@signalwire/js 4.0.0-beta.11 → 4.0.0-beta.12

package/dist/index.d.mts

@@ -84,7 +84,7 @@ type WebSocketAdapter = new (url: string | URL, protocols?: string | string[]) =
  * At least one of `token` or `authorizationState` must be provided.
  */
 interface SDKCredential {
-  /** JWT subscriber access token (SAT). */
+  /** JWT user access token (SAT). */
   token?: string;
   /** Pre-authorized session state (alternative to token). */
   authorizationState?: string;
@@ -520,7 +520,7 @@ interface HTTPRequestControllerOptions {
   retryDelayMax?: number;
   requestTimeout?: number;
 }
-declare class HTTPRequestController {
+declare class HTTPRequestController extends Destroyable {
   private baseURL;
   private readonly getCredential;
   private static readonly defaultMaxRetries;
@@ -626,57 +626,57 @@ interface SATClaims {
   expires_at?: number;
 }
 //#endregion
-//#region src/core/types/subscriber.types.d.ts
-/** Raw subscriber profile response from the SignalWire Fabric API. */
-interface GetSubscriberInfoResponse {
-  /** Unique subscriber identifier. */
+//#region src/core/types/user.types.d.ts
+/** Raw user profile response from the SignalWire Fabric API. */
+interface GetUserInfoResponse {
+  /** Unique user identifier. */
   id: string;
-  /** Subscriber's email address. */
+  /** User's email address. */
   email: string;
-  /** Subscriber's first name. */
+  /** User's first name. */
   first_name?: string;
-  /** Subscriber's last name. */
+  /** User's last name. */
   last_name?: string;
-  /** Subscriber's display name. */
+  /** User's display name. */
   display_name?: string;
-  /** Subscriber's job title. */
+  /** User's job title. */
   job_title?: string;
-  /** Subscriber's time zone offset. */
+  /** User's time zone offset. */
   time_zone?: number;
-  /** Subscriber's country. */
+  /** User's country. */
   country?: string;
-  /** Subscriber's region or state. */
+  /** User's region or state. */
   region?: string;
-  /** Subscriber's company name. */
+  /** User's company name. */
   company_name?: string;
   /** Key for push notification delivery. */
   push_notification_key: string;
-  /** Application-level settings for this subscriber. */
+  /** Application-level settings for this user. */
   app_settings?: {
     /** Display name configured at the application level. */
     display_name: string;
-    /** Permission scopes granted to this subscriber. */
+    /** Permission scopes granted to this user. */
     scopes: string[];
   };
-  /** Fabric addresses associated with this subscriber. */
+  /** Fabric addresses associated with this user. */
   fabric_addresses: GetAddressResponse[];
   /** Filtered SAT claims (scope, cnf, expires_at) returned when the token has special capabilities. */
   sat_claims?: SATClaims;
 }
 //#endregion
-//#region src/core/entities/Subscriber.d.ts
-/** Subscriber online presence state. */
-type SubscriberPresence = 'online' | 'offline' | 'busy';
+//#region src/core/entities/User.d.ts
+/** User online presence state. */
+type UserPresence = 'online' | 'offline' | 'busy';
 /**
- * Authenticated subscriber profile.
+ * Authenticated user profile.
  *
  * Fetched automatically when a {@link SignalWire} connects.
  * Contains identity, contact, and organization details.
  */
-declare class Subscriber extends Fetchable<GetSubscriberInfoResponse> {
-  /** Unique subscriber identifier. */
+declare class User extends Fetchable<GetUserInfoResponse> {
+  /** Unique user identifier. */
   id: string;
-  /** Subscriber email address. */
+  /** User email address. */
   email: string;
   /** First name. */
   firstName?: string;
@@ -701,12 +701,12 @@ declare class Subscriber extends Fetchable<GetSubscriberInfoResponse> {
     displayName: string;
     scopes: string[];
   };
-  /** Fabric addresses associated with this subscriber. */
+  /** Fabric addresses associated with this user. */
   addresses: GetAddressResponse[];
   /** Filtered SAT claims when the token has special capabilities (e.g., refresh scope). */
   satClaims?: SATClaims;
   constructor(http: HTTPRequestController);
-  protected populateInstance(data: GetSubscriberInfoResponse): void;
+  protected populateInstance(data: GetUserInfoResponse): void;
 }
 //#endregion
 //#region src/core/types/call.types.d.ts
@@ -1742,11 +1742,31 @@ declare class Participant extends Destroyable implements CallParticipant {
   get videoMuted$(): Observable<boolean | undefined>;
   /** Observable indicating whether the participant is deafened. */
   get deaf$(): Observable<boolean | undefined>;
-  /** Observable of the participant's microphone input volume. */
+  /**
+   * Observable of the participant's **server-side** microphone input volume
+   * as reported by the mix engine. This is gain applied on the bridged audio
+   * leg (FreeSWITCH channel read volume), NOT the local browser mic. For a
+   * local PC mic control, see {@link Call.setLocalMicrophoneGain}.
