@relaymessenger/livekit 0.1.0-staging.15 → 0.1.0-staging.17

package/README.md

@@ -1,10 +1,10 @@
 # `@relaymessenger/livekit`
 
 `@relaymessenger/livekit` connects Relay Calls to TypeScript LiveKit Agents.
-Relay stays responsible for the Call resource and signaling; the package owns
-the Node WebRTC peer and translates audio between Relay and LiveKit `AudioFrame`
-objects, and sends and receives video with LiveKit's video API. Application code does not handle Cloudflare, SDP, ICE, or SFU
-credentials.
+It translates audio between the Relay SDK's call transport
+(`RelayCallTransport` from `@relaymessenger/sdk/calls`) and LiveKit
+`AudioFrame` objects. The transport owns the Node WebRTC peer, so application
+code does not handle Cloudflare, SDP, ICE, or SFU credentials.
 
 Install it next to the Relay SDK and the LiveKit Agents runtime:
 
@@ -43,197 +43,26 @@ 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
+`RelayAudioOutput` has the shape of LiveKit's own `ParticipantAudioOutput`.
+Like LiveKit's, it waits until the person is receiving the agent's audio: the
+transport holds each frame until then, so a greeting that starts early is not
+cut, sends silence meanwhile, and skips nothing. Then `captureFrame()` hands each
+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,
 `call.transport.reconnect()` keeps the existing media peer and replays the same
 audio publication so Relay can return its cached answer.
 
-## Provider-neutral transport
-
-The WebRTC/PCM layer is a separate public entry point for future adapters that
-do not use LiveKit Agents:
-
-```ts
-import { RelayCallTransport } from "@relaymessenger/livekit/transport";
-
-const room = relay.calls.room(callId);
-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({
-  samples,
-  sampleRate: 48_000,
-  channelCount: 1,
-});
-await transport.waitForPlayout();
-```
-
-## Video
-
-Relay calls carry video whenever a camera is on. The video API copies
-LiveKit's `@livekit/rtc-node` names: `VideoSource`, `LocalVideoTrack`,
-`VideoFrame`, `VideoBufferType`, `VideoStream`. Video needs the default
-`"werift"` engine and the optional dependency `node-webcodecs` (prebuilt for
-macOS arm64 and Linux x64/arm64).
-
-Send a video feed:
-
-```ts
-import {
-  LocalVideoTrack,
-  VideoBufferType,
-  VideoFrame,
-  VideoSource,
-} from "@relaymessenger/livekit/transport";
-
-await transport.connect();
-
-const source = new VideoSource(640, 480);
-const track = LocalVideoTrack.createVideoTrack("camera", source);
-await transport.publishTrack(track, {
-  videoEncoding: { maxBitrate: 800_000, maxFramerate: 15 },
-});
-
-// RGBA, BGRA or I420 bytes, tightly packed.
-source.captureFrame(new VideoFrame(rgba, 640, 480, VideoBufferType.RGBA));
-
-// Camera off, then on again; the track stays negotiated.
-await transport.unpublishTrack(track);
-await transport.publishTrack(track);
-```
-
-Receive the other participant's video:
-
-```ts
-import { VideoBufferType, VideoStream } from "@relaymessenger/livekit/transport";
-
-transport.on("trackSubscribed", async (track) => {
-  const stream = new VideoStream(track, { format: VideoBufferType.RGBA, capacity: 2 });
-  for await (const { frame, timestampUs } of stream) {
-    // frame.data is width x height x 4 bytes of RGBA.
-  }
-});
-transport.on("remoteVideo", (on) => {
-  // The other participant's camera started or stopped sending.
-});
-```
-
-Frames are decoded only while a `VideoStream` is open. `transport.videoStats()`
-reports frames, packets, keyframes and decode errors in both directions.
-
-## ICE servers, TURN, and diagnostics
-
-By default the peer uses the servers the Call room sends after it joins:
-Cloudflare's STUN server and TURN credentials that Relay mints for the call.
-An agent in a container or behind a firewall that blocks outbound UDP connects
-through TURN with no setup. Each restart uses the room's latest servers. A room
-that sends none falls back to Cloudflare's STUN server
-(`stun:stun.cloudflare.com:3478`), as Cloudflare's own Realtime echo example
-does.
-
-The offer leaves as soon as the first local candidate exists, without waiting
-for ICE gathering to finish: the SFU is ICE-lite and learns the agent's address
-from its connectivity checks, TURN relay checks included.
-
-To use your own servers instead, pass `iceServers` in the standard
-`RTCIceServer` shape and, if every path must go through TURN,
-`iceTransportPolicy: "relay"`. Your value replaces the room's. Both options are
-accepted by `RelayLiveKitCall.connect()` and `RelayCallTransport`:
-
-```ts
-const call = await RelayLiveKitCall.connect({
-  relay,
-  callId,
-  iceServers: [
-    {
-      urls: [
-        "turn:turn.example.com:3478?transport=udp",
-        "turn:turn.example.com:3478?transport=tcp",
-        "turns:turn.example.com:5349?transport=tcp",
-      ],
-      username: process.env.TURN_USERNAME!,
-      credential: process.env.TURN_CREDENTIAL!,
-    },
-  ],
-  iceTransportPolicy: "all",
-});
-```
-
-`iceServers` may also be a function; it is called before every peer
-connection, so it can mint fresh TURN credentials for each restart. The room's
-servers are on `room.iceServers` and its `iceServers` event.
-
-`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()`, 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,
-paced by the monotonic clock: a timer that fires late sends every packet due
-by then, so the wire carries exactly 50 packets a second; a stall longer than
-200 ms restarts the clock instead of bursting. The track carries
-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
-`@evan/opus` (prebuilt Opus, WASM fallback), so no native WebRTC binding is
-loaded. Pass `engine: "wrtc"` to use the optional `@roamhq/wrtc` binding
-instead. The transport boundary remains provider-neutral for additional adapters. `@livekit/agents`
-and `@livekit/rtc-node` are peer dependencies so the host agent process owns
-those runtimes.
+`RelayLiveKitCall.connect()` accepts the transport's `iceServers`,
+`iceTransportPolicy` and `sessionConnectTimeoutMs`; `call.waitForPeerAudio()`
+and `call.diagnostics()` read the transport. Video, ICE servers, restarts and
+diagnostics are documented with the transport in the
+[`@relaymessenger/sdk` README](https://github.com/RelayMessenger/Relay-SDK/tree/main/packages/sdk#join-a-call-as-the-agent).
+`@livekit/agents` and `@livekit/rtc-node` are peer dependencies so the host
+agent process owns those runtimes.

package/dist/index.d.ts

@@ -1,5 +1,2 @@
 export { RelayAudioInput, RelayAudioOutput, LIVEKIT_ROOM_INPUT_AUDIO, RelayLiveKitCall, createRelayLiveKitAudio, type RelayLiveKitAudio, type RelayLiveKitAudioOptions, type RelayLiveKitConnectOptions, } from "./livekit.js";
-export { RelayCallTransport, RelayCallTransportError, type RelayAudioFrame, type RelayCallVideoStats, 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";
-export { LocalVideoTrack, RemoteVideoTrack, VideoBufferType, VideoCodec, VideoFrame, VideoRotation, VideoSource, VideoStream, type RelayVideoReceiverStats, type RelayVideoSenderStats, type TrackPublishOptions, type VideoEncoding, type VideoFrameEvent, type VideoStreamOptions, } from "./video.js";
 //# sourceMappingURL=index.d.ts.map

package/dist/index.d.ts.map

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

@@ -1,5 +1,2 @@
 export { RelayAudioInput, RelayAudioOutput, LIVEKIT_ROOM_INPUT_AUDIO, RelayLiveKitCall, createRelayLiveKitAudio, } from "./livekit.js";
-export { RelayCallTransport, RelayCallTransportError, } from "./transport.js";
-export { createWeriftWebRTCFactory } from "./engine-werift.js";
-export { LocalVideoTrack, RemoteVideoTrack, VideoBufferType, VideoCodec, VideoFrame, VideoRotation, VideoSource, VideoStream, } from "./video.js";
 //# sourceMappingURL=index.js.map

package/dist/index.js.map

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

@@ -2,7 +2,7 @@ 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 RelayIceServersProvider, type RelayInboundAudioFormat, type RelayIceTransportPolicy } from "./transport.js";
+import { RelayCallTransport, type RelayCallIceDiagnostics, type RelayIceServer, type RelayIceServersProvider, type RelayInboundAudioFormat, type RelayIceTransportPolicy } from "@relaymessenger/sdk/calls";
 /**
  * The format LiveKit's own room input hands an AgentSession:
  * `@livekit/agents` dist/voice/room_io/room_io.js:46-48
@@ -34,6 +34,12 @@ export declare class RelayAudioInput extends AudioInput {
 export declare class RelayAudioOutput extends AudioOutput {
     #private;
     constructor(transport: RelayCallTransport, sampleRate?: number);
+    /**
+     * Hands the frame to the transport, which holds it until the person receives
+     * the agent's audio (PROTOCOL.md section 6b) and resolves once it is queued,
+     * so this waits as LiveKit's room audio output waits for the track's
+     * subscription. A frame dropped by `clearBuffer()` meanwhile is not counted.
+     */
     captureFrame(frame: AudioFrame): Promise<void>;
     /** Mark the segment complete; `onPlaybackFinished` fires once the transport has drained. */
     flush(): void;

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package/dist/livekit.js

@@ -1,7 +1,7 @@
 import { AudioInput, AudioOutput, Future } from "@livekit/agents";
 import { AudioFrame } from "@livekit/rtc-node";
 import { TransformStream } from "node:stream/web";
-import { RelayCallTransport, } from "./transport.js";
+import { RelayCallTransport, } from "@relaymessenger/sdk/calls";
 /**
  * The format LiveKit's own room input hands an AgentSession:
  * `@livekit/agents` dist/voice/room_io/room_io.js:46-48
@@ -80,11 +80,19 @@ export class RelayAudioOutput extends AudioOutput {
     #flushDone = true;
     /** Resolved by `clearBuffer()` with the milliseconds still queued at that moment. */
     #interruptedFuture = new Future();
+    /** Bumped by `clearBuffer()`; a frame the transport dropped across it is not counted (agents-js `interruptionGeneration`). */
+    #interruptionGeneration = 0;
     #closed = false;
     constructor(transport, sampleRate = 48_000) {
         super(sampleRate, undefined, { pause: false });
         this.#transport = transport;
     }
+    /**
+     * Hands the frame to the transport, which holds it until the person receives
+     * the agent's audio (PROTOCOL.md section 6b) and resolves once it is queued,
+     * so this waits as LiveKit's room audio output waits for the track's
+     * subscription. A frame dropped by `clearBuffer()` meanwhile is not counted.
+     */
     async captureFrame(frame) {
         if (this.#closed)
             throw new Error("Relay LiveKit audio output is closed.");
@@ -92,18 +100,21 @@ export class RelayAudioOutput extends AudioOutput {
             this.logger.error("captureFrame called while flush is in progress");
             await this.#flushTask;
         }
-        await super.captureFrame(frame);
-        if (!this.#firstFrameEmitted) {
-            this.#firstFrameEmitted = true;
-            this.onPlaybackStarted(Date.now());
-        }
-        this.#pushedDuration += frame.samplesPerChannel / frame.sampleRate;
+        const interruptionGeneration = this.#interruptionGeneration;
         // 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,
         });
+        if (interruptionGeneration !== this.#interruptionGeneration)
+            return;
+        await super.captureFrame(frame);
+        if (!this.#firstFrameEmitted) {
+            this.#firstFrameEmitted = true;
+            this.onPlaybackStarted(Date.now());
+        }
+        this.#pushedDuration += frame.samplesPerChannel / frame.sampleRate;
     }
     /** Mark the segment complete; `onPlaybackFinished` fires once the transport has drained. */
     flush() {
@@ -119,6 +130,7 @@ export class RelayAudioOutput extends AudioOutput {
         void this.#flushTask.catch(() => undefined);
     }
     clearBuffer() {
+        this.#interruptionGeneration += 1;
         const queuedMs = this.#transport.queuedAudioMs();
         this.#transport.clearAudio();
         if (this.#interruptedFuture.done)

package/dist/livekit.js.map

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package/package.json

@@ -1,7 +1,7 @@
 {
   "name": "@relaymessenger/livekit",
-  "version": "0.1.0-staging.15",
-  "description": "Relay WebRTC audio and video transport and LiveKit Agents integration for Node.js.",
+  "version": "0.1.0-staging.17",
+  "description": "LiveKit Agents audio input and output for Relay Calls, over @relaymessenger/sdk/calls.",
   "type": "module",
   "sideEffects": false,
   "main": "./dist/index.js",
@@ -12,11 +12,6 @@
       "import": "./dist/index.js",
       "default": "./dist/index.js"
     },
-    "./transport": {
-      "types": "./dist/transport.d.ts",
-      "import": "./dist/transport.js",
-      "default": "./dist/transport.js"
-    },
     "./package.json": "./package.json"
   },
   "files": [
@@ -38,7 +33,7 @@
   "license": "MIT",
   "dependencies": {
     "@evan/opus": "1.0.3",
-    "@relaymessenger/sdk": "0.3.6-staging.22",
+    "@relaymessenger/sdk": "0.3.6-staging.24",
     "rtp-packet": "0.4.2",
     "werift": "0.24.4"
   },

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

@@ -1,70 +0,0 @@
-/**
- * Video for the werift engine: `node-webcodecs` 1.3.0 encodes and decodes
- * (its static build: LGPL FFmpeg with openh264 and libvpx), `rtp-packet`
- * packetizes, `video-rtp.ts` depacketizes, werift carries RTP and RTCP.
- *
- * Choices measured on 2026-09-22 (/tmp/video-spike-webcodecs, the spike that
- * picked this engine), each kept here:
- * - `NODE_WEBCODECS_FORCE=static` selects the bundled FFmpeg
- *   (node-webcodecs dist/native.js `load()`; the default order tries a
- *   dynamic build linked against system FFmpeg first).
- * - `hardwareAcceleration: "prefer-software"` on encoder and decoder: on a
- *   Mac, VideoToolbox otherwise takes the stream.
- * - H.264 is sent as constrained baseline `avc1.42e01f`, Annex-B; the codec
- *   string FFmpeg reports for H.264 is "MPEG4/ISO/AVC", so the codec is never
- *   matched by FFmpeg's name, only by the negotiated MIME type.
- * - A keyframe every 1 s, and on every PLI or FIR from the SFU.
- * - The send codec is the first codec of the SFU's answer (werift
- *   rtpSender.js:408 `this.codec = params.codecs[0]`).
- * - The receive codec is learned from the payload type of the first RTP
- *   packet, looked up in the pulled m-line's codecs: the SFU's pull offer
- *   lists every codec (spike receiver.mjs).
- * - werift sends no PLI on loss by itself: the receiver sends one when the
- *   stream starts, when a frame is dropped and on a decode error, then every
- *   1 s until a keyframe decodes (spike receiver.mjs `pliTimer`).
- */
-import { MediaStreamTrack, type RTCRtpTransceiver } from "werift";
-import type { RelayMediaStreamTrackLike } from "./transport.js";
-import { type RelayVideoFactory, type RelayVideoReceiverLike, type RelayVideoReceiverStats, type RelayVideoSenderLike, type RelayVideoSenderStats, type TrackPublishOptions } from "./video.js";
-import { VideoFrame, VideoRotation } from "./video-frame.js";
-type WebCodecs = typeof import("node-webcodecs");
-export declare const KEYFRAME_INTERVAL_MS = 1000;
-export declare const KEYFRAME_REQUEST_INTERVAL_MS = 1000;
-export declare const DEFAULT_VIDEO_BITRATE = 800000;
-export declare const DEFAULT_VIDEO_FRAMERATE = 30;
-/** Loads node-webcodecs once, on first video use, so audio-only calls never load FFmpeg. */
-export declare const loadWebCodecs: () => Promise<WebCodecs>;
-/**
- * Raw frames in, H.264 or VP8 RTP out on one werift track. One sender lives
- * for the whole call; a restart binds it to the new peer's transceiver.
- */
-export declare class WeriftVideoSender implements RelayVideoSenderLike {
-    #private;
-    readonly track: RelayMediaStreamTrackLike;
-    constructor(wc: WebCodecs, options?: TrackPublishOptions);
-    /**
-     * Offer the codecs this sender can encode, the preferred one first (werift
-     * fills an empty `transceiver.codecs` from the peer's config at
-     * `createOffer`, peerConnection.js:520-523, so setting it here wins).
-     */
-    bind(transceiver: unknown, peer: unknown): void;
-    setEnabled(enabled: boolean): void;
-    capture(frame: VideoFrame, timestampUs: bigint, rotation: VideoRotation): void;
-    stats(): RelayVideoSenderStats;
-    close(): void;
-}
-/**
- * One pulled video m-line: RTP in, decoded I420 frames out. RTP is counted
- * always; it is reordered, assembled and decoded only while active.
- */
-export declare class WeriftVideoReceiver implements RelayVideoReceiverLike {
-    #private;
-    onframe: RelayVideoReceiverLike["onframe"];
-    constructor(track: MediaStreamTrack, transceiver: RTCRtpTransceiver | undefined);
-    setActive(active: boolean): void;
-    stats(): RelayVideoReceiverStats;
-    stop(): void;
-}
-export declare const createWeriftVideoFactory: () => RelayVideoFactory;
-export {};
-//# sourceMappingURL=engine-werift-video.d.ts.map

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

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