@craftedxp/voice-js 0.4.2 → 0.6.0

package/dist/browser.d.ts

@@ -1,3 +1,5 @@
+import { RemoteTrack, LocalVideoTrack } from 'livekit-client';
+
 interface ClientTool {
     description: string;
     parameters: Record<string, unknown>;
@@ -79,6 +81,201 @@ interface BuildWsUrlArgs {
 }
 declare function buildWsUrl(args: BuildWsUrlArgs): string;
 
+type SystemMessage = {
+    kind: 'room.starting';
+    at: string;
+} | {
+    kind: 'room.ending.soon';
+    minutesRemaining: 5 | 1;
+} | {
+    kind: 'room.ended';
+    reason: 'duration_reached' | 'manual' | 'empty';
+} | {
+    kind: 'role.promoted';
+    participantId: string;
+    name: string;
+} | {
+    kind: 'role.demoted';
+    participantId: string;
+    name: string;
+} | {
+    kind: 'participant.removed';
+    participantId: string;
+    name: string;
+    byHost?: string;
+} | {
+    kind: 'notetaker.connected';
+} | {
+    kind: 'notetaker.disconnected';
+} | {
+    kind: 'notetaker.partial_degraded';
+    participantId: string;
+};
+type TranscriptMessage = {
+    kind: 'partial';
+    participantId: string;
+    speakerName: string;
+    text: string;
+    startedAt: string;
+};
+
+interface JoinRoomOptions {
+    /** Full HTTPS URL of the Voissia server. Same shape as VoiceClientConfig.apiBase. */
+    apiBase: string;
+    /** Server-generated room id (`rm_…`). */
+    roomId: string;
+    /** Shared room join token from the invite link. Mints a fresh participant each call. */
+    joinCode: string;
+    /** Display name the joiner registers under for this participant. */
+    name: string;
+}
+interface RoomParticipantInfo {
+    /** Stable participant id (`p_…`); strips any `guest:` LiveKit identity prefix. */
+    participantId: string;
+    /** Display name as the worker registered it; may be empty. */
+    name: string;
+}
+type RoomTrackKind = 'audio' | 'video';
+/** What a track is — lets consumers tell a camera apart from a screen share
+ *  (a participant can publish both at once). Mirrors livekit `Track.Source`. */
+type RoomTrackSource = 'camera' | 'microphone' | 'screen_share' | 'screen_share_audio' | 'unknown';
+interface RoomTrackEvent {
+    /** Stable participant id (`p_…`); `guest:` prefix stripped. */
+    participantId: string;
+    kind: RoomTrackKind;
+    /** Distinguishes camera vs screen_share so each can render as its own tile. */
+    source: RoomTrackSource;
+    /** livekit-client track — call `.attach(el)` / `.detach()` to render. */
+    track: RemoteTrack;
+}
+type RoomEventName = 'participant.joined' | 'participant.left' | 'transcript.partial' | 'transcript.final' | 'system.message' | 'room.ended' | 'track.subscribed' | 'track.unsubscribed' | 'active.speakers';
+interface RoomEventPayloads {
+    'participant.joined': RoomParticipantInfo;
+    'participant.left': RoomParticipantInfo;
+    'transcript.partial': TranscriptMessage;
+    /**
+     * Reserved — the worker currently emits only partials over the transcript
+     * topic. Final-utterance events will land in Phase 8 once the worker
+     * publishes a `final` kind; the SDK keeps the slot reserved so consumers
+     * can register handlers today.
+     */
+    'transcript.final': {
+        participantId: string;
+        speakerName: string;
+        text: string;
+        startedAt: string;
+    };
+    'system.message': SystemMessage;
+    'room.ended': undefined;
+    'track.subscribed': RoomTrackEvent;
+    'track.unsubscribed': RoomTrackEvent;
+    /** participantIds currently speaking (drives an active-speaker UI). */
+    'active.speakers': string[];
+}
+type Handler<E extends RoomEventName> = (payload: RoomEventPayloads[E]) => void;
+interface RoomSession {
+    /** This session's own stable participant id (`p_…`). Useful to filter
+     *  yourself out of `active.speakers`, which includes the local participant. */
+    readonly participantId: string;
+    /** Snapshot of the remote participants currently connected. */
+    readonly participants: RoomParticipantInfo[];
+    /** Subscribe to a typed event. No unsubscribe surface yet (mirrors `Call.onX`). */
+    on<E extends RoomEventName>(event: E, handler: Handler<E>): void;
+    /** Publish the local mic track. Resolves once the track is live on LiveKit. */
+    publishMic(): Promise<void>;
+    /** Publish the local camera track. */
+    publishCamera(): Promise<void>;
+    /** Mid-call mute/unmute of the local mic. */
+    setMicEnabled(on: boolean): Promise<void>;
+    /** Mid-call camera on/off. */
+    setCameraEnabled(on: boolean): Promise<void>;
+    /** Current local mic state (for toggle UI). */
+    isMicEnabled(): boolean;
+    /** Current local camera state (for toggle UI). */
+    isCameraEnabled(): boolean;
+    /** The local camera track for self-view, or null before publishCamera resolves. */
+    getLocalCameraTrack(): LocalVideoTrack | null;
+    /**
+     * Remote tracks already subscribed at this moment. A late joiner misses the
+     * live `track.subscribed` events for tracks published before it connected
+     * (LiveKit delivers them during `connect`, before consumer listeners attach).
+     * Call this right after registering `track.subscribed` to backfill them.
+     */
+    getRemoteTracks(): RoomTrackEvent[];
+    /**
+     * Start/stop sharing the screen (via `getDisplayMedia`). Pass `{ audio: true }`
+     * to also capture shared/system audio where the browser allows it (Chrome:
+     * tab or system audio; macOS Chrome is tab-audio only; Safari/Firefox don't
+     * capture share audio). Publishes a `screen_share` video track (+ optional
+     * `screen_share_audio`); remote peers receive them via `track.subscribed`.
+     */
+    setScreenShareEnabled(on: boolean, opts?: {
+        audio?: boolean;
+    }): Promise<void>;
+    /** Current local screen-share state (for toggle UI). */
+    isScreenShareEnabled(): boolean;
+    /** The local screen-share video track for self-preview, or null when off. */
+    getLocalScreenTrack(): LocalVideoTrack | null;
+    /** Disconnect from LiveKit. Idempotent. Triggers `room.ended` via Disconnected. */
+    leave(): Promise<void>;
+}
+declare const joinRoom: (opts: JoinRoomOptions) => Promise<RoomSession>;
+
+/**
+ * Browser-friendly text-channel chat session. Mint a `ct_` token with
+ * `channel: 'text'` on your backend, then call `startTextSession({...})`
+ * to open the SSE stream.
+ *
+ * Each `.send(text)` is a fresh POST; SSE-per-turn means the connection
+ * closes when each turn ends. Conversation state lives server-side on the
+ * underlying CallRecord.
+ */
+type ChatEvent = {
+    type: 'chat.started';
+    chatId: string;
+    callId: string;
+} | {
+    type: 'token';
+    text: string;
+} | {
+    type: 'tool.call';
+    name: string;
+    args: unknown;
+} | {
+    type: 'tool.result';
+    name: string;
+    ok?: boolean;
+    [key: string]: unknown;
+} | {
+    type: 'turn.end';
+    finishReason: 'stop' | 'aborted' | 'length' | 'tool_error';
+    committedText?: string;
+} | {
+    type: 'error';
+    code: string;
+    message: string;
+};
+interface StartTextSessionOpts {
+    baseUrl: string;
+    token: string;
+    agentId: string;
+    /** Optional inline first user message; otherwise the agent's greeting opens the stream. */
+    text?: string;
+    /** Override the global fetch (useful for tests; defaults to globalThis.fetch). */
+    fetch?: typeof fetch;
+}
+interface TextSession {
+    id: string;
+    callId: string;
+    /** Async iterable for the opening turn — greeting tokens / first reply if text was inlined. */
+    greeting: AsyncIterable<ChatEvent>;
+    /** Send a user message; returns an async iterable for the agent's reply. */
+    send(text: string): Promise<AsyncIterable<ChatEvent>>;
+    /** End the session — DELETE /v1/calls/:callId. */
+    end(): Promise<void>;
+}
+declare function startTextSession(opts: StartTextSessionOpts): Promise<TextSession>;
+
 interface FetchTokenArgs {
     /** The agent the SDK is about to call. */
     agentId: string;
@@ -228,6 +425,25 @@ interface VoiceClientFactory {
      * don't reject this promise.