+   *
+   * @see {@link setAudioInputVolume}
+   */
   get inputVolume$(): Observable<number | undefined>;
-  /** Observable of the participant's speaker output volume. */
+  /**
+   * Observable of the participant's **server-side** speaker output volume as
+   * reported by the mix engine (FreeSWITCH channel write volume). NOT the
+   * local HTML `<audio>` element volume — set that on your own element.
+   *
+   * @see {@link setAudioOutputVolume}
+   */
   get outputVolume$(): Observable<number | undefined>;
-  /** Observable of the microphone input sensitivity level. */
+  /**
+   * Observable of the **conference-only** microphone energy/gate sensitivity
+   * level for this member. Routes through the conferencing mix engine and has
+   * no effect on 1:1 WebRTC calls. Populated from `member.updated` events for
+   * conference members.
+   *
+   * @see {@link setAudioInputSensitivity}
+   */
   get inputSensitivity$(): Observable<number | undefined>;
   /** Observable indicating whether echo cancellation is enabled. */
   get echoCancellation$(): Observable<boolean | undefined>;
@@ -1760,8 +1780,8 @@ declare class Participant extends Destroyable implements CallParticipant {
   get denoise$(): Observable<boolean | undefined>;
   /** Observable of custom metadata for this participant. */
   get meta$(): Observable<Record<string, unknown> | undefined>;
-  /** Observable of the participant's subscriber ID. */
-  get subscriberId$(): Observable<string | undefined>;
+  /** Observable of the participant's user ID. */
+  get userId$(): Observable<string | undefined>;
   /** Observable of the participant's address ID. */
   get addressId$(): Observable<string | undefined>;
   /** Observable of the server node ID for this participant. */
@@ -1790,11 +1810,21 @@ declare class Participant extends Destroyable implements CallParticipant {
   get videoMuted(): boolean;
   /** Whether the participant is deafened (incoming audio muted). */
   get deaf(): boolean;
-  /** Current microphone input volume level, or `undefined` if not set. */
+  /**
+   * Current **server-side** microphone input volume as reported by the mix
+   * engine, or `undefined` if not set. Not the local PC mic — see
+   * {@link Call.setLocalMicrophoneGain} for browser-side control.
+   */
   get inputVolume(): number | undefined;
-  /** Current speaker output volume level, or `undefined` if not set. */
+  /**
+   * Current **server-side** speaker output volume from the mix engine, or
+   * `undefined` if not set. Not the local `<audio>` element volume.
+   */
   get outputVolume(): number | undefined;
-  /** Current microphone input sensitivity level, or `undefined` if not set. */
+  /**
+   * Current **conference-only** microphone sensitivity/gate level, or
+   * `undefined` if not set. Applies only to conference members.
+   */
   get inputSensitivity(): number | undefined;
   /** Whether echo cancellation is enabled. */
   get echoCancellation(): boolean;
@@ -1808,8 +1838,8 @@ declare class Participant extends Destroyable implements CallParticipant {
   get denoise(): boolean;
   /** Custom metadata for this participant, or `undefined` if not set. */
   get meta(): Record<string, unknown> | undefined;
-  /** Subscriber ID of this participant, or `undefined` if not available. */
-  get subscriberId(): string | undefined;
+  /** User ID of this participant, or `undefined` if not available. */
+  get userId(): string | undefined;
   /** Address ID of this participant, or `undefined` if not available. */
   get addressId(): string | undefined;
   /** Server node ID for this participant, or `undefined` if not available. */
@@ -1839,15 +1869,40 @@ declare class Participant extends Destroyable implements CallParticipant {
   /** Toggles noise suppression on the audio input. */
   toggleNoiseSuppression(): Promise<void>;
   toggleLowbitrate(): Promise<void>;
-  /** Sets the microphone input sensitivity level. */
+  /**
+   * Adjusts the **conference-only** microphone energy gate / sensitivity level
+   * for this member. Routes through the conferencing mix engine
+   * (`signalwire.conferencing member.set_input_sensitivity`) and has no effect
+   * on 1:1 WebRTC calls — for those, use browser audio constraints via
+   * {@link Call.setNoiseSuppression} / {@link Call.setAutoGainControl}.
+   *
+   * This is **not** a local PC mic gain control; it only changes how the
+   * server-side mixer decides to open the mic gate on this participant.
+   *
+   * @param value - Sensitivity level as understood by the conference engine
+   *   (integer, larger values are more sensitive).
+   */
   setAudioInputSensitivity(value: number): Promise<void>;
   /**
-   * Sets the microphone input volume level.
+   * Sets the **server-side** microphone volume on this participant's bridged
+   * call leg. Applies a multiplier to the audio flowing through the mix
+   * engine (FreeSWITCH channel read volume) — changes what other participants
+   * hear, not what the local browser captures.
+   *
+   * For local PC mic gain, use {@link Call.setLocalMicrophoneGain} instead.
+   *
    * @param value - Volume level (0-100).
    */
   setAudioInputVolume(value: number): Promise<void>;
   /**
-   * Sets the speaker output volume level.
+   * Sets the **server-side** speaker volume on this participant's bridged call
+   * leg (FreeSWITCH channel write volume) — what this participant hears from
+   * the mix before it reaches their client.
+   *
+   * For local playback volume (the `<audio>` element the consumer attaches
+   * `remoteStream` to), set `audioElement.volume` directly in the consumer's
+   * code.
+   *
    * @param value - Volume level (0-100).
    */
   setAudioOutputVolume(value: number): Promise<void>;
@@ -2002,7 +2057,7 @@ interface CallParticipant {
   readonly lowbitrate$: Observable<boolean | undefined>;
   readonly denoise$: Observable<boolean | undefined>;
   readonly meta$: Observable<Record<string, unknown> | undefined>;
-  readonly subscriberId$: Observable<string | undefined>;
+  readonly userId$: Observable<string | undefined>;
   readonly addressId$: Observable<string | undefined>;
   readonly nodeId$: Observable<string | undefined>;
   readonly isTalking$: Observable<boolean | undefined>;
@@ -2023,7 +2078,7 @@ interface CallParticipant {
   readonly lowbitrate: boolean;
   readonly denoise: boolean;
   readonly meta: Record<string, unknown> | undefined;
-  readonly subscriberId: string | undefined;
+  readonly userId: string | undefined;
   readonly addressId: string | undefined;
   readonly nodeId: string | undefined;
   readonly isTalking: boolean;
@@ -2107,7 +2162,12 @@ interface CallAddress {
 type CallStatus = 'new' | 'trying' | 'ringing' | 'connecting' | 'connected' | 'recovering' | 'disconnecting' | 'disconnected' | 'failed' | 'destroyed';
 /** Configuration options for creating a call. */
 interface CallOptions extends MediaOptions {
-  /** Target server node ID. */
+  /**
+   * Optional. Hint to the cluster about which node should host this call.
+   * Used by reattach to pin to the original node; on fresh dials acts as a
+   * steering preference (the server may ignore for placement reasons).
+   * Leave undefined for normal load-balanced placement.
+   */
   readonly nodeId?: string;
   /** Pre-assigned call ID (used for reattach). */
   readonly callId?: string;
@@ -2168,6 +2228,9 @@ interface Call extends CallState {
   readonly self$: Observable<CallSelfParticipant | null>;
   readonly self: CallSelfParticipant | null;
   readonly to?: string;
+  readonly toName?: string;
+  readonly from?: string;
+  readonly fromName?: string;
   readonly direction: CallDirection;
   readonly layouts$: Observable<string[]>;
   readonly layouts: string[];
@@ -2225,9 +2288,6 @@ interface Call extends CallState {
  */
 interface CallManager extends Call {
   readonly options: CallOptions;
-  readonly fromName?: string;
-  readonly from?: string;
-  readonly toName?: string;
   readonly selfId$: Observable<string | null>;
   readonly selfId: string | null;
   readonly nodeId$: Observable<string | null>;
@@ -2497,11 +2557,19 @@ declare class AttachManager {
   private readonly reconnectCallsTimeout;
   private attachKey;
   private session;
+  private writeQueue;
   constructor(storage: StorageManager, deviceController: DeviceController, reconnectCallsTimeout: number, attachKey: string);
   detachAll(): Promise<void>;
   setSession(session: OutboundCallProvider): void;
   private readAttached;
   private writeAttached;
+  /**
+   * Serialize a read-modify-write operation against the attached-calls
+   * storage. The mutator receives the current state and returns the new
+   * state. Concurrent calls queue behind the in-flight one so writes never
+   * interleave.
+   */
+  private mutate;
   attach(call: AttachableCall): Promise<void>;
   detach(call: AttachableCall): Promise<void>;
   flush(): Promise<void>;
@@ -2527,10 +2595,15 @@ declare class AttachManager {
    */
   buildCallOptions(attachment: Attachment): CallOptions;
   /**
-   * Consume stored attachment data for a pending call (used by session-level
-   * verto.attach handler as a future path when server supports it).
+   * Look up stored attachment data for a call id and return CallOptions
+   * suitable for rehydrating a reattached call. Returns undefined when no
+   * matching entry exists in storage.
+   *
+   * Used by the session-level verto.attach handler when the server pushes
+   * an attach event for a call the client doesn't have an object for yet
+   * (e.g. after a reload).
