@komaa/standin-sdk 0.1.0

package/dist/plugins/livekit/room.js

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+// Copyright (c) 2026 Komaa DigiTech
+// SPDX-License-Identifier: MIT
+/**
+ * One LiveKit room per call, joined from your worker.
+ *
+ * The shape is the opposite of the other plugins. The providers are a
+ * socket you send audio to; LiveKit is a room you JOIN, publishing the caller's
+ * voice as a track and subscribing to the agent's. Your agent does not know it
+ * is on a Microsoft Teams call at all: it sees a participant who talks.
+ *
+ * The framework is imported lazily, inside the function that needs it, which is
+ * the rule that lets one package ship every plugin. `import { CallServer }
+ * from "@komaa/standin-sdk"` must keep working on a machine with no LiveKit
+ * installed, so nothing here is imported at module load.
+ *
+ * The Python twin is `standin.plugins.livekit.call`, and the two data
+ * topics below are deliberately the same strings in both languages: an agent
+ * written for one works with the other.
+ */
+import { StandInError } from "../../errors.js";
+import { logger } from "../../log.js";
+import { NUM_CHANNELS, SAMPLE_RATE_HZ } from "../../protocol.js";
+/** Non-interrupting call context, published into the room as data. */
+export const TOPIC_CONTEXT = "msteams.context";
+/** The closing line StandIn wants spoken before the call ends. */
+export const TOPIC_GOODBYE = "msteams.goodbye";
+/** LiveKit's own transcription topic. Both sides of the call publish on it. */
+export const TOPIC_TRANSCRIPTION = "lk.transcription";
+/** Which track a transcript stream is OF. Absent on some publishers. */
+export const ATTRIBUTE_TRANSCRIBED_TRACK_ID = "lk.transcribed_track_id";
+/** Marks the closing header of a finished turn. */
+export const ATTRIBUTE_TRANSCRIPTION_FINAL = "lk.transcription_final";
+/** The identity this plugin joins under. */
+export const LOCAL_IDENTITY = "standin-msteams";
+/**
+ * Whether this transcript is of the CALLER rather than of the agent.
+ *
+ * The track id is exact when present. Without it, the sender identity is the
+ * fallback: livekit publishes a participant's transcript under that
+ * participant's identity, and ours is the caller's audio.
+ */
+export function isCallerTranscript(attributes, senderIdentity, route) {
+    const trackId = attributes?.[ATTRIBUTE_TRANSCRIBED_TRACK_ID];
+    if (trackId !== undefined && trackId !== "") {
+        return (route.callerTrackSid !== undefined && trackId === route.callerTrackSid);
+    }
+    if (senderIdentity === undefined)
+        return false;
+    if (senderIdentity === route.localIdentity)
+        return true;
+    // Once the agent is bound, anything that is not the agent is the caller.
+    return (route.agentIdentity !== undefined && senderIdentity !== route.agentIdentity);
+}
+/**
+ * Read one transcript stream to its end and report the caller's words.
+ *
+ * Every stream is drained, including one this call ignores: an abandoned reader
+ * keeps its subscription for the life of the process, and the agent publishes
+ * one stream per turn.
+ *
+ * Each yield is the cumulative transcript so far, so every yield is delivered as
+ * a partial and the last one as final. A partial exists so a wake phrase is
+ * noticed as it is said; only a finished turn is a turn.
+ */
+export async function readTranscript(reader, senderIdentity, route) {
+    const mine = isCallerTranscript(reader.info?.attributes, senderIdentity, {
+        callerTrackSid: route.callerTrackSid(),
+        localIdentity: route.localIdentity,
+        agentIdentity: route.agentIdentity(),
+    });
+    let latest = "";
+    try {
+        for await (const chunk of reader) {
+            if (route.isClosed())
+                return;
+            latest = chunk;
+            // Drained either way, but only the caller's words are reported.
+            if (mine && latest !== "")
+                await route.deliver(latest, false);
+        }
+    }
+    catch (err) {
+        logger.debug(`standin: a transcript stream ended early: ${String(err)}`);
+        return;
+    }
+    if (mine && latest !== "" && !route.isClosed())
+        await route.deliver(latest, true);
+}
+/**
+ * What goes on a topic: a JSON object carrying `text`, never the bare string.
+ *
+ * This is a cross-language contract. The Python SDK's `TeamsCall._on_data`
+ * json-decodes the packet and requires a dict with a `text` string, so a bare
+ * string is dropped in silence at the far end: every context sentence and the
+ * goodbye would reach nothing.
+ */
+export function contextPayload(text) {
+    return JSON.stringify({ text });
+}
+/** How long a room token lives. Longer than any call, shorter than forever. */
+const TOKEN_TTL = "6h";
+/**
+ * Load the framework at USE time, never at module load.
+ *
+ * A missing framework must name the install that fixes it. A reader who gets
+ * `Cannot find module '@livekit/rtc-node'` out of somebody else's package
+ * cannot tell a missing optional dependency from a broken install.
+ */
+async function loadLiveKit() {
+    try {
+        const [rtc, server] = await Promise.all([
+            import("@livekit/rtc-node"),
+            import("livekit-server-sdk"),
+        ]);
+        return { rtc, server };
+    }
+    catch (err) {
+        logger.debug(`standin: loading LiveKit failed: ${String(err)}`);
+        throw new StandInError("the livekit plugin needs @livekit/rtc-node and livekit-server-sdk: " +
+            "run `npm install @livekit/rtc-node livekit-server-sdk`");
+    }
+}
+/**
+ * Join a room for one call, publishing the caller and subscribing to the agent.
+ *
+ * `metadata` is handed to the dispatched agent as job metadata, which is how an
+ * agent learns who is calling without this plugin inventing a side channel.
+ */
+/** The control-plane form of a room URL. The websocket one is not an API base. */
+function httpUrl(url) {
+    const trimmed = url.trim();
+    if (trimmed.startsWith("wss://"))
+        return `https://${trimmed.slice(6)}`;
+    if (trimmed.startsWith("ws://"))
+        return `http://${trimmed.slice(5)}`;
+    return trimmed;
+}
+export async function connectRoom(config, callId, metadata, handlers) {
+    const { rtc, server } = await loadLiveKit();
+    // The call id arrives from a decoded URL segment, so a "%2F" would smuggle a
+    // slash into the room name. Keep it to a safe charset and a bounded length.
+    const safeCallId = callId.replace(/[^A-Za-z0-9._@:-]/g, "-");
+    const roomName = `${config.roomPrefix}${safeCallId}`.slice(0, 100);
+    const token = new server.AccessToken(config.apiKey, config.apiSecret, {
+        identity: "standin-msteams",
+        ttl: TOKEN_TTL,
+    });
+    token.addGrant({
+        roomJoin: true,
+        room: roomName,
+        canPublish: true,
+        canSubscribe: true,
+        canPublishData: true,
+    });
+    if (config.agentName) {
+        token.roomConfig = new server.RoomConfiguration({
+            agents: [
+                new server.RoomAgentDispatch({
+                    agentName: config.agentName,
+                    metadata: JSON.stringify(metadata),
+                }),
+            ],
+        });
+    }
+    const room = new rtc.Room();
+    await room.connect(config.url, await token.toJwt(), {
+        autoSubscribe: true,
+        dynacast: false,
+    });
+    const local = room.localParticipant;
+    if (!local) {
+        try {
+            await room.disconnect();
+        }
+        catch {
+            // already closing
+        }
+        throw new StandInError("the room connected without a local participant");
+    }
+    logger.info(`standin: joined LiveKit room "${roomName}"` +
+        (config.agentName ? ` and dispatched "${config.agentName}"` : ""));
+    const source = new rtc.AudioSource(SAMPLE_RATE_HZ, NUM_CHANNELS);
+    const track = rtc.LocalAudioTrack.createAudioTrack("msteams-caller", source);
+    // The publication is KEPT, and its sid is read at use time. The SDK re-issues
+    // the sid in place after a reconnect, so a cached string stops matching and
+    // every caller transcript is then classified as the agent's own speech, which
+    // takes the wake-phrase path silently dead for the rest of the call.
+    const callerPublication = await local.publishTrack(track, new rtc.TrackPublishOptions({ source: rtc.TrackSource.SOURCE_MICROPHONE }));
+    let closed = false;
+    // The identity whose audio we relay is "the agent". Captured on the first
+    // audio subscription, and only THAT identity leaving ends the call: a monitor
+    // or a second participant dropping out must not hang up on the caller.
+    let agentIdentity;
+    // Any value other than auto or off names the participant whose video the tile
+    // relay should take, rather than letting the room decide.
+    const pinnedIdentity = config.tileVideo === "auto" || config.tileVideo === "off"
+        ? undefined
+        : config.tileVideo;
+    // One live pump at a time, reset when the stream ends, so an agent that
+    // republishes its audio (an avatar track swap, a mute cycle) gets pumped
+    // again instead of going silent for the rest of the call.
+    let activePumpSid;
+    const startPump = (remote, identity) => {
+        if (activePumpSid !== undefined)
+            return;
+        activePumpSid = remote.sid ?? "unknown";
+        void (async () => {
+            try {
+                // Ask for 16 kHz mono: the framework resamples, so our side stays a copy.
+                const stream = new rtc.AudioStream(remote, SAMPLE_RATE_HZ, NUM_CHANNELS);
+                for await (const frame of stream) {
+                    if (closed)
+                        break;
+                    const pcm = Buffer.from(frame.data.buffer, frame.data.byteOffset, frame.data.length * 2);
+                    await handlers.onAgentAudio(Buffer.from(pcm));
+                }
+            }
+            catch (err) {
+                if (!closed)
+                    logger.warn(`standin: the LiveKit audio pump failed: ${String(err)}`);
+            }
+            finally {
+                activePumpSid = undefined;
+                logger.debug(`standin: the audio pump for "${identity}" ended`);
+            }
+        })();
+    };
+    if (handlers.onCallerTranscript !== undefined) {
+        // LiveKit publishes transcripts of BOTH sides on this one topic. Telling
+        // them apart matters: an agent greeting that says its own name would
+        // otherwise open the follow-up window and make the assistant answer the
+        // next turn of a meeting nobody addressed it in.
+        try {
+            room.registerTextStreamHandler(TOPIC_TRANSCRIPTION, (reader, info) => {
+                void readTranscript(reader, info?.identity, {
+                    callerTrackSid: () => callerPublication?.sid,
+                    localIdentity: LOCAL_IDENTITY,
+                    agentIdentity: () => agentIdentity,
+                    isClosed: () => closed,
+                    deliver: handlers.onCallerTranscript,
+                });
+            });
+        }
+        catch (err) {
+            logger.warn(`standin: caller transcripts are unavailable on this room: ${String(err)}`);
+        }
+    }
+    room.on(rtc.RoomEvent.TrackSubscribed, (remote, _publication, participant) => {
+        if (remote.kind !== rtc.TrackKind.KIND_AUDIO)
+            return;
+        if (agentIdentity === undefined) {
+            agentIdentity = participant.identity;
+            handlers.onAnswered?.();
+        }
+        startPump(remote, participant.identity);
+    });
+    room.on(rtc.RoomEvent.TrackUnsubscribed, (remote) => {
+        if (remote.sid !== undefined && remote.sid === activePumpSid)
+            activePumpSid = undefined;
+    });
+    room.on(rtc.RoomEvent.ParticipantDisconnected, (participant) => {
+        if (agentIdentity !== undefined && participant.identity === agentIdentity) {
+            void handlers.onClosed(`the agent ${participant.identity} disconnected`);
+        }
+    });
+    // Disconnected is FINAL: the framework retries transient drops internally
+    // before this fires.
+    room.on(rtc.RoomEvent.Disconnected, () => {
+        void handlers.onClosed("the room disconnected");
+    });
+    /**
+     * Relay the avatar's video onto the bot's tile.
+     *
+     * The participant is chosen by LiveKit's own publish-on-behalf attribute
+     * first, because an avatar worker publishes video as a SEPARATE participant
+     * acting for the agent, and only then by the agent's own identity.
+     *
+     * Tracks are matched by KIND, never by source: an avatar worker publishes its
+     * video untagged, so it arrives as an unknown source rather than a camera. A
+     * source filter would pick the right participant and then stream nothing,
+     * which is the most expensive way to get this wrong.
+     */
+    const startAvatarRelay = async (sink) => {
+        let stopped = false;
+        let activeSid;
+        let activeReader;
+        const cancelActive = () => {
+            const reader = activeReader;
+            activeReader = undefined;
+            activeSid = undefined;
+            // Cancel through the LOCK HOLDER. A VideoStream is a ReadableStream, the
+            // drain loop's reader locks it, and cancelling the stream itself rejects
+            // while it is locked. Cancelling the reader also unblocks a parked read,
+            // which a quiet or swapped track would otherwise never wake.