      */
     startCall: (options: StartCallOptions) => Promise<Call>;
+    /**
+     * Phase 7 (multi-party rooms). Browser only. Exchange a single-use
+     * joinCode for a LiveKit JWT and connect to the room. The returned
+     * `RoomSession` exposes a typed event surface
+     * (participant.joined / participant.left / transcript.partial /
+     *  transcript.final / system.message / room.ended) plus
+     * publishMic / publishCamera / leave. The Node bundle does NOT
+     * implement this — livekit-client is a browser-only WebRTC client.
+     */
+    joinRoom?: (options: Omit<JoinRoomOptions, 'apiBase'>) => Promise<RoomSession>;
+    /**
+     * Open a text-channel chat session (no microphone / audio required).
+     * Mint a `ct_` token with `channel: 'text'` server-side, then call
+     * this to connect. Returns a `TextSession` with:
+     *   - `.greeting` — async iterable for the opening turn
+     *   - `.send(text)` — send a user message; returns an async iterable for the reply
+     *   - `.end()` — close the session (DELETE /v1/calls/:callId)
+     */
+    startTextSession?: (opts: Omit<StartTextSessionOpts, 'baseUrl' | 'fetch'>) => Promise<TextSession>;
 }
 
 type OnChunk = (pcm: ArrayBuffer) => void;
@@ -307,6 +523,31 @@ declare const createReconnectingWebSocket: (options: RWSOptions, onEvent: (ev: R
 };
 type ReconnectingWebSocket = ReturnType<typeof createReconnectingWebSocket>;
 
+/**
+ * Canonical payload a tenant places in their VoIP/FCM push so an
+ * agent-initiated call can connect. It is the `callTokens.mint` result
+ * (token + transport) plus two optional display fields for the native
+ * incoming-call UI. The host receives the push, runs `parseIncomingCall`,
+ * and on accept passes `token` (+ transport / webrtcGatewayBase) into the
+ * voice client. Web background-wake is best-effort (Web Push); native is
+ * the real target. See docs/sdks.md "Agent-initiated calls".
+ */
+interface IncomingCallPayload {
+    token: string;
+    agentId: string;
+    transport: 'ws' | 'webrtc';
+    webrtcGatewayBase?: string;
+    expiresAt?: number;
+    agentName?: string;
+    agentAvatarUrl?: string;
+}
+/**
+ * Validate + normalise a raw push payload into an IncomingCallPayload.
+ * Throws synchronously on malformed input. Unknown transports fall back to
+ * 'ws'; webrtcGatewayBase is ignored unless transport === 'webrtc'.
+ */
+declare const parseIncomingCall: (raw: unknown) => IncomingCallPayload;
+
 /**
  * One-time SDK setup. Returns a factory you call `startCall` on for
  * every voice call.
@@ -334,4 +575,4 @@ type ReconnectingWebSocket = ReturnType<typeof createReconnectingWebSocket>;
  */
 declare function configureVoiceClient(config: VoiceClientConfig): VoiceClientFactory;
 
-export { type Call, type CallEndEvent, type CallEndReason, type CallError, type CallErrorCode, type CallState, type CaptureController, type CaptureOptions, type ClientTool, type ClientToolMap, type FetchToken, type FetchTokenArgs, type FetchTokenResult, type OnAgentSpeakingChange, type OnChunk, type OnError, type OnVolume$1 as OnVolume, type PlaybackController, type PlaybackOptions, type ProtocolCallbacks, type ProtocolState, type RWSEvent, type RWSOptions, type ReconnectingWebSocket, type ServerMessage, type StartCallOptions, type TranscriptEntry, type VoiceClientConfig, type VoiceClientFactory, type VolumeEvent, type WebSocketFactory, type WebSocketLike, buildWsUrl, configureVoiceClient, createAudioCapture, createAudioPlayback, createProtocolState, createReconnectingWebSocket, handleServerMessage };
+export { type Call, type CallEndEvent, type CallEndReason, type CallError, type CallErrorCode, type CallState, type CaptureController, type CaptureOptions, type ChatEvent, type ClientTool, type ClientToolMap, type FetchToken, type FetchTokenArgs, type FetchTokenResult, type IncomingCallPayload, type JoinRoomOptions, type OnAgentSpeakingChange, type OnChunk, type OnError, type OnVolume$1 as OnVolume, type PlaybackController, type PlaybackOptions, type ProtocolCallbacks, type ProtocolState, type RWSEvent, type RWSOptions, type ReconnectingWebSocket, type RoomEventName, type RoomEventPayloads, type RoomParticipantInfo, type RoomSession, type ServerMessage, type StartCallOptions, type StartTextSessionOpts, type SystemMessage, type TextSession, type TranscriptEntry, type TranscriptMessage, type VoiceClientConfig, type VoiceClientFactory, type VolumeEvent, type WebSocketFactory, type WebSocketLike, buildWsUrl, configureVoiceClient, createAudioCapture, createAudioPlayback, createProtocolState, createReconnectingWebSocket, handleServerMessage, joinRoom, parseIncomingCall, startTextSession };

package/dist/browser.js

@@ -26,7 +26,10 @@ __export(browser_exports, {
   createAudioPlayback: () => createAudioPlayback,
   createProtocolState: () => createProtocolState,
   createReconnectingWebSocket: () => createReconnectingWebSocket,
-  handleServerMessage: () => handleServerMessage
+  handleServerMessage: () => handleServerMessage,
+  joinRoom: () => joinRoom,
+  parseIncomingCall: () => parseIncomingCall,
+  startTextSession: () => startTextSession
 });
 module.exports = __toCommonJS(browser_exports);
 
@@ -1047,6 +1050,284 @@ async function createWebRtcCall(opts) {
   };
 }
 
+// src/room.ts
+var import_livekit_client = require("livekit-client");
+
+// src/roomProtocol.ts
+var SYSTEM_TOPIC = "system";
+var TRANSCRIPT_TOPIC = "transcript";
+var decodeSystem = (bytes) => {
+  try {
+    const v = JSON.parse(new TextDecoder().decode(bytes));
+    if (v && typeof v.kind === "string") return v;
+    return null;
+  } catch {
+    return null;
+  }
+};
+var decodeTranscript = (bytes) => {
+  try {
+    const v = JSON.parse(new TextDecoder().decode(bytes));
+    if (v && v.kind === "partial") return v;
+    return null;
+  } catch {
+    return null;
+  }
+};
+
+// src/room.ts
+var identityToPid = (identity) => identity.startsWith("guest:") ? identity.slice("guest:".length) : identity;
+var joinRoom = async (opts) => {
+  const exchangeUrl = `${opts.apiBase.replace(/\/+$/, "")}/v1/rooms/${encodeURIComponent(
+    opts.roomId
+  )}/join`;
+  const exchangeRes = await fetch(exchangeUrl, {
+    method: "POST",
+    headers: { "Content-Type": "application/json" },
+    body: JSON.stringify({ code: opts.joinCode, name: opts.name })
+  });
+  if (!exchangeRes.ok) {
+    const err = await exchangeRes.json().catch(() => ({}));
+    throw new Error(err.error?.code ?? `join_failed_${exchangeRes.status}`);
+  }
+  const exchange = await exchangeRes.json();
+  const handlers = /* @__PURE__ */ new Map();
+  const emit = (e, payload) => {
+    handlers.get(e)?.forEach((h) => {
+      try {
+        h(payload);
+      } catch {
+      }
+    });
+  };
+  const room = new import_livekit_client.Room({ adaptiveStream: true, dynacast: true });
+  room.on(
+    import_livekit_client.RoomEvent.ParticipantConnected,
+    (p) => emit("participant.joined", {
+      participantId: identityToPid(p.identity),
+      name: p.name ?? ""
+    })
+  );
+  room.on(
+    import_livekit_client.RoomEvent.ParticipantDisconnected,
+    (p) => emit("participant.left", {
+      participantId: identityToPid(p.identity),
+      name: p.name ?? ""
+    })
+  );
+  room.on(import_livekit_client.RoomEvent.Disconnected, () => emit("room.ended", void 0));
+  room.on(import_livekit_client.RoomEvent.DataReceived, (data, _participant, _kind, topic) => {
+    if (topic === SYSTEM_TOPIC) {
+      const m = decodeSystem(data);
+      if (m) emit("system.message", m);
+    } else if (topic === TRANSCRIPT_TOPIC) {
+      const m = decodeTranscript(data);
+      if (m) emit("transcript.partial", m);
+    }
+  });
+  const trackKind = (t) => t.kind === import_livekit_client.Track.Kind.Video ? "video" : "audio";
+  const trackSource = (s) => {
+    switch (s) {
+      case import_livekit_client.Track.Source.Camera:
+        return "camera";
+      case import_livekit_client.Track.Source.Microphone:
+        return "microphone";
+      case import_livekit_client.Track.Source.ScreenShare:
+        return "screen_share";
+      case import_livekit_client.Track.Source.ScreenShareAudio:
+        return "screen_share_audio";
+      default:
+        return "unknown";
+    }
+  };
+  room.on(
+    import_livekit_client.RoomEvent.TrackSubscribed,
+    (track, pub, participant) => emit("track.subscribed", {
+      participantId: identityToPid(participant.identity),
+      kind: trackKind(track),
+      source: trackSource(pub.source),
+      track
+    })
+  );
+  room.on(
+    import_livekit_client.RoomEvent.TrackUnsubscribed,
+    (track, pub, participant) => emit("track.unsubscribed", {
+      participantId: identityToPid(participant.identity),
+      kind: trackKind(track),
+      source: trackSource(pub.source),
+      track
+    })
+  );
+  room.on(
+    import_livekit_client.RoomEvent.ActiveSpeakersChanged,
+    (speakers) => emit(
+      "active.speakers",
+      speakers.map((p) => identityToPid(p.identity))
+    )
+  );
+  await room.connect(exchange.livekit.url, exchange.livekit.token);
+  return {
+    participantId: exchange.participantId,
+    get participants() {
+      return [...room.remoteParticipants.values()].map((p) => ({
+        participantId: identityToPid(p.identity),
+        name: p.name ?? ""
+      }));
+    },
+    on(event, handler) {
+      const set = handlers.get(event) ?? /* @__PURE__ */ new Set();
+      set.add(handler);
+      handlers.set(event, set);
+    },
+    publishMic: async () => {
+      await room.localParticipant.setMicrophoneEnabled(true);
+    },
+    publishCamera: async () => {
+      await room.localParticipant.setCameraEnabled(true);
+    },
+    setMicEnabled: async (on) => {
+      await room.localParticipant.setMicrophoneEnabled(on);
+    },
+    setCameraEnabled: async (on) => {
+      await room.localParticipant.setCameraEnabled(on);
+    },
+    isMicEnabled: () => room.localParticipant.isMicrophoneEnabled,
+    isCameraEnabled: () => room.localParticipant.isCameraEnabled,
+    getLocalCameraTrack: () => room.localParticipant.getTrackPublication(import_livekit_client.Track.Source.Camera)?.videoTrack ?? null,
+    getRemoteTracks: () => {
+      const out = [];
+      for (const p of room.remoteParticipants.values()) {
+        for (const pub of p.trackPublications.values()) {
+          const track = pub.track;
+          if (!track) continue;
+          out.push({
+            participantId: identityToPid(p.identity),
+            kind: trackKind(track),
+            source: trackSource(pub.source),
+            track
+          });
+        }
+      }
+      return out;
+    },
+    setScreenShareEnabled: async (on, opts2) => {
+      await room.localParticipant.setScreenShareEnabled(on, { audio: opts2?.audio ?? false });
+    },
+    isScreenShareEnabled: () => room.localParticipant.isScreenShareEnabled,
+    getLocalScreenTrack: () => room.localParticipant.getTrackPublication(import_livekit_client.Track.Source.ScreenShare)?.videoTrack ?? null,
+    leave: async () => {
+      await room.disconnect();
+    }
+  };
+};
+
+// src/textSession.ts
+async function startTextSession(opts) {
+  const f = opts.fetch ?? fetch;
+  const tokenQs = `?token=${encodeURIComponent(opts.token)}`;
+  const startUrl = `${opts.baseUrl}/v1/agents/${opts.agentId}/chat${tokenQs}`;
+  const startBody = opts.text ? JSON.stringify({ text: opts.text }) : "{}";
+  const res = await f(startUrl, {
+    method: "POST",
+    headers: { "Content-Type": "application/json", Accept: "text/event-stream" },
+    body: startBody
+  });
+  if (!res.ok || !res.body) {
+    const text = await res.text().catch(() => "");
+    throw new Error(`startTextSession failed: ${res.status} ${text}`);
+  }
+  const iter = parseSse(res.body);
+  let chatId = "";
+  let callId = "";
+  const buffered = [];
+  const it = iter[Symbol.asyncIterator]();
+  while (true) {
+    const { value, done } = await it.next();
+    if (done) break;
+    if (value.type === "chat.started") {
+      chatId = value.chatId;
+      callId = value.callId;
+      break;
+    }
+    buffered.push(value);
+  }
+  return {
+    id: chatId,
+    callId,
+    greeting: replayThen(buffered, { [Symbol.asyncIterator]: () => it }),
+    async send(text) {
+      const r = await f(`${opts.baseUrl}/v1/chats/${chatId}/messages${tokenQs}`, {
+        method: "POST",
+        headers: { "Content-Type": "application/json", Accept: "text/event-stream" },
+        body: JSON.stringify({ text })
+      });
+      if (!r.ok || !r.body) {
+        const errText = await r.text().catch(() => "");
+        throw new Error(`send failed: ${r.status} ${errText}`);
+      }
+      return parseSse(r.body);
+    },
+    async end() {
+      await f(`${opts.baseUrl}/v1/calls/${callId}`, { method: "DELETE" });
+    }
+  };
+}
+async function* parseSse(body) {
+  const reader = body.getReader();
+  const decoder = new TextDecoder();
+  let buf = "";
+  while (true) {
+    const { value, done } = await reader.read();
+    if (done) return;
+    buf += decoder.decode(value, { stream: true });
+    let idx;
+    while ((idx = buf.indexOf("\n\n")) >= 0) {
+      const chunk = buf.slice(0, idx);
+      buf = buf.slice(idx + 2);
+      let event = "message";
+      let data = "";
+      for (const line of chunk.split("\n")) {
+        if (line.startsWith(":")) continue;
+        if (line.startsWith("event:")) event = line.slice(6).trim();
+        else if (line.startsWith("data:")) data += line.slice(5).trim();
+      }
+      if (!data) continue;
+      try {
+        const parsed = JSON.parse(data);
+        yield { type: event, ...parsed };
+      } catch {
+      }
+    }
+  }
+}
+async function* replayThen(buffered, rest) {
+  for (const x of buffered) yield x;
+  for await (const x of rest) yield x;
+}
+
+// src/incomingCall.ts
+var parseIncomingCall = (raw) => {
+  if (typeof raw !== "object" || raw === null) {
+    throw new Error("parseIncomingCall: payload must be an object");
+  }
+  const p = raw;
+  if (typeof p.token !== "string" || !p.token.startsWith("ct_")) {
+    throw new Error("parseIncomingCall: missing or invalid `token` (expected a ct_ string)");
+  }
+  if (typeof p.agentId !== "string" || p.agentId.length === 0) {
+    throw new Error("parseIncomingCall: missing `agentId`");
+  }
+  const transport = p.transport === "webrtc" ? "webrtc" : "ws";
+  const out = { token: p.token, agentId: p.agentId, transport };
+  if (transport === "webrtc" && typeof p.webrtcGatewayBase === "string") {
+    out.webrtcGatewayBase = p.webrtcGatewayBase;
+  }
+  if (typeof p.expiresAt === "number") out.expiresAt = p.expiresAt;
+  if (typeof p.agentName === "string") out.agentName = p.agentName;
+  if (typeof p.agentAvatarUrl === "string") out.agentAvatarUrl = p.agentAvatarUrl;
+  return out;
+};
+
 // src/browser.ts
 var browserWsFactory = (url) => new globalThis.WebSocket(url);
 var BrowserVoiceFactory = class {
@@ -1104,6 +1385,21 @@ var BrowserVoiceFactory = class {
       await client.start();
       return client;
     };
+    // Multi-party rooms (Phase 7 video).
+    //
+    // The guest's browser calls this with the roomId + joinCode it parsed
+    // out of the invite link. The SDK exchanges the code for a LiveKit
+    // JWT against `${apiBase}/v1/rooms/:roomId/join` (an AUTH-EXEMPT
+    // endpoint — the opaque code is the only credential), then connects
+    // to LiveKit and returns a typed event surface.
+    this.joinRoom = (opts) => joinRoom({ apiBase: this.config.apiBase, ...opts });
+    // Text-channel chat session (no microphone / audio).
+    // Mint a `ct_` token with `channel: 'text'` on your backend, then call
+    // this to open an SSE stream against the chat API.
+    this.startTextSession = (opts) => startTextSession({
+      ...opts,
+      baseUrl: this.config.apiBase
+    });
     this.config = config;
   }
 };
@@ -1118,6 +1414,9 @@ function configureVoiceClient(config) {
   createAudioPlayback,
   createProtocolState,
   createReconnectingWebSocket,
-  handleServerMessage
+  handleServerMessage,
+  joinRoom,
+  parseIncomingCall,
+  startTextSession
 });
 //# sourceMappingURL=browser.js.map