    */
-  consumePendingAttachment(_callId: string): CallOptions | undefined;
+  consumePendingAttachment(callId: string): Promise<CallOptions | undefined>;
   private detachExpired;
 }
 //#endregion
@@ -2826,14 +2899,14 @@ declare class ClientSessionManager extends Destroyable implements SessionState {
   private _authState$;
   /** Sticky flag — once true, stays true for the session lifetime. */
   private _wasClientBound;
-  private _subscriberInfo$;
+  private _userInfo$;
   private _calls$;
   private _iceServers$;
   constructor(getCredential: () => SDKCredential, transport: TransportManager, storage: StorageManager, authorizationStateKey: string, deviceController: DeviceController, attachManager: AttachManager, webRTCApiProvider: WebRTCApiProvider, dpopManager?: CryptoController | undefined, networkChange$?: Observable<NetworkChangeEvent>);
   get incomingCalls$(): Observable<Call[]>;
   get incomingCalls(): Call[];
-  get subscriberInfo$(): Observable<Address | null>;
-  get subscriberInfo(): Address | null;
+  get userInfo$(): Observable<Address | null>;
+  get userInfo(): Address | null;
   get calls$(): Observable<Call[]>;
   get calls(): Call[];
   get iceServers(): RTCIceServer[] | undefined;
@@ -3221,11 +3294,30 @@ interface SDKLogger {
   debug(...args: unknown[]): void;
   trace(...args: unknown[]): void;
 }
+/** Options for WebSocket traffic logging. */
+interface WsTrafficOptions {
+  type: 'send' | 'recv' | 'http';
+  /** Parsed object or raw string — will be JSON.stringify'd for display if an object. */
+  payload: unknown;
+}
+/**
+ * Options for WebSocket traffic logging using raw strings.
+ * The string is only parsed when logging is enabled, avoiding
+ * unnecessary JSON.parse on every message.
+ */
+interface WsTrafficRawOptions {
+  type: 'send' | 'recv';
+  raw: string;
+}
 /** Debug options that control verbose SDK logging. */
 interface DebugOptions {
   /** Log all WebSocket send/recv traffic to the console. */
   logWsTraffic?: boolean;
 }
+/** Extended logger with SDK-internal helpers (wsTraffic). */
+interface InternalSDKLogger extends SDKLogger {
+  wsTraffic: (options: WsTrafficOptions | WsTrafficRawOptions) => void;
+}
 /** Replace the built-in logger with a custom implementation. Pass `null` to restore defaults. */
 declare const setLogger: (logger: SDKLogger | null) => void;
 /** Configure debug options (e.g., `{ logWsTraffic: true }`). */
@@ -3235,13 +3327,14 @@ declare const setDebugOptions: (options: DebugOptions | null) => void;
  * Has no effect when a custom logger is set via `setLogger()`.
  */
 declare const setLogLevel: (level: LogLevel) => void;
+declare const getLogger: () => InternalSDKLogger;
 //#endregion
 //#region src/clients/SignalWire.d.ts
 /** Options for constructing a {@link SignalWire}. */
 interface SignalWireOptions {
   /** Skip automatic WebSocket connection on construction. */
   skipConnection?: boolean;
-  /** Skip automatic subscriber registration on construction. */
+  /** Skip automatic user registration on construction. */
   skipRegister?: boolean;
   /** Skip monitoring media device changes. */
   skipDeviceMonitoring?: boolean;
@@ -3298,6 +3391,12 @@ interface DialOptions extends MediaOptions {
   stereo?: boolean;
   /** Optional node ID for routing the call */
   nodeId?: string;
+  /**
+   * Custom variables sent with the Verto invite. Merged with
+   * `client.preferences.userVariables` and any query-string variables on the
+   * destination URI; values here take precedence.
+   */
+  userVariables?: Record<string, unknown>;
 }
 /**
  * Main entry point for the SignalWire Browser SDK.
@@ -3314,7 +3413,7 @@ interface DialOptions extends MediaOptions {
 declare class SignalWire extends Destroyable implements DeviceController {
   /** Global SDK preferences (timeouts, ICE config, media defaults). */
   preferences: ClientPreferences;
-  private _subscriber$;
+  private _user$;
   private _directory$;
   private _transport;
   private _clientSession;
@@ -3404,19 +3503,19 @@ declare class SignalWire extends Destroyable implements DeviceController {
    */
   connect(): Promise<void>;
   /**
-   * Observable that emits the {@link Subscriber} profile once fetched,
+   * Observable that emits the {@link User} profile once fetched,
    * or `undefined` before authentication completes.
    *
    * @example
    * ```ts
-   * client.subscriber$.subscribe(sub => {
-   *   if (sub) console.log('Logged in as', sub.email);
+   * client.user$.subscribe(u => {
+   *   if (u) console.log('Logged in as', u.email);
    * });
    * ```
    */
-  get subscriber$(): Observable<Subscriber | undefined>;
-  /** Current subscriber snapshot, or `undefined` if not yet authenticated. */
-  get subscriber(): Subscriber | undefined;
+  get user$(): Observable<User | undefined>;
+  /** Current user snapshot, or `undefined` if not yet authenticated. */
+  get user(): User | undefined;
   /**
    * Observable that emits the {@link Directory} instance once the client is connected,
    * or `undefined` while disconnected. Subscribe to this to safely wait for the directory
@@ -3435,9 +3534,9 @@ declare class SignalWire extends Destroyable implements DeviceController {
    * Prefer {@link directory$} when you need to react to the directory becoming available.
    */
   get directory(): Directory | undefined;
-  /** Observable that emits when the subscriber registration state changes. */
+  /** Observable that emits when the user registration state changes. */
   get isRegistered$(): Observable<boolean>;
-  /** Whether the subscriber is currently registered. */
+  /** Whether the user is currently registered. */
   get isRegistered(): boolean;
   /** Whether the client is currently connected. */
   get isConnected(): boolean;
@@ -3468,9 +3567,14 @@ declare class SignalWire extends Destroyable implements DeviceController {
    * which creates a fresh transport and session.
    */
   disconnect(): Promise<void>;
+  /**
+   * Tear down the current transport / session / attach manager. Safe to call
+   * when nothing has been initialized yet (e.g. first connect()).
+   */
+  private teardownTransportAndSession;
   private waitAuthentication;
   /**
-   * Registers the subscriber as online to receive inbound calls and events.
+   * Registers the user as online to receive inbound calls and events.
    *
    * Waits for authentication to complete before sending the registration.
    * If the initial attempt fails, reauthentication is attempted automatically.
@@ -3479,7 +3583,7 @@ declare class SignalWire extends Destroyable implements DeviceController {
    */
   register(): Promise<void>;
   /**
-   * Unregisters the subscriber, going offline for inbound calls.
+   * Unregisters the user, going offline for inbound calls.
    *
    * The WebSocket connection remains open; use {@link disconnect} to fully close it.
    */
@@ -3568,6 +3672,21 @@ declare class SignalWire extends Destroyable implements DeviceController {
   selectVideoInputDevice(device: MediaDeviceInfo | null): void;
   /** Sets the preferred audio output device. */
   selectAudioOutputDevice(device: MediaDeviceInfo | null): void;
+  /**
+   * Apply the currently selected audio output device to an HTMLMediaElement
+   * (e.g. the `<audio>` or `<video>` element the consumer attached the
+   * remote stream to). Uses `HTMLMediaElement.setSinkId` under the hood.
+   * Returns a `Promise<boolean>`: `true` if the sink was applied,
+   * `false` if the browser doesn't support `setSinkId` or no device is
+   * selected.
+   *
+   * @example
+   * ```ts
+   * audioEl.srcObject = call.remoteStream;
+   * await client.applySelectedAudioOutputDevice(audioEl);
+   * ```
+   */
+  applySelectedAudioOutputDevice(element: HTMLMediaElement): Promise<boolean>;
   /** Starts monitoring for media device changes (connect/disconnect). */
   enableDeviceMonitoring(): void;
   /** Stops monitoring for media device changes. */
@@ -3668,6 +3787,126 @@ declare class StaticCredentialProvider implements CredentialProvider {
   authenticate(): Promise<SDKCredential>;
 }
 //#endregion
+//#region src/dependencies/EmbedTokenCredentialProvider.d.ts
+/** Credential provider that exchanges an embed token for a SAT via the host's token endpoint. */
+declare class EmbedTokenCredentialProvider implements CredentialProvider {
+  private host;
+  private embedToken;
+  constructor(host: string, embedToken: string);
+  private fetchSAT;
+  authenticate(): Promise<{
+    token: string;
+    expiry_at: number;
+  }>;
+  refresh(): Promise<{
+    token: string;
+    expiry_at: number;
+  }>;
+}
+//#endregion
+//#region src/controllers/LocalAudioPipeline.d.ts
+/**
+ * Options for {@link LocalAudioPipeline}.
+ */
+interface LocalAudioPipelineOptions {
+  /** Factory for AudioContext — override for tests. Defaults to `new AudioContext()`. */
+  audioContextFactory?: () => AudioContext;
+  /** Initial gain (0..2, where 1 is unity). Defaults to 1. */
+  initialGain?: number;
+  /** RMS level [0..1] above which speaking$ emits true. Defaults to {@link VAD_THRESHOLD}. */
+  speakingThreshold?: number;
+  /**
+   * Milliseconds of silence below the threshold before speaking$ flips back to
+   * false. Prevents flicker on normal speech gaps. Defaults to {@link VAD_HOLD_MS}.
+   */
+  speakingHoldMs?: number;
+  /** Polling interval for level$. Defaults to {@link AUDIO_LEVEL_POLL_INTERVAL_MS}. */
+  pollIntervalMs?: number;
+}
+/**
+ * Web Audio pipeline for the local microphone stream.
+ *
+ * Wraps the raw mic `MediaStreamTrack` in a graph of:
+ *
+ * ```
+ *   MediaStreamAudioSourceNode  →  GainNode  →  AnalyserNode  →  MediaStreamAudioDestinationNode
+ * ```
+ *
+ * The {@link outputTrack} from the destination node is what callers should
+ * attach to the `RTCRtpSender` in place of the raw mic track. The same
+ * destination track is reused across input changes (device switch, mute /
+ * unmute track replacement) so the sender reference stays stable — only the
+ * source end of the graph is rebuilt.
+ *
+ * The pipeline owns a single {@link AudioContext}. Callers must invoke
+ * {@link destroy} to release it when the call ends.
+ */
+declare class LocalAudioPipeline extends Destroyable {
+  private readonly _audioContext;
+  private readonly _gainNode;
+  private readonly _analyser;
+  private readonly _destination;
+  private readonly _analyserBuffer;
+  private readonly _speakingThreshold;
+  private readonly _speakingHoldMs;
+  private readonly _pollIntervalMs;
+  private _inputSource;
+  private _inputStream;
+  private _lastSpokeAt;
+  private _gain$;
+  /** 1 when audio should pass through, 0 when silenced by PTT. */
+  private _pttMultiplier;
+  constructor(options?: LocalAudioPipelineOptions);
+  /** Observable of the current gain value (0..2). */
+  get gain$(): Observable<number>;
+  /** Current gain value (0..2). */
+  get gain(): number;
+  /**
+   * Processed output track to attach to the RTCRtpSender. Stable reference
+   * across input changes, so `sender.replaceTrack(pipeline.outputTrack)` only
+   * needs to be called once.
+   */
+  get outputTrack(): MediaStreamTrack;
+  /**
+   * Root-mean-square audio level of the input signal, 0..1. Emits on a fixed
+   * interval (~30fps by default).
+   */
+  get level$(): Observable<number>;
+  /**
+   * Boolean VAD derived from {@link level$}. True while level ≥ threshold or
+   * during the hold window after the last frame that crossed the threshold.
+   */
+  get speaking$(): Observable<boolean>;
+  /**
+   * Set gain multiplier applied to the input signal. 0 = silence,
+   * 1 = unity, 2 = 2x. Values are clamped to [0, 2]. The effective gain on
+   * the graph also respects the current PTT state.
+   */
+  setGain(value: number): void;
+  /**
+   * Silence the graph when `active = false`, otherwise restore the configured
+   * gain. Use this from a PTT handler: released → `false`, held → `true`.
+   * Orthogonal to {@link setGain} — once PTT returns to active, the last
+   * configured gain reappears.
+   */
+  setPTTActive(active: boolean): void;
+  private applyEffectiveGain;
+  /**
+   * Wire a new raw mic track as the pipeline's input. Replaces any previous
+   * input source and reconnects the graph so {@link outputTrack} continues
+   * to emit the processed audio. Pass `null` to disconnect the input (the
+   * output track stays alive but emits silence).
+   *
+   * Also resumes the underlying AudioContext on attach — Chrome creates it
+   * in a suspended state and the graph won't process (the destination
+   * track emits silence) until resume() succeeds.
+   */
+  setInputTrack(track: MediaStreamTrack | null): void;
+  destroy(): void;
+  private computeLevel;
+  private evaluateSpeaking;
+}
+//#endregion
 //#region src/controllers/RTCPeerConnectionController.d.ts
 interface RTCPeerConnectionControllerOptions extends MediaOptions {
   callId?: string;
@@ -3728,6 +3967,7 @@ declare class RTCPeerConnectionController extends Destroyable {
   private _remoteDescription$;
   private _remoteStream$;
   private _remoteOfferMediaDirections;
+  private _localAudioPipeline;
   constructor(options?: RTCPeerConnectionControllerOptionsPartial, remoteSessionDescription?: string, deviceController?: DeviceController);
   private get iceGatheringController();
   private get shouldEmitLocalDescription();
@@ -3815,7 +4055,6 @@ declare class RTCPeerConnectionController extends Destroyable {
    */
   private setupEventListeners;
   private negotiationEnded;
-  restarIce(): void;
   /**
    * Trigger an ICE restart through the existing negotiation pipeline.
    *
@@ -3841,6 +4080,17 @@ declare class RTCPeerConnectionController extends Destroyable {
   private getDisplayMedia;
   private setupRemoteTracks;
   restoreTrackSender(kind: 'audio' | 'video' | 'both'): Promise<void>;
+  /**
+   * Return the lazily-created {@link LocalAudioPipeline}, constructing it on
+   * first access. On creation the current audio sender's track is routed
+   * through the pipeline (input → gain → analyser → destination) and the
+   * sender is switched to emit the processed track. Returns `null` when no
+   * audio sender exists yet (pre-negotiation).
+   */
+  ensureLocalAudioPipeline(): LocalAudioPipeline | null;
+  /** The active LocalAudioPipeline, or null if it hasn't been created yet. */
+  get localAudioPipeline(): LocalAudioPipeline | null;
+  private applyLocalAudioPipelineToSender;
   /**
    * Add a local media track to the peer connection.
    * @param track - The MediaStreamTrack to add
@@ -3911,6 +4161,10 @@ interface WebRTCVerto extends VertoManager {
   requestIceRestartAll?: (relayOnly?: boolean) => Promise<void>;
   /** Request keyframes on all video-receiving legs (skips send-only screen share). */
   requestKeyframeAll?: () => void;
+  /** Lazily create (or return) the local audio pipeline for the main peer connection. */
+  ensureLocalAudioPipeline(): LocalAudioPipeline | null;
+  /** Current local audio pipeline, or null if it has not been created yet. */
+  readonly localAudioPipeline: LocalAudioPipeline | null;
 }
 //#endregion
 //#region src/managers/CallEventsManager.d.ts
@@ -4058,6 +4312,8 @@ declare class WebRTCCall extends Destroyable implements CallManager {
   private _bandwidthConstrained$;
   private _mediaParamsUpdated$;
   private _customSubscriptions;
+  private _pushToTalkEnabled;
+  private _remoteAudioMeter;
   constructor(clientSession: ClientSession, options: CallOptions, initialization: CallInitialization, address?: Address | undefined);
   /** Observable stream of errors from media, signaling, and peer connection layers. */
   get errors$(): Observable<CallError>;
@@ -4221,6 +4477,16 @@ declare class WebRTCCall extends Destroyable implements CallManager {
    * Called from within the status subscription to wire stats and recovery.
    */
   private initResilienceSubsystems;
+  /**
+   * Wait for the underlying RTCPeerConnection to reach 'connected' after
+   * triggering an ICE restart. Resolves true on success, false on failure
+   * or if the state doesn't transition within the configured timeout.
+   *
+   * Polls connectionState directly because the recovery manager already
+   * wraps this call in its own withTimeout(); a separate listener-based
+   * implementation would race the outer timeout in subtle ways.
+   */
+  private waitForPeerConnectionConnected;
   /**
    * @internal Stop and destroy resilience subsystems (on disconnect/destroy).
    * Clears references so they can be re-created on reconnect.
@@ -4375,12 +4641,86 @@ declare class WebRTCCall extends Destroyable implements CallManager {
    * @see {@link status$} to observe the transfer progress.
    */
   transfer(options: TransferOptions): Promise<void>;
+  /**
+   * Set the local microphone gain as a percentage applied before transmission.
+   *
+   *   - `0`   = silent
+   *   - `100` = unity (no change, default)
+   *   - `200` = 2× digital boost (max; expect clipping / noise amplification)
+   *
+   * Values are clamped to [0, 200]. Engages the local audio pipeline on
+   * first use (one-time cost).
+   *
+   * Note: this is a **digital** multiplier applied in a Web Audio GainNode
+   * between your mic track and the RTCRtpSender — it does not change the
+   * physical mic's hardware sensitivity. Browsers' autoGainControl can
+   * fight the setting; call {@link setAutoGainControl}(false) for
+   * predictable behaviour.
+   *
+   * @param value - Gain percentage (0..200; 100 = unity).
+   */
+  setLocalMicrophoneGain(value: number): void;
+  /** Observable of the current local microphone gain (0..200, where 100 = unity). */
+  get localMicrophoneGain$(): Observable<number>;
+  /**
+   * Observable of the RMS audio level of the local microphone, 0..1.
+   * Emits at ~30fps while a mic track is active. Engages the local audio
+   * pipeline on first subscription.
+   */
+  get localAudioLevel$(): Observable<number>;
+  /**
+   * Observable that is `true` while the local participant is speaking
+   * (RMS level above the VAD threshold, with hold time to avoid flicker).
+   */
+  get localSpeaking$(): Observable<boolean>;
+  /**
+   * Enable push-to-talk: while {@link setPushToTalkActive} has been called
+   * with `false`, the microphone gain is forced to 0; calling
+   * {@link setPushToTalkActive} with `true` restores the configured gain.
+   * Use this instead of mute/unmute for instant talk/silence transitions
+   * because it doesn't rebuild the track.
+   *
+   * This method installs the pipeline but does not attach any keyboard
+   * listener — consumers bind the key themselves and call
+   * {@link setPushToTalkActive} on keydown/keyup.
+   */
+  enablePushToTalk(): void;
+  /** Disable push-to-talk; mic gain returns to the configured value. */
+  disablePushToTalk(): void;
+  /**
+   * While push-to-talk is enabled, sets the talk state. `true` = transmitting,
+   * `false` = silent. No-op if push-to-talk has not been enabled.
+   */
+  setPushToTalkActive(active: boolean): void;
+  /**
+   * Toggle echo cancellation on the local mic at runtime. Applied via
+   * `track.applyConstraints`; browsers that don't honour runtime constraints
+   * (notably iOS Safari) fall back to re-acquiring the track with the new
+   * constraint set and plumbing the replacement through the local audio
+   * pipeline if one is active.
+   */
+  setEchoCancellation(enabled: boolean): Promise<void>;
+  /** Toggle browser noise suppression on the local mic at runtime. */
+  setNoiseSuppression(enabled: boolean): Promise<void>;
+  /** Toggle browser automatic gain control on the local mic at runtime. */
+  setAutoGainControl(enabled: boolean): Promise<void>;
+  /**
+   * Observable of the aggregate remote audio level, 0..1 RMS. The server
+   * delivers a single mixed audio stream for all remote participants — this
+   * meter reports that mix. Per-participant audio is not available client-side.
+   *
+   * Engages a shared AudioContext on first subscription (cheap — one
+   * AnalyserNode, no GainNode, no destination) so it does not affect the
+   * caller's audio element playback.
+   */
+  get remoteAudioLevel$(): Observable<number>;
   /** Destroys the call, releasing all resources and subscriptions. */
   destroy(): void;
   /**
    * @internal Send a verto.subscribe message to add an event type to the
-   * server's subscription list for this call. Best-effort — failures are
-   * logged but don't prevent the filtered observable from being returned.
+   * server's subscription list for this call. Returns the underlying RPC
+   * promise so callers can decide whether to cache the observable on success
+   * or retry on failure.
    */
   private _sendVertoSubscribe;
 }
@@ -4397,5 +4737,5 @@ declare const version: string;
  */
 declare const ready: boolean;
 //#endregion
-export { Address, type AddressHistory, type AudioConstraintsEvent, type AuthenticateContext, type Call, type CallAddress, type CallCapabilitiesState, CallCreateError, type CallDiagnosticSummary, type CallDirection, type CallError, type CallErrorKind, type NetworkIssue as CallNetworkIssue, type NetworkIssue, type NetworkMetrics as CallNetworkMetrics, type NetworkMetrics, type CallOptions, type CallParticipant, type CallSelfParticipant, type CallState, type CallStatus, type Capability, ClientPreferences, CollectionFetchError, type ConstraintFallbackEvent, type CredentialProvider, DPoPInitError, type DebugOptions, type DeviceController, type DeviceRecoveryEvent, DeviceTokenError, type DiagnosticEvent, type DialOptions, type Directory, type ExecuteMethod, InvalidCredentialsError, type JSONRPCErrorResponse, type JSONRPCRequest, type JSONRPCResponse, type JSONRPCSuccessResponse, type LayoutLayer, type LogLevel, type MediaDirection, type MediaDirections, type MediaOptions, type MediaParamsEvent, MediaTrackError, type MemberCapabilities, MessageParseError, type NodeSocketAdapter, type OnOffCapability, OverconstrainedFallbackError, Participant, type PendingRPCOptions, type PermissionResult, type PlatformCapabilities, PreflightError, type PreflightOptions, type PreflightResult, type QualityLevel, RecoveryError, type RecoveryEvent, type RecoveryState, type ResilienceCallStatus, type SATClaims, type SDKCredential, type SDKLogger, type ScreenShareStatus, type SelectDeviceOptions, SelfCapabilities, SelfParticipant, type SessionDiagnostics, type SessionState, SignalWire, type SignalWireOptions, StaticCredentialProvider, type Storage, type StoredDevicePreference, Subscriber, type SubscriberPresence, type TextMessage, TokenRefreshError, type TransferOptions, UnexpectedError, VertoPongError, type VideoPosition, type WebRTCApiProvider, WebRTCCall, type WebRTCMediaDevices, type WebSocketAdapter, embeddableCall, isSelfParticipant, ready, setDebugOptions, setLogLevel, setLogger, version };
+export { Address, type AddressHistory, type AudioConstraintsEvent, type AuthenticateContext, type Call, type CallAddress, type CallCapabilitiesState, CallCreateError, type CallDiagnosticSummary, type CallDirection, type CallError, type CallErrorKind, type NetworkIssue as CallNetworkIssue, type NetworkIssue, type NetworkMetrics as CallNetworkMetrics, type NetworkMetrics, type CallOptions, type CallParticipant, type CallSelfParticipant, type CallState, type CallStatus, type Capability, ClientPreferences, CollectionFetchError, type ConstraintFallbackEvent, type CredentialProvider, DPoPInitError, type DebugOptions, type DeviceController, type DeviceRecoveryEvent, DeviceTokenError, type DiagnosticEvent, type DialOptions, type Directory, EmbedTokenCredentialProvider, type ExecuteMethod, InvalidCredentialsError, type JSONRPCErrorResponse, type JSONRPCRequest, type JSONRPCResponse, type JSONRPCSuccessResponse, type LayoutLayer, type LogLevel, type MediaDirection, type MediaDirections, type MediaOptions, type MediaParamsEvent, MediaTrackError, type MemberCapabilities, MessageParseError, type NodeSocketAdapter, type OnOffCapability, OverconstrainedFallbackError, Participant, type PendingRPCOptions, type PermissionResult, type PlatformCapabilities, PreflightError, type PreflightOptions, type PreflightResult, type QualityLevel, RecoveryError, type RecoveryEvent, type RecoveryState, type ResilienceCallStatus, type SATClaims, type SDKCredential, type SDKLogger, type ScreenShareStatus, type SelectDeviceOptions, SelfCapabilities, SelfParticipant, type SessionDiagnostics, type SessionState, SignalWire, type SignalWireOptions, StaticCredentialProvider, type Storage, type StoredDevicePreference, type TextMessage, TokenRefreshError, type TransferOptions, UnexpectedError, User, type UserPresence, VertoPongError, type VideoPosition, type WebRTCApiProvider, WebRTCCall, type WebRTCMediaDevices, type WebSocketAdapter, embeddableCall, getLogger, isSelfParticipant, ready, setDebugOptions, setLogLevel, setLogger, version };
 //# sourceMappingURL=index.d.mts.map