+            if (reader)
+                void reader.cancel().catch(() => undefined);
+        };
+        const drain = (track, identity) => {
+            cancelActive();
+            const sid = track.sid ?? "unknown";
+            activeSid = sid;
+            logger.info(`standin: relaying avatar video from "${identity}"`);
+            void (async () => {
+                const stream = new rtc.VideoStream(track);
+                const reader = stream.getReader();
+                activeReader = reader;
+                try {
+                    for (;;) {
+                        const { done, value } = await reader.read();
+                        if (done || stopped)
+                            break;
+                        const rgb = value.frame.convert(rtc.VideoBufferType.RGB24);
+                        sink.offerRgb(Buffer.from(rgb.data.buffer, rgb.data.byteOffset, rgb.data.length), rgb.width, rgb.height);
+                    }
+                }
+                catch (err) {
+                    if (!stopped)
+                        logger.warn(`standin: the avatar video stream ended: ${String(err)}`);
+                }
+                finally {
+                    reader.releaseLock();
+                    if (activeReader === reader)
+                        activeReader = undefined;
+                    if (activeSid === sid)
+                        activeSid = undefined;
+                }
+            })();
+        };
+        const pickParticipant = () => {
+            const remotes = [...room.remoteParticipants.values()];
+            // A pinned identity wins outright. It is set precisely when a separate
+            // worker publishes the avatar and publish-on-behalf is NOT, in which case
+            // guessing relays whichever participant published first.
+            if (pinnedIdentity !== undefined) {
+                return remotes.find((p) => p.identity === pinnedIdentity);
+            }
+            const behalf = remotes.find((p) => p.attributes?.["lk.publish_on_behalf"] === agentIdentity);
+            if (behalf)
+                return behalf;
+            return remotes.find((p) => p.identity === agentIdentity);
+        };
+        const startFromExisting = () => {
+            if (stopped || activeSid !== undefined)
+                return;
+            const chosen = pickParticipant();
+            if (!chosen)
+                return;
+            for (const pub of chosen.trackPublications.values()) {
+                if (pub.kind === rtc.TrackKind.KIND_VIDEO && pub.track) {
+                    drain(pub.track, chosen.identity);
+                    return;
+                }
+            }
+        };
+        const onSubscribed = (track, _pub, participant) => {
+            if (stopped || activeSid !== undefined)
+                return;
+            if (track.kind === rtc.TrackKind.KIND_AUDIO) {
+                // The audio subscribe is what binds the agent identity. The chosen
+                // participant's video may already be subscribed, and that event will
+                // not fire again, so re-scan here or the relay never starts.
+                startFromExisting();
+                return;
+            }
+            if (track.kind !== rtc.TrackKind.KIND_VIDEO)
+                return;
+            const chosen = pickParticipant();
+            if (!chosen || chosen.identity !== participant.identity)
+                return;
+            drain(track, participant.identity);
+        };
+        const onUnsubscribed = (track) => {
+            if (track.sid !== undefined && track.sid === activeSid)
+                cancelActive();
+        };
+        room.on(rtc.RoomEvent.TrackSubscribed, onSubscribed);
+        room.on(rtc.RoomEvent.TrackUnsubscribed, onUnsubscribed);
+        startFromExisting();
+        return () => {
+            stopped = true;
+            room.off(rtc.RoomEvent.TrackSubscribed, onSubscribed);
+            room.off(rtc.RoomEvent.TrackUnsubscribed, onUnsubscribed);
+            cancelActive();
+        };
+    };
+    return {
+        startAvatarRelay,
+        get isOpen() {
+            return !closed;
+        },
+        get agentIdentity() {
+            return agentIdentity;
+        },
+        sendCallerAudio(pcm) {
+            if (closed || pcm.length === 0)
+                return;
+            // Copied into a fresh, aligned buffer rather than viewed in place. A
+            // Buffer from Node's pool can sit at an ODD byteOffset, and an Int16Array
+            // view there does not read wrong, it throws RangeError and kills the
+            // call. Predicting the allocator is not worth one memcpy per frame.
+            const aligned = new ArrayBuffer(pcm.length);
+            Buffer.from(aligned).set(pcm);
+            const samples = new Int16Array(aligned);
+            const frame = new rtc.AudioFrame(samples, SAMPLE_RATE_HZ, NUM_CHANNELS, samples.length / NUM_CHANNELS);
+            void source.captureFrame(frame).catch((err) => {
+                if (!closed)
+                    logger.warn(`standin: publishing caller audio failed: ${String(err)}`);
+            });
+        },
+        async publish(topic, payload) {
+            if (closed)
+                return;
+            try {
+                await local.publishData(new TextEncoder().encode(payload), {
+                    topic,
+                    reliable: true,
+                });
+            }
+            catch (err) {
+                // Context is additive: an agent that never reads it is not a failed call.
+                logger.warn(`standin: publishing ${topic} failed: ${String(err)}`);
+            }
+        },
+        async aclose() {
+            if (closed)
+                return;
+            closed = true;
+            try {
+                await room.disconnect();
+            }
+            catch {
+                // already closing
+            }
+            if (!config.deleteRoomOnEnd)
+                return;
+            // Deleted, not just left. A dispatched agent job whose room still exists
+            // sits there until LiveKit's own empty-room timeout, which is minutes of
+            // a worker slot doing nothing. Deleting it ends the job at once.
+            try {
+                const client = new server.RoomServiceClient(httpUrl(config.url), config.apiKey, config.apiSecret);
+                await client.deleteRoom(roomName);
+            }
+            catch (err) {
+                logger.warn(`standin: could not delete the room, it will idle out: ${String(err)}`);
+            }
+        },
+    };
+}
+//# sourceMappingURL=room.js.map

package/dist/plugins/livekit/room.js.map

@@ -0,0 +1 @@
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package/dist/plugins/openai/agent.d.ts

@@ -0,0 +1,74 @@
+import type { OpenAIConfig } from "./config.js";
+/** An event from the Realtime API. Only `type` is guaranteed. */
+export type AgentMessage = Record<string, unknown> & {
+    type: string;
+};
+/** Caller details folded into the session instructions. */
+export interface CallerContext {
+    callerName: string;
+    tenantId: string;
+    direction: string;
+}
+/**
+ * The call capabilities every session gets, declared as tools.
+ *
+ * The list is the SDK's, in the Realtime API's shape, so every provider offers
+ * the same agent and a new capability needs no edit here.
+ */
+export declare const BUILT_IN_TOOLS: Array<Record<string, unknown>>;
+/** The names above, for the shadowing check. */
+export declare const BUILT_IN_TOOL_NAMES: ReadonlySet<string>;
+/**
+ * Normalise a remote MCP tool entry for the session's tools array.
+ *
+ * The Realtime API dials the server itself and runs those tools server-side, so
+ * nothing here executes at call time. Approval defaults to never because a live
+ * voice call has no approval interface, and a pending approval is a caller
+ * listening to silence.
+ */
+export declare function mcpTool(entry: Record<string, unknown>): Record<string, unknown>;
+/** Compose the effective instructions: your prompt, the caller, and tool guidance. */
+export declare function buildInstructions(base: string, caller: CallerContext): string;
+/** Per-call session configuration, the Realtime equivalent of a conversation init. */
+export declare function buildSessionUpdate(config: OpenAIConfig, instructions: string, extraTools?: Array<Record<string, unknown>>): Record<string, unknown>;
+/** What the relay needs from a session. Tests substitute their own. */
+export interface AgentPort {
+    readonly isOpen: boolean;
+    sendSessionUpdate(update: Record<string, unknown>): void;
+    /** Caller audio at the WIRE rate. This resamples up to the model's rate. */
+    sendAudio(pcm: Buffer): void;
+    sendContext(text: string): void;
+    sendUserTurn(text: string): void;
+    cancelResponse(): void;
+    sendToolResult(callId: string, output: string): void;
+    aclose(): Promise<void>;
+}
+/** The socket to one OpenAI Realtime session. */
+export declare class AgentSocket implements AgentPort {
+    #private;
+    private constructor();
+    /** Open the session, retrying once on a transient failure. */
+    static connect(config: OpenAIConfig, onMessage: (message: AgentMessage) => void | Promise<void>, 
+    /** Agent audio, already resampled DOWN to the wire rate. */
+    onAudio: (pcm: Buffer) => void | Promise<void>, onClose: (code: number, reason: string) => void | Promise<void>): Promise<AgentSocket>;
+    get isOpen(): boolean;
+    sendSessionUpdate(update: Record<string, unknown>): void;
+    /** Caller audio at the wire rate, resampled up to the model's rate. */
+    sendAudio(pcm: Buffer): void;
+    /**
+     * Background context, with no response requested, so the model sees it on its
+     * next turn rather than interrupting to acknowledge it.
+     */
+    sendContext(text: string): void;
+    /** An interrupting user turn, which is how a goodbye gets spoken. */
+    sendUserTurn(text: string): void;
+    cancelResponse(): void;
+    /**
+     * Answer a tool call, then ask for the turn that speaks the answer. Without
+     * the second message the model has the result and says nothing about it.
+     */
+    sendToolResult(callId: string, output: string): void;
+    /** Close the session. Safe to call twice. */
+    aclose(): Promise<void>;
+}
+//# sourceMappingURL=agent.d.ts.map

package/dist/plugins/openai/agent.d.ts.map

@@ -0,0 +1 @@
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