@absolutejs/voice 0.0.22-beta.67 → 0.0.22-beta.670

package/dist/core/types.d.ts

@@ -0,0 +1,1786 @@
+import type { MediaWebRTCStatsCollector, MediaWebRTCStatsReport, MediaWebRTCStatsReportInput, MediaWebRTCStreamContinuityInput, MediaWebRTCStreamContinuityReport } from "@absolutejs/media";
+import type { VoiceOpsDispositionTaskPolicies, VoiceOpsTaskAssignmentRule, VoiceOpsTaskAssignmentRules, VoiceIntegrationWebhookConfig, StoredVoiceIntegrationEvent, StoredVoiceOpsTask, VoiceIntegrationEventStore, VoiceOpsTaskPolicy, VoiceOpsTask, VoiceOpsTaskStore } from "./ops";
+import type { VoiceIntegrationSink } from "./opsSinks";
+import type { StoredVoiceCallReviewArtifact, VoiceCallReviewArtifact, VoiceCallReviewStore } from "../testing/review";
+import type { VoiceTraceEventStore } from "./trace";
+import type { VoiceLiveOpsControlState } from "./liveOps";
+import type { VoiceAuditActor, VoiceAuditEventStore } from "./audit";
+import type { VoiceProfileSwitchGuardDecision, VoiceProfileSwitchGuardMode, VoiceProfileSwitchObservedSignals } from "./profileSwitchRecommendation";
+import type { VoiceRealCallProfileDefaultsReport, VoiceRealCallProfileHistoryReport } from "./proofTrends";
+export type AudioFormat = {
+    container: "raw";
+    encoding: "alaw" | "mulaw" | "pcm_s16le";
+    sampleRateHz: number;
+    channels: 1 | 2;
+};
+export type AudioChunk = ArrayBuffer | ArrayBufferView;
+/**
+ * Structural contract for an inbound noise suppressor applied before STT.
+ * Matches `@absolutejs/media`'s `NoiseSuppressor`, so any media suppressor
+ * (`createFFmpegNoiseSuppressor`, `createEnergyGateNoiseSuppressor`, a Krisp
+ * adapter, etc.) satisfies it without a hard import.
+ */
+export type VoiceNoiseSuppressor = {
+    readonly kind: string;
+    process: (input: {
+        format: AudioFormat;
+        pcm: ArrayBuffer | ArrayBufferView;
+    }) => Promise<{
+        bytes: Uint8Array;
+        format: AudioFormat;
+    }> | {
+        bytes: Uint8Array;
+        format: AudioFormat;
+    };
+    close?: () => Promise<void> | void;
+};
+export type VoiceLanguageStrategy = {
+    mode: "auto-detect";
+    allowedLanguages?: string[];
+} | {
+    mode: "fixed";
+    primaryLanguage: string;
+    secondaryLanguages?: string[];
+} | {
+    mode: "allow-switching";
+    primaryLanguage?: string;
+    secondaryLanguages: string[];
+};
+export type VoiceLanguageStrategyResolver<TContext = unknown> = (input: {
+    context: TContext;
+    scenarioId?: string;
+    sessionId: string;
+}) => Promise<VoiceLanguageStrategy | void> | VoiceLanguageStrategy | void;
+export type VoicePhraseHint = {
+    text: string;
+    aliases?: string[];
+    boost?: number;
+    metadata?: Record<string, unknown>;
+};
+export type VoiceCorrectionRiskTier = "safe" | "balanced" | "risky";
+export type VoiceDomainTerm = {
+    text: string;
+    aliases?: string[];
+    boost?: number;
+    language?: string;
+    metadata?: Record<string, unknown>;
+    pronunciation?: string;
+};
+export type VoiceLexiconEntry = {
+    text: string;
+    aliases?: string[];
+    language?: string;
+    metadata?: Record<string, unknown>;
+    pronunciation?: string;
+};
+export type VoiceTranscriptSentiment = {
+    label: "negative" | "neutral" | "positive" | (string & {});
+    metadata?: Record<string, unknown>;
+    score?: number;
+};
+/** A provider-normalized word hypothesis. Keeping this evidence attached to the
+ * transcript lets applications detect a single risky name or number that would
+ * otherwise disappear inside a high turn-level average confidence. */
+export type TranscriptWord = {
+    confidence?: number;
+    endedAtMs?: number;
+    language?: string;
+    punctuatedText?: string;
+    speaker?: string | number;
+    startedAtMs?: number;
+    text: string;
+};
+/** Provider token evidence. Tokens are intentionally kept separate from words:
+ * subword log probabilities are useful for calibration and routing, but are not
+ * word-level timestamps or confidence scores. */
+export type TranscriptToken = {
+    bytes?: number[];
+    confidence?: number;
+    logProbability?: number;
+    text: string;
+};
+export type Transcript = {
+    id: string;
+    text: string;
+    isFinal: boolean;
+    confidence?: number;
+    language?: string;
+    sentiment?: VoiceTranscriptSentiment;
+    speaker?: string | number;
+    startedAtMs?: number;
+    endedAtMs?: number;
+    vendor?: string;
+    tokens?: TranscriptToken[];
+    words?: TranscriptWord[];
+};
+export type VoiceSTTSessionConfiguration = {
+    languageHints?: string[];
+    lexicon?: VoiceLexiconEntry[];
+    phraseHints?: VoicePhraseHint[];
+    turnDetection?: Partial<VoiceTurnDetectionConfig>;
+};
+export type VoiceTranscriptQuality = {
+    averageConfidence?: number;
+    confidenceSampleCount: number;
+    correction?: VoiceTurnCorrectionDiagnostics;
+    cost?: VoiceTurnCostEstimate;
+    fallbackUsed: boolean;
+    finalTranscriptCount: number;
+    fallback?: VoiceFallbackDiagnostics;
+    partialTranscriptCount: number;
+    selectedTranscriptCount: number;
+    source: "fallback" | "primary";
+    lowestWordConfidence?: number;
+    wordConfidenceSampleCount?: number;
+};
+export type VoiceTurnCorrectionDiagnostics = {
+    attempted: boolean;
+    changed: boolean;
+    correctedText: string;
+    metadata?: Record<string, unknown>;
+    originalText: string;
+    provider?: string;
+    reason?: string;
+};
+export type VoiceTurnCostEstimate = {
+    estimatedRelativeCostUnits: number;
+    fallbackAttemptCount: number;
+    fallbackReplayAudioMs: number;
+    primaryAudioMs: number;
+    totalBillableAudioMs: number;
+};
+export type VoiceFallbackSelectionReason = "fallback-empty" | "primary-empty" | "word-count-margin" | "confidence-margin" | "word-count-tiebreak" | "policy-preference" | "kept-primary";
+export type VoiceFallbackDiagnostics = {
+    attempted: boolean;
+    fallbackConfidence?: number;
+    fallbackText?: string;
+    fallbackWordCount?: number;
+    primaryConfidence: number;
+    primaryText: string;
+    primaryWordCount: number;
+    selected: boolean;
+    selectionReason: VoiceFallbackSelectionReason;
+    trigger: "empty-turn" | "low-confidence" | "empty-or-low-confidence" | "always";
+    triggerReason?: VoiceFallbackTriggerReason;
+};
+export type VoicePartialEvent = {
+    type: "partial";
+    transcript: Transcript;
+    receivedAt: number;
+};
+export type VoiceFinalEvent = {
+    type: "final";
+    transcript: Transcript;
+    receivedAt: number;
+};
+export type VoiceEndOfTurnEvent = {
+    type: "endOfTurn";
+    reason: "vendor" | "silence" | "manual";
+    receivedAt: number;
+};
+export type VoiceErrorEvent = {
+    type: "error";
+    error: Error;
+    recoverable: boolean;
+    code?: string;
+};
+export type VoiceCloseEvent = {
+    type: "close";
+    code?: number;
+    reason?: string;
+    recoverable?: boolean;
+};
+export type STTSessionEventMap = {
+    partial: VoicePartialEvent;
+    final: VoiceFinalEvent;
+    endOfTurn: VoiceEndOfTurnEvent;
+    error: VoiceErrorEvent;
+    close: VoiceCloseEvent;
+};
+export type STTAdapterSession = {
+    on: <K extends keyof STTSessionEventMap>(event: K, handler: (payload: STTSessionEventMap[K]) => void | Promise<void>) => () => void;
+    send: (audio: AudioChunk) => Promise<void>;
+    /** Update provider-supported STT context without restarting the stream. */
+    configure?: (configuration: VoiceSTTSessionConfiguration) => Promise<void>;
+    close: (reason?: string) => Promise<void>;
+};
+export type STTAdapterOpenOptions = {
+    sessionId: string;
+    format: AudioFormat;
+    languageStrategy?: VoiceLanguageStrategy;
+    lexicon?: VoiceLexiconEntry[];
+    phraseHints?: VoicePhraseHint[];
+    signal?: AbortSignal;
+};
+export type STTAdapter<TOptions extends STTAdapterOpenOptions = STTAdapterOpenOptions> = {
+    kind: "stt";
+    open: (options: TOptions) => Promise<STTAdapterSession> | STTAdapterSession;
+};
+export type TTSAudioEvent = {
+    type: "audio";
+    chunk: AudioChunk;
+    format: AudioFormat;
+    receivedAt: number;
+};
+export type TTSSessionEventMap = {
+    audio: TTSAudioEvent;
+    error: VoiceErrorEvent;
+    close: VoiceCloseEvent;
+};
+export type TTSAdapterSession = {
+    on: <K extends keyof TTSSessionEventMap>(event: K, handler: (payload: TTSSessionEventMap[K]) => void | Promise<void>) => () => void;
+    send: (text: string) => Promise<void>;
+    cancel?: (reason?: string) => Promise<void>;
+    close: (reason?: string) => Promise<void>;
+};
+export declare const ttsAdapterSessionCanCancel: (session: TTSAdapterSession) => session is TTSAdapterSession & {
+    cancel: (reason?: string) => Promise<void>;
+};
+export type VoiceTTSProsody = {
+    emphasis?: number;
+    pitch?: number;
+    speed?: number;
+    style?: string;
+};
+export type TTSAdapterOpenOptions = {
+    sessionId: string;
+    lexicon?: VoiceLexiconEntry[];
+    prosody?: VoiceTTSProsody;
+    signal?: AbortSignal;
+};
+export type TTSAdapter<TOptions extends TTSAdapterOpenOptions = TTSAdapterOpenOptions> = {
+    kind: "tts";
+    open: (options: TOptions) => Promise<TTSAdapterSession> | TTSAdapterSession;
+};
+export type RealtimeSessionEventMap = STTSessionEventMap & {
+    audio: TTSAudioEvent;
+};
+export type RealtimeAdapterSession = {
+    on: <K extends keyof RealtimeSessionEventMap>(event: K, handler: (payload: RealtimeSessionEventMap[K]) => void | Promise<void>) => () => void;
+    send: (input: AudioChunk | string) => Promise<void>;
+    /** Update provider-supported input transcription context in place. */
+    configure?: (configuration: VoiceSTTSessionConfiguration) => Promise<void>;
+    close: (reason?: string) => Promise<void>;
+};
+export type RealtimeAdapterOpenOptions = {
+    sessionId: string;
+    format: AudioFormat;
+    languageStrategy?: VoiceLanguageStrategy;
+    lexicon?: VoiceLexiconEntry[];
+    modalities?: ReadonlyArray<"audio" | "text">;
+    phraseHints?: VoicePhraseHint[];
+    promptCacheKey?: string;
+    semanticVAD?: import("./assistantMode").VoiceSemanticVADConfig;
+    signal?: AbortSignal;
+};
+export type RealtimeAdapter<TOptions extends RealtimeAdapterOpenOptions = RealtimeAdapterOpenOptions> = {
+    kind: "realtime";
+    open: (options: TOptions) => Promise<RealtimeAdapterSession> | RealtimeAdapterSession;
+};
+export type VoiceSessionStatus = "active" | "reconnecting" | "completed" | "failed";
+export type VoiceReconnectClientStatus = "idle" | "reconnecting" | "resumed" | "exhausted";
+export type VoiceReconnectClientState = {
+    attempts: number;
+    lastDisconnectAt?: number;
+    lastResumedAt?: number;
+    maxAttempts: number;
+    nextAttemptAt?: number;
+    status: VoiceReconnectClientStatus;
+};
+export type VoiceTurnCitation = {
+    chunkId: string;
+    score: number;
+    source?: string;
+    title?: string;
+};
+export type VoiceTurnRecord<TResult = unknown> = {
+    id: string;
+    text: string;
+    quality?: VoiceTranscriptQuality;
+    transcripts: Transcript[];
+    assistantText?: string;
+    attachments?: import("./agent").VoiceAgentMessageAttachment[];
+    citations?: VoiceTurnCitation[];
+    committedAt: number;
+    result?: TResult;
+};
+export type VoiceCostTelemetryConfig<TContext = unknown, TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown> = {
+    fallbackPassCostUnit?: number;
+    onTurnCost?: (input: {
+        api: VoiceSessionHandle<TContext, TSession, TResult>;
+        context: TContext;
+        estimate: VoiceTurnCostEstimate;
+        session: TSession;
+        turn: VoiceTurnRecord<TResult>;
+    }) => Promise<void> | void;
+    primaryPassCostUnit?: number;
+};
+export type VoiceSessionRecord<TMeta = Record<string, never>, TResult = unknown> = {
+    id: string;
+    createdAt: number;
+    /** Persisted before the initial assistant greeting is emitted so reconnects
+     *  cannot replay the full introduction when the caller has not answered yet. */
+    greetingDeliveredAt?: number;
+    lastActivityAt?: number;
+    status: VoiceSessionStatus;
+    transcripts: Transcript[];
+    currentTurn: {
+        transcripts: Transcript[];
+        partialText: string;
+        partialStartedAt?: number;
+        partialEndedAt?: number;
+        finalText: string;
+        lastAudioAt?: number;
+        lastSpeechAt?: number;
+        lastTranscriptAt?: number;
+        silenceStartedAt?: number;
+    };
+    turns: VoiceTurnRecord<TResult>[];
+    committedTurnIds: string[];
+    reconnect: {
+        attempts: number;
+        lastDisconnectAt?: number;
+    };
+    /** Durable caller pause state so a reconnect or process replacement restores
+     * the remaining pause window instead of rearming conversation watchdogs. */
+    pause?: {
+        expiresAt: number;
+        pausedAt: number;
+    };
+    lastCommittedTurn?: {
+        signature: string;
+        text: string;
+        transcriptIds: string[];
+        committedAt: number;
+    };
+    call?: VoiceCallLifecycleState;
+    metadata?: TMeta;
+    scenarioId?: string;
+};
+export type VoiceSessionSummary = {
+    id: string;
+    createdAt: number;
+    lastActivityAt?: number;
+    status: VoiceSessionStatus;
+    turnCount: number;
+};
+export type VoiceCallDisposition = "completed" | "transferred" | "escalated" | "voicemail" | "no-answer" | "failed" | "silence-timeout" | "closed";
+export type VoiceCallLifecycleEvent = {
+    at: number;
+    type: "start" | "end" | "transfer" | "escalation" | "voicemail" | "no-answer";
+    disposition?: VoiceCallDisposition;
+    metadata?: Record<string, unknown>;
+    reason?: string;
+    target?: string;
+};
+export type VoiceCallLifecycleState = {
+    disposition?: VoiceCallDisposition;
+    endedAt?: number;
+    events: VoiceCallLifecycleEvent[];
+    lastEventAt: number;
+    startedAt: number;
+};
+export type VoiceHandoffAction = "escalate" | "no-answer" | "transfer" | "voicemail";
+export type VoiceHandoffStatus = "delivered" | "failed" | "skipped";
+export type VoiceHandoffResult = {
+    deliveredAt?: number;
+    deliveredTo?: string;
+    error?: string;
+    metadata?: Record<string, unknown>;
+    status: VoiceHandoffStatus;
+};
+export type VoiceHandoffDeliveryQueueStatus = VoiceHandoffStatus | "pending";
+export type StoredVoiceHandoffDelivery<TContext = unknown, TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown> = {
+    action: VoiceHandoffAction;
+    context: TContext;
+    createdAt: number;
+    deliveredAt?: number;
+    deliveries?: Record<string, VoiceHandoffResult & {
+        adapterId: string;
+        adapterKind?: string;
+    }>;
+    deliveryAttempts?: number;
+    deliveryError?: string;
+    deliveryStatus: VoiceHandoffDeliveryQueueStatus;
+    id: string;
+    metadata?: Record<string, unknown>;
+    reason?: string;
+    result?: TResult;
+    session: TSession;
+    sessionId: string;
+    target?: string;
+    updatedAt: number;
+};
+export type VoiceHandoffDeliveryStore<TDelivery extends StoredVoiceHandoffDelivery = StoredVoiceHandoffDelivery> = {
+    get: (id: string) => Promise<TDelivery | undefined> | TDelivery | undefined;
+    list: () => Promise<TDelivery[]> | TDelivery[];
+    remove: (id: string) => Promise<void> | void;
+    set: (id: string, delivery: TDelivery) => Promise<void> | void;
+};
+export type VoiceHandoffInput<TContext = unknown, TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown> = {
+    action: VoiceHandoffAction;
+    api: VoiceSessionHandle<TContext, TSession, TResult>;
+    context: TContext;
+    metadata?: Record<string, unknown>;
+    reason?: string;
+    result?: TResult;
+    session: TSession;
+    target?: string;
+};
+export type VoiceHandoffAdapter<TContext = unknown, TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown> = {
+    actions?: VoiceHandoffAction[];
+    handoff: (input: VoiceHandoffInput<TContext, TSession, TResult>) => Promise<VoiceHandoffResult> | VoiceHandoffResult;
+    id: string;
+    kind?: string;
+};
+export type VoiceHandoffConfig<TContext = unknown, TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown> = {
+    adapters: VoiceHandoffAdapter<TContext, TSession, TResult>[];
+    deliveryQueue?: VoiceHandoffDeliveryStore<StoredVoiceHandoffDelivery<TContext, TSession, TResult>>;
+    enqueueOnly?: boolean;
+    failMode?: "record" | "throw";
+};
+export type VoiceSessionStore<TSession extends VoiceSessionRecord = VoiceSessionRecord> = {
+    get: (id: string) => Promise<TSession | undefined>;
+    getOrCreate: (id: string) => Promise<TSession>;
+    set: (id: string, value: TSession) => Promise<void>;
+    list: () => Promise<VoiceSessionSummary[]>;
+    remove: (id: string) => Promise<void>;
+};
+export type VoiceLogger = {
+    debug?: (message: string, meta?: Record<string, unknown>) => void;
+    info?: (message: string, meta?: Record<string, unknown>) => void;
+    warn?: (message: string, meta?: Record<string, unknown>) => void;
+    error?: (message: string, meta?: Record<string, unknown>) => void;
+};
+export type VoiceReconnectConfig = {
+    strategy?: "resume-last-turn" | "restart" | "fail";
+    timeout?: number;
+    maxAttempts?: number;
+};
+export type VoiceRuntimePreset = "default" | "chat" | "guided-intake" | "dictation" | "noisy-room" | "pstn-balanced" | "pstn-fast" | "reliability";
+export type VoiceSTTLifecycle = "continuous" | "turn-scoped";
+export type VoiceTurnProfile = "fast" | "balanced" | "long-form";
+export type VoiceTurnQualityProfile = "general" | "accent-heavy" | "noisy-room" | "short-command";
+export type VoiceTurnFallbackTrigger = "empty-turn" | "low-confidence" | "empty-or-low-confidence" | "always";
+export type VoiceFallbackTriggerReason = "always" | "empty-turn" | "risk-policy" | "transcript-confidence" | "word-confidence";
+export type VoiceSTTFallbackCandidate = {
+    averageConfidence: number;
+    lowestWordConfidence?: number;
+    text: string;
+    transcripts: Transcript[];
+    wordCount: number;
+    words: TranscriptWord[];
+};
+export type VoiceSTTFallbackRiskPolicy = (candidate: VoiceSTTFallbackCandidate) => boolean;
+export type VoiceSTTFallbackConfig = {
+    adapter: STTAdapter;
+    trigger?: VoiceTurnFallbackTrigger;
+    confidenceThreshold?: number;
+    minTextLength?: number;
+    replayWindowMs?: number;
+    settleMs?: number;
+    completionTimeoutMs?: number;
+    maxAttemptsPerTurn?: number;
+    /** Run the fallback when any provider word falls below this value. */
+    wordConfidenceThreshold?: number;
+    /** Application-specific routing for high-value turns, entities, or correction
+     * language that should receive a second transcription pass regardless of the
+     * aggregate confidence. */
+    riskPolicy?: VoiceSTTFallbackRiskPolicy;
+    /** Prefer a non-empty independent fallback for these audited trigger reasons,
+     * including providers that do not expose comparable confidence scores. */
+    preferFallbackOn?: VoiceFallbackTriggerReason[];
+};
+export type VoiceResolvedSTTFallbackConfig = {
+    adapter: STTAdapter;
+    trigger: VoiceTurnFallbackTrigger;
+    confidenceThreshold: number;
+    minTextLength: number;
+    replayWindowMs: number;
+    settleMs: number;
+    completionTimeoutMs: number;
+    maxAttemptsPerTurn: number;
+    wordConfidenceThreshold?: number;
+    riskPolicy?: VoiceSTTFallbackRiskPolicy;
+    preferFallbackOn?: VoiceFallbackTriggerReason[];
+};
+export type VoiceTurnDetectionConfig = {
+    profile?: VoiceTurnProfile;
+    qualityProfile?: VoiceTurnQualityProfile;
+    silenceMs?: number;
+    minSilenceMs?: number;
+    speechThreshold?: number;
+    transcriptStabilityMs?: number;
+};
+export type VoiceResolvedTurnDetectionConfig = {
+    qualityProfile: VoiceTurnQualityProfile;
+    profile: VoiceTurnProfile;
+    silenceMs: number;
+    minSilenceMs: number;
+    speechThreshold: number;
+    transcriptStabilityMs: number;
+};
+export type VoiceAudioConditioningConfig = {
+    enabled?: boolean;
+    targetLevel?: number;
+    maxGain?: number;
+    noiseGateThreshold?: number;
+    noiseGateAttenuation?: number;
+};
+export type VoiceResolvedAudioConditioningConfig = {
+    enabled: true;
+    targetLevel: number;
+    maxGain: number;
+    noiseGateThreshold: number;
+    noiseGateAttenuation: number;
+};
+export type VoiceSocket = {
+    send: (data: string | Uint8Array | ArrayBuffer) => void | Promise<void>;
+    close: (code?: number, reason?: string) => void | Promise<void>;
+    /**
+     * Discard any audio already buffered downstream (e.g. a telephony carrier's
+     * outbound media buffer). Called on barge-in so the caller stops hearing the
+     * assistant immediately, even when frames have already been flushed to the
+     * transport. Optional: transports without a flush concept omit it.
+     */
+    clear?: () => void | Promise<void>;
+};
+export type VoiceMonitorRuntimeSessionBinding = {
+    deregister: (reason?: string) => void;
+    emitAudio: (chunk: Uint8Array | ArrayBuffer, options?: {
+        source?: "assistant" | "caller" | (string & {});
+    }) => void;
+};
+export type VoiceMonitorRuntimeRegisterInput = {
+    handle: VoiceSessionHandle<unknown, VoiceSessionRecord, unknown>;
+    metadata?: Record<string, unknown>;
+    sessionId: string;
+};
+export type VoiceMonitorRuntimeBinding = {
+    registerSession: (input: VoiceMonitorRuntimeRegisterInput) => VoiceMonitorRuntimeSessionBinding;
+};
+export type VoiceSessionHandle<TContext = unknown, TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown> = {
+    id: string;
+    connect: (socket: VoiceSocket) => Promise<void>;
+    receiveAudio: (audio: AudioChunk) => Promise<void>;
+    attachUserMedia: (attachment: import("./agent").VoiceAgentMessageAttachment) => Promise<void>;
+    commitTurn: (reason?: VoiceEndOfTurnEvent["reason"]) => Promise<void>;
+    disconnect: (event?: VoiceCloseEvent) => Promise<void>;
+    complete: (result?: TResult) => Promise<void>;
+    escalate: (input: {
+        metadata?: Record<string, unknown>;
+        reason: string;
+        result?: TResult;
+    }) => Promise<void>;
+    fail: (error: unknown) => Promise<void>;
+    markNoAnswer: (input?: {
+        metadata?: Record<string, unknown>;
+        result?: TResult;
+    }) => Promise<void>;
+    markVoicemail: (input?: {
+        metadata?: Record<string, unknown>;
+        result?: TResult;
+    }) => Promise<void>;
+    transfer: (input: {
+        metadata?: Record<string, unknown>;
+        reason?: string;
+        result?: TResult;
+        target: string;
+        transferMode?: "cold" | "warm";
+    }) => Promise<void>;
+    close: (reason?: string) => Promise<void>;
+    /**
+     * Caller-driven in-call pause: the session stays LIVE (socket, state,
+     * transcript) but every idle/silence watchdog is suspended and turn
+     * processing is skipped, so a paused caller is never nudged or hung up on.
+     * Past `pause.maxMs` (default 10 min) the session closes gracefully with
+     * reason "pause-timeout" — completed turns stay durably persisted, so the
+     * host's normal draft/resume path takes over.
+     */
+    pause: () => Promise<void>;
+    /** Lift a caller-driven pause: watchdogs re-arm and turns process again. */
+    resume: () => Promise<void>;
+    snapshot: () => Promise<TSession>;
+    /** Ask connected browser clients to change assistant-audio playback speed. */
+    setPlaybackRate: (rate: number) => Promise<number>;
+    /**
+     * Mutate the live turn-detection config for this session — useful when a
+     * tool call wants to dial silenceMs up ("the caller asked for more time")
+     * or down. The change takes effect on the NEXT silence-timer schedule, so
+     * an in-flight commit isn't cancelled. Returns the merged config so the
+     * caller can confirm.
+     */
+    setTurnDetection: (patch: Partial<VoiceTurnDetectionConfig>) => Promise<{
+        silenceMs: number;
+        speechThreshold: number;
+        transcriptStabilityMs: number;
+    }>;
+    /** Refresh vocabulary, language hints, or provider turn settings while a
+     * call is active. Unsupported fields are ignored by the active adapter. */
+    configureSTT: (configuration: VoiceSTTSessionConfiguration) => Promise<void>;
+};
+export type VoiceLLMUsage = {
+    provider?: string;
+    model?: string;
+    inputTokens?: number;
+    outputTokens?: number;
+    cachedInputTokens?: number;
+};
+export type VoiceRouteResult<TResult = unknown> = {
+    complete?: boolean;
+    result?: TResult;
+    assistantText?: string;
+    usage?: VoiceLLMUsage;
+    citations?: ReadonlyArray<VoiceTurnCitation>;
+    transfer?: {
+        metadata?: Record<string, unknown>;
+        reason?: string;
+        target: string;
+        transferMode?: "cold" | "warm";
+    };
+    escalate?: {
+        metadata?: Record<string, unknown>;
+        reason: string;
+    };
+    voicemail?: {
+        metadata?: Record<string, unknown>;
+    };
+    noAnswer?: {
+        metadata?: Record<string, unknown>;
+    };
+};
+export type VoiceTurnCorrectionResult = string | {
+    text: string;
+    reason?: string;
+    provider?: string;
+    metadata?: Record<string, unknown>;
+};
+export type VoiceTurnCorrectionHandler<TContext = unknown, TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown> = (input: {
+    api: VoiceSessionHandle<TContext, TSession, TResult>;
+    context: TContext;
+    fallback?: VoiceFallbackDiagnostics;
+    lexicon: VoiceLexiconEntry[];
+    phraseHints: VoicePhraseHint[];
+    session: TSession;
+    text: string;
+    transcripts: Transcript[];
+}) => Promise<VoiceTurnCorrectionResult | void> | VoiceTurnCorrectionResult | void;
+export type VoicePhraseHintResolver<TContext = unknown> = (input: {
+    context: TContext;
+    scenarioId?: string;
+    sessionId: string;
+}) => Promise<VoicePhraseHint[] | void> | VoicePhraseHint[] | void;
+export type VoiceLexiconResolver<TContext = unknown> = (input: {
+    context: TContext;
+    scenarioId?: string;
+    sessionId: string;
+}) => Promise<VoiceLexiconEntry[] | void> | VoiceLexiconEntry[] | void;
+export type VoiceOnTurnObjectHandler<TContext = unknown, TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown> = (input: {
+    context: TContext;
+    liveOps?: {
+        control: VoiceLiveOpsControlState;
+        injectedInstruction?: string;
+    };
+    onTextDelta?: (delta: string) => void;
+    session: TSession;
+    turn: VoiceTurnRecord;
+    api: VoiceSessionHandle<TContext, TSession, TResult>;
+    speculativeReply?: {
+        text: string;
+    };
+}) => Promise<VoiceRouteResult<TResult> | void> | VoiceRouteResult<TResult> | void;
+export type VoiceOnTurnHandler<TContext = unknown, TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown> = VoiceOnTurnObjectHandler<TContext, TSession, TResult> | ((session: TSession, turn: VoiceTurnRecord, api: VoiceSessionHandle<TContext, TSession, TResult>, context: TContext) => Promise<VoiceRouteResult<TResult> | void> | VoiceRouteResult<TResult> | void);
+export type VoiceRouteConfig<TContext = unknown, TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown> = {
+    onCallStart?: (input: {
+        context: TContext;
+        session: TSession;
+        api: VoiceSessionHandle<TContext, TSession, TResult>;
+    }) => Promise<void> | void;
+    onCallEnd?: (input: {
+        api: VoiceSessionHandle<TContext, TSession, TResult>;
+        context: TContext;
+        disposition: VoiceCallDisposition;
+        metadata?: Record<string, unknown>;
+        reason?: string;
+        session: TSession;
+        target?: string;
+    }) => Promise<void> | void;
+    onSession?: (input: {
+        context: TContext;
+        session: TSession;
+        api: VoiceSessionHandle<TContext, TSession, TResult>;
+    }) => Promise<void> | void;
+    /** Fires when a connect RESUMES an existing session (a reconnect, or a fresh
+     *  process after a restart finding the session in a persistent store) instead
+     *  of starting a new call — onCallStart/onSession do NOT fire on a resume.
+     *  Use it to rebuild any per-session in-memory state (caller context, paced
+     *  flags) that didn't survive the process, so the resumed call keeps the
+     *  personalization the original had. Runs before the resume re-greeting. */
+    onResume?: (input: {
+        context: TContext;
+        session: TSession;
+        api: VoiceSessionHandle<TContext, TSession, TResult>;
+    }) => Promise<void> | void;
+    correctTurn?: VoiceTurnCorrectionHandler<TContext, TSession, TResult>;
+    speculate?: (input: {
+        api: VoiceSessionHandle<TContext, TSession, TResult>;
+        context: TContext;
+        session: TSession;
+        turn: VoiceTurnRecord;
+        signal?: AbortSignal;
+    }) => Promise<{
+        text: string;
+    } | null | void>;
+    onTurn: VoiceOnTurnHandler<TContext, TSession, TResult>;
+    onComplete: (input: {
+        context: TContext;
+        session: TSession;
+        api: VoiceSessionHandle<TContext, TSession, TResult>;
+    }) => Promise<void> | void;
+    onError?: (input: {
+        context: TContext;
+        sessionId: string;
+        session?: TSession;
+        error: unknown;
+        api?: VoiceSessionHandle<TContext, TSession, TResult>;
+    }) => Promise<void> | void;
+    onEscalation?: (input: {
+        api: VoiceSessionHandle<TContext, TSession, TResult>;
+        context: TContext;
+        metadata?: Record<string, unknown>;
+        reason: string;
+        session: TSession;
+    }) => Promise<void> | void;
+    onNoAnswer?: (input: {
+        api: VoiceSessionHandle<TContext, TSession, TResult>;
+        context: TContext;
+        metadata?: Record<string, unknown>;
+        session: TSession;
+    }) => Promise<void> | void;
+    onTransfer?: (input: {
+        api: VoiceSessionHandle<TContext, TSession, TResult>;
+        context: TContext;
+        metadata?: Record<string, unknown>;
+        reason?: string;
+        session: TSession;
+        target: string;
+    }) => Promise<void> | void;
+    onVoicemail?: (input: {
+        api: VoiceSessionHandle<TContext, TSession, TResult>;
+        context: TContext;
+        metadata?: Record<string, unknown>;
+        session: TSession;
+    }) => Promise<void> | void;
+};
+export type VoiceRuntimeOpsConfig<TContext = unknown, TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown> = {
+    buildReview?: (input: {
+        api: VoiceSessionHandle<TContext, TSession, TResult>;
+        context: TContext;
+        disposition: VoiceCallDisposition;
+        metadata?: Record<string, unknown>;
+        reason?: string;
+        result?: TResult;
+        session: TSession;
+        target?: string;
+    }) => Promise<VoiceCallReviewArtifact | StoredVoiceCallReviewArtifact | void> | VoiceCallReviewArtifact | StoredVoiceCallReviewArtifact | void;
+    createTaskFromReview?: (input: {
+        api: VoiceSessionHandle<TContext, TSession, TResult>;
+        context: TContext;
+        disposition: VoiceCallDisposition;
+        review: StoredVoiceCallReviewArtifact;
+        session: TSession;
+    }) => Promise<Omit<VoiceOpsTask, "id"> | VoiceOpsTask | StoredVoiceOpsTask | null | void> | Omit<VoiceOpsTask, "id"> | VoiceOpsTask | StoredVoiceOpsTask | null | void;
+    resolveTaskPolicy?: (input: {
+        api: VoiceSessionHandle<TContext, TSession, TResult>;
+        context: TContext;
+        disposition: VoiceCallDisposition;
+        metadata?: Record<string, unknown>;
+        reason?: string;
+        review?: StoredVoiceCallReviewArtifact;
+        session: TSession;
+        target?: string;
+        task: StoredVoiceOpsTask;
+    }) => Promise<VoiceOpsTaskPolicy | void> | VoiceOpsTaskPolicy | void;
+    resolveTaskAssignment?: (input: {
+        api: VoiceSessionHandle<TContext, TSession, TResult>;
+        context: TContext;
+        disposition: VoiceCallDisposition;
+        metadata?: Record<string, unknown>;
+        reason?: string;
+        review?: StoredVoiceCallReviewArtifact;
+        session: TSession;
+        target?: string;
+        task: StoredVoiceOpsTask;
+    }) => Promise<VoiceOpsTaskAssignmentRule | void> | VoiceOpsTaskAssignmentRule | void;
+    taskAssignmentRules?: VoiceOpsTaskAssignmentRules;
+    taskPolicies?: VoiceOpsDispositionTaskPolicies;
+    events?: VoiceIntegrationEventStore;
+    onEvent?: (input: {
+        api: VoiceSessionHandle<TContext, TSession, TResult>;
+        context: TContext;
+        event: StoredVoiceIntegrationEvent;
+        session: TSession;
+    }) => Promise<void> | void;
+    reviews?: VoiceCallReviewStore;
+    sinks?: VoiceIntegrationSink[];
+    tasks?: VoiceOpsTaskStore;
+    webhook?: VoiceIntegrationWebhookConfig;
+};
+export type VoiceLiveOpsRuntimeConfig = {
+    getControl: (sessionId: string) => Promise<VoiceLiveOpsControlState | null | undefined> | VoiceLiveOpsControlState | null | undefined;
+};
+export type VoiceProfileSwitchGuardResolverInput<TContext = unknown> = {
+    context: TContext;
+    scenarioId?: string;
+    sessionId: string;
+};
+export type VoicePluginProfileSwitchGuardConfig<TContext = unknown, TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown> = {
+    actor?: VoiceAuditActor;
+    allowedProfileIds?: string[] | ((input: VoiceProfileSwitchGuardResolverInput<TContext>) => Promise<string[] | undefined> | string[] | undefined);
+    audit?: VoiceAuditEventStore;
+    blockedProfileIds?: string[] | ((input: VoiceProfileSwitchGuardResolverInput<TContext>) => Promise<string[] | undefined> | string[] | undefined);
+    currentProfileId?: string | ((input: VoiceProfileSwitchGuardResolverInput<TContext>) => Promise<string | undefined> | string | undefined);
+    defaultProfileId?: string;
+    defaults: VoiceRealCallProfileDefaultsReport | VoiceRealCallProfileHistoryReport | ((input: VoiceProfileSwitchGuardResolverInput<TContext>) => Promise<VoiceRealCallProfileDefaultsReport | VoiceRealCallProfileHistoryReport> | VoiceRealCallProfileDefaultsReport | VoiceRealCallProfileHistoryReport);
+    metadata?: Record<string, unknown> | ((input: VoiceProfileSwitchGuardResolverInput<TContext>) => Promise<Record<string, unknown> | undefined> | Record<string, unknown> | undefined);
+    minConfidence?: number | ((input: VoiceProfileSwitchGuardResolverInput<TContext>) => Promise<number | undefined> | number | undefined);
+    maxAutoSwitchesPerSession?: number | ((input: VoiceProfileSwitchGuardResolverInput<TContext>) => Promise<number | undefined> | number | undefined);
+    mode?: VoiceProfileSwitchGuardMode | ((input: VoiceProfileSwitchGuardResolverInput<TContext>) => Promise<VoiceProfileSwitchGuardMode | undefined> | VoiceProfileSwitchGuardMode | undefined);
+    observed?: VoiceProfileSwitchObservedSignals | ((input: VoiceProfileSwitchGuardResolverInput<TContext>) => Promise<VoiceProfileSwitchObservedSignals | undefined> | VoiceProfileSwitchObservedSignals | undefined);
+    onDecision?: (input: {
+        context: TContext;
+        decision: VoiceProfileSwitchGuardDecision;
+        scenarioId?: string;
+        sessionId: string;
+    }) => Promise<void> | void;
+    sessionMetadataKey?: string | false;
+    trace?: false | VoiceTraceEventStore;
+};
+export type VoiceNormalizedRouteConfig<TContext = unknown, TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown> = Omit<VoiceRouteConfig<TContext, TSession, TResult>, "onTurn"> & {
+    onTurn: VoiceOnTurnObjectHandler<TContext, TSession, TResult>;
+};
+export type VoiceScenario = {
+    id: string;
+    name?: string;
+    description?: string;
+    metadata?: Record<string, unknown>;
+};
+export type VoiceExpectedSpeakerTurn = {
+    speaker: string;
+    text: string;
+};
+/**
+ * Configures one of the plugin's optional route surfaces. Pass that surface's
+ * options object to mount it, `false` (or omit) to leave it off. Surfaces whose
+ * options are entirely optional may also be enabled with `true` for defaults.
+ */
+export type VoiceSurfaceConfig<O> = false | O | (Record<string, never> extends O ? true : never);
+export type VoicePluginConfig<TContext = unknown, TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown> = {
+    costAccountant?: import("./costAccounting").VoiceCostAccountant;
+    costTelephony?: {
+        provider?: string;
+    };
+    costTelemetry?: VoiceCostTelemetryConfig<TContext, TSession, TResult>;
+    path: string;
+    /**
+     * Authorize a WebSocket before the plugin supersedes an existing session or
+     * creates provider-backed resources. Return false to close it with 4401.
+     * Applications can validate a short-lived admission credential from `query`
+     * without coupling Voice to a particular authentication implementation.
+     */
+    authorizeConnection?: (input: {
+        context: TContext;
+        headers: Headers;
+        path: string;
+        query: Record<string, unknown>;
+        scenarioId?: string;
+        sessionId: string;
+    }) => boolean | Promise<boolean>;
+    greeting?: string | ((input: {
+        session: TSession;
+    }) => string | Promise<string>);
+    resumeGreeting?: string | ((input: {
+        session: TSession;
+    }) => string | Promise<string>);
+    sttRecoveryLine?: string | ((input: {
+        session: TSession;
+    }) => string | Promise<string>);
+    stuckCallClose?: {
+        afterMs: number;
+        /** Whether this fault-driven close invokes the route completion hook.
+         * Defaults to true for backward compatibility. Set false when an
+         * interrupted conversation must remain resumable. */
+        invokeOnComplete?: boolean;
+        line?: string | ((input: {
+            session: TSession;
+        }) => string | Promise<string>);
+        reason?: string;
+    };
+    /** Gentle re-engagement before a silent call is given up on. When the caller
+     *  makes no progress (no committed turn / user partial) for `afterMs`, the
+     *  assistant speaks `line` ("still there?") instead of ending, and the close
+     *  deadline (stuckCallClose) is pushed out — up to `maxReprompts` times (default
+     *  1). After the budget is spent the normal close path runs. Caller speech
+     *  resets the budget. Assistant speech does NOT, so this can only extend a
+     *  bounded amount. `afterMs` should be < stuckCallClose.afterMs so the nudge
+     *  fires first. */
+    idleReprompt?: {
+        afterMs: number;
+        line: string | ((input: {
+            session: TSession;
+        }) => string | Promise<string>);
+        maxReprompts?: number;
+    };
+    languageStrategy?: VoiceLanguageStrategy | VoiceLanguageStrategyResolver<TContext>;
+    lexicon?: VoiceLexiconEntry[] | VoiceLexiconResolver<TContext>;
+    phraseHints?: VoicePhraseHint[] | VoicePhraseHintResolver<TContext>;
+    preset?: VoiceRuntimePreset;
+    stt?: STTAdapter;
+    sttFallback?: VoiceSTTFallbackConfig;
+    sttLifecycle?: VoiceSTTLifecycle;
+    realtime?: RealtimeAdapter;
+    realtimeInputFormat?: AudioFormat;
+    recording?: VoiceSessionRecordingConfig;
+    tts?: TTSAdapter;
+    session: VoiceSessionStore<NoInfer<TSession>>;
+    reconnect?: VoiceReconnectConfig;
+    turnDetection?: VoiceTurnDetectionConfig;
+    semanticTurnDetector?: import("./semanticTurn").VoiceSemanticTurnDetector;
+    bargeInDetector?: import("./bargeInDetector").VoiceBargeInDetector;
+    bargeInMinPartialWords?: number;
+    /**
+     * When true, a pure listening cue ("mm-hm", "yeah", "right", "got it") spoken
+     * WHILE the assistant is talking does NOT barge-in — the assistant keeps going
+     * and the cue is dropped so it never becomes the caller's next turn. A bare
+     * "yeah" said AFTER the assistant finishes is a normal answer, unaffected.
+     * Default false (any in-speech words interrupt, the prior behavior).
+     */
+    backchannelBargeInGuard?: boolean;
+    fillerPhrases?: ReadonlyArray<string>;
+    fillerDelayMs?: number;
+    fillerFor?: (input: {
+        sessionId: string;
+        turnId: string;
+        userText: string;
+    }) => Promise<string | null>;
+    fillerForTimeoutMs?: number;
+    backchannel?: import("./backchannel").VoiceBackchannelConfig;
+    defaultSilentTurnAck?: string;
+    routeOnTurnTimeoutMs?: number;
+    audioConditioning?: VoiceAudioConditioningConfig;
+    normalizeNumbers?: boolean;
+    noiseSuppressor?: VoiceNoiseSuppressor;
+    noiseSuppressorFormat?: AudioFormat;
+    logger?: VoiceLogger;
+    htmx?: boolean | VoiceHTMXConfig<TSession, NoInfer<TResult>>;
+    handoff?: VoiceHandoffConfig<TContext, TSession, TResult>;
+    ops?: VoiceRuntimeOpsConfig<TContext, TSession, TResult>;
+    liveOps?: VoiceLiveOpsRuntimeConfig;
+    monitor?: VoiceMonitorRuntimeBinding;
+    profileSwitchGuard?: VoicePluginProfileSwitchGuardConfig<TContext, TSession, TResult>;
+    trace?: VoiceTraceEventStore;
+    assistantHealth?: VoiceSurfaceConfig<import("./assistantHealth").VoiceAssistantHealthRoutesOptions>;
+    auditDelivery?: VoiceSurfaceConfig<import("./auditDeliveryRoutes").VoiceAuditDeliveryRoutesOptions>;
+    auditTrail?: VoiceSurfaceConfig<import("./auditRoutes").VoiceAuditTrailRoutesOptions>;
+    bargeIn?: VoiceSurfaceConfig<import("./bargeInRoutes").VoiceBargeInRoutesOptions>;
+    browserCallProfile?: VoiceSurfaceConfig<import("./browserCallProfiles").VoiceBrowserCallProfileRoutesOptions>;
+    browserMedia?: VoiceSurfaceConfig<import("./browserMediaRoutes").VoiceBrowserMediaRoutesOptions>;
+    callDebugger?: VoiceSurfaceConfig<import("./callDebugger").VoiceCallDebuggerRoutesOptions>;
+    campaign?: VoiceSurfaceConfig<import("./campaign").VoiceCampaignRoutesOptions>;
+    competitiveCoverage?: VoiceSurfaceConfig<import("./competitiveCoverage").VoiceCompetitiveCoverageRoutesOptions>;
+    dataControl?: VoiceSurfaceConfig<import("./dataControl").VoiceDataControlRoutesOptions>;
+    deliveryRuntime?: VoiceSurfaceConfig<import("./deliveryRuntime").VoiceDeliveryRuntimeRoutesOptions>;
+    deliverySink?: VoiceSurfaceConfig<import("./deliverySinkRoutes").VoiceDeliverySinkRoutesOptions>;
+    demoReady?: VoiceSurfaceConfig<import("./demoReadyRoutes").VoiceDemoReadyRoutesOptions>;
+    diagnostics?: VoiceSurfaceConfig<import("./diagnosticsRoutes").VoiceDiagnosticsRoutesOptions>;
+    eval?: VoiceSurfaceConfig<import("./evalRoutes").VoiceEvalRoutesOptions>;
+    guardrail?: VoiceSurfaceConfig<import("./guardrails").VoiceGuardrailRoutesOptions>;
+    handoffHealth?: VoiceSurfaceConfig<import("./handoffHealth").VoiceHandoffHealthRoutesOptions>;
+    htmxDashboard?: VoiceSurfaceConfig<import("./htmxDashboardRoutes").VoiceHTMXDashboardRoutesOptions>;
+    incidentBundle?: VoiceSurfaceConfig<import("./incidentBundle").VoiceIncidentBundleRoutesOptions>;
+    incidentTimeline?: VoiceSurfaceConfig<import("./incidentTimeline").VoiceIncidentTimelineRoutesOptions>;
+    liveLatency?: VoiceSurfaceConfig<import("./liveLatency").VoiceLiveLatencyRoutesOptions>;
+    liveMonitor?: VoiceSurfaceConfig<import("./monitor").VoiceLiveMonitorRoutesOptions>;
+    liveOpsConsole?: VoiceSurfaceConfig<import("./liveOps").VoiceLiveOpsRoutesOptions>;
+    mediaPipeline?: VoiceSurfaceConfig<import("./mediaPipelineRoutes").VoiceMediaPipelineRoutesOptions>;
+    monitorReport?: VoiceSurfaceConfig<import("./voiceMonitoring").VoiceMonitorRoutesOptions>;
+    monitorRunner?: VoiceSurfaceConfig<import("./voiceMonitoring").VoiceMonitorRunnerRoutesOptions>;
+    observabilityExport?: VoiceSurfaceConfig<import("./observabilityExport").VoiceObservabilityExportRoutesOptions>;
+    observabilityExportReplay?: VoiceSurfaceConfig<import("./observabilityExport").VoiceObservabilityExportReplayRoutesOptions>;
+    operationalStatus?: VoiceSurfaceConfig<import("./operationalStatus").VoiceOperationalStatusRoutesOptions>;
+    operationsRecord?: VoiceSurfaceConfig<import("./operationsRecord").VoiceOperationsRecordRoutesOptions>;
+    opsActionAudit?: VoiceSurfaceConfig<import("./opsActionAuditRoutes").VoiceOpsActionAuditRoutesOptions>;
+    opsConsole?: VoiceSurfaceConfig<import("./opsConsoleRoutes").VoiceOpsConsoleRoutesOptions>;
+    opsRecovery?: VoiceSurfaceConfig<import("./opsRecovery").VoiceOpsRecoveryRoutesOptions>;
+    opsStatus?: VoiceSurfaceConfig<import("./opsStatus").VoiceOpsStatusRoutesOptions>;
+    opsWebhookReceiver?: VoiceSurfaceConfig<import("./opsWebhook").VoiceOpsWebhookReceiverRoutesOptions>;
+    outcomeContract?: VoiceSurfaceConfig<import("./outcomeContract").VoiceOutcomeContractRoutesOptions>;
+    phoneAgentProductionSmoke?: VoiceSurfaceConfig<import("./phoneAgentProductionSmoke").VoicePhoneAgentProductionSmokeRoutesOptions>;
+    platformCoverage?: VoiceSurfaceConfig<import("./platformCoverage").VoicePlatformCoverageRoutesOptions>;
+    postCallAnalysis?: VoiceSurfaceConfig<import("./postCallAnalysis").VoicePostCallAnalysisRoutesOptions>;
+    productionReadiness?: VoiceSurfaceConfig<import("./productionReadiness").VoiceProductionReadinessRoutesOptions>;
+    profileSwitchLiveDecision?: VoiceSurfaceConfig<import("./profileSwitchRecommendation").VoiceProfileSwitchLiveDecisionRoutesOptions>;
+    profileSwitchPolicyProof?: VoiceSurfaceConfig<import("./profileSwitchRecommendation").VoiceProfileSwitchPolicyProofRoutesOptions>;
+    profileSwitchReadiness?: VoiceSurfaceConfig<import("./profileSwitchRecommendation").VoiceProfileSwitchReadinessRoutesOptions>;
+    proofPack?: VoiceSurfaceConfig<import("./proofPack").VoiceProofPackRoutesOptions>;
+    proofTrend?: VoiceSurfaceConfig<import("./proofTrends").VoiceProofTrendRoutesOptions>;
+    proofTrendRecommendation?: VoiceSurfaceConfig<import("./proofTrends").VoiceProofTrendRecommendationRoutesOptions>;
+    providerCapability?: VoiceSurfaceConfig<import("./providerCapabilities").VoiceProviderCapabilityRoutesOptions>;
+    providerContractMatrix?: VoiceSurfaceConfig<import("./providerStackRecommendations").VoiceProviderContractMatrixRoutesOptions>;
+    providerDecisionTrace?: VoiceSurfaceConfig<import("./providerDecisionTraces").VoiceProviderDecisionTraceRoutesOptions>;
+    providerHealth?: VoiceSurfaceConfig<import("./providerHealth").VoiceProviderHealthRoutesOptions>;
+    providerOrchestration?: VoiceSurfaceConfig<import("./providerOrchestration").VoiceProviderOrchestrationRoutesOptions>;
+    providerSlo?: VoiceSurfaceConfig<import("./providerSlo").VoiceProviderSloRoutesOptions>;
+    quality?: VoiceSurfaceConfig<import("./qualityRoutes").VoiceQualityRoutesOptions>;
+    realCallEvidenceRuntime?: VoiceSurfaceConfig<import("./proofTrends").VoiceRealCallEvidenceRuntimeRoutesOptions>;
+    realCallProfileHistory?: VoiceSurfaceConfig<import("./proofTrends").VoiceRealCallProfileHistoryRoutesOptions>;
+    realCallProfileRecoveryAction?: VoiceSurfaceConfig<import("./proofTrends").VoiceRealCallProfileRecoveryActionRoutesOptions>;
+    realtimeChannel?: VoiceSurfaceConfig<import("./realtimeChannel").VoiceRealtimeChannelRoutesOptions>;
+    realtimeProviderContract?: VoiceSurfaceConfig<import("./realtimeProviderContracts").VoiceRealtimeProviderContractRoutesOptions>;
+    reconnectContract?: VoiceSurfaceConfig<import("./reconnectContract").VoiceReconnectContractRoutesOptions>;
+    reconnectProof?: VoiceSurfaceConfig<import("./reconnectContract").VoiceReconnectProofRoutesOptions>;
+    resilience?: VoiceSurfaceConfig<import("./resilienceRoutes").VoiceResilienceRoutesOptions>;
+    sessionList?: VoiceSurfaceConfig<import("./sessionReplay").VoiceSessionListRoutesOptions>;
+    sessionObservability?: VoiceSurfaceConfig<import("./sessionObservability").VoiceSessionObservabilityRoutesOptions>;
+    sessionReplay?: VoiceSurfaceConfig<import("./sessionReplay").VoiceSessionReplayRoutesOptions>;
+    sessionSnapshot?: VoiceSurfaceConfig<import("./sessionSnapshot").VoiceSessionSnapshotRoutesOptions>;
+    simulationSuite?: VoiceSurfaceConfig<import("./simulationSuite").VoiceSimulationSuiteRoutesOptions>;
+    sloCalibration?: VoiceSurfaceConfig<import("./sloCalibration").VoiceSloCalibrationRoutesOptions>;
+    sloReadinessThreshold?: VoiceSurfaceConfig<import("./sloCalibration").VoiceSloReadinessThresholdRoutesOptions>;
+    telephonyCarrierMatrix?: VoiceSurfaceConfig<import("../telephony/matrix").VoiceTelephonyCarrierMatrