT_AUDIO, RelayLiveKitCall, createRelayLiveKitAudio, type RelayLiveKitAudio, type RelayLiveKitAudioOptions, type RelayLiveKitConnectOptions, } from "./livekit.js";
+export { RelayCallTransport, RelayCallTransportError, type RelayAudioFrame, type RelayCallEngine, type RelayCallIceDiagnostics, type RelayCallInboundDiagnostics, type RelayCallOutboundDiagnostics, type RelayCallRestartEvent, type RelayCallRestartReason, type RelayCallRoomDiagnostics, type RelayCallTransportCloseEvent, type RelayCallTransportOptions, type RelayIceServer, type RelayIceServersProvider, type RelayIceTransportPolicy, type RelayInboundAudioFormat, } from "./transport.js";
 export { createWeriftWebRTCFactory } from "./engine-werift.js";
 //# sourceMappingURL=index.d.ts.map

package/dist/index.d.ts.map

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

@@ -1,4 +1,4 @@
-export { RelayAudioInput, RelayAudioOutput, RelayLiveKitCall, createRelayLiveKitAudio, } from "./livekit.js";
+export { RelayAudioInput, RelayAudioOutput, LIVEKIT_ROOM_INPUT_AUDIO, RelayLiveKitCall, createRelayLiveKitAudio, } from "./livekit.js";
 export { RelayCallTransport, RelayCallTransportError, } from "./transport.js";
 export { createWeriftWebRTCFactory } from "./engine-werift.js";
 //# sourceMappingURL=index.js.map

package/dist/index.js.map

@@ -1 +1 @@
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+{"version":3,"file":"index.js","sourceRoot":"","sources":["../src/index.ts"],"names":[],"mappings":"AAAA,OAAO,EACL,eAAe,EACf,gBAAgB,EAChB,wBAAwB,EACxB,gBAAgB,EAChB,uBAAuB,GAIxB,MAAM,cAAc,CAAC;AACtB,OAAO,EACL,kBAAkB,EAClB,uBAAuB,GAexB,MAAM,gBAAgB,CAAC;AACxB,OAAO,EAAE,yBAAyB,EAAE,MAAM,oBAAoB,CAAC"}

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;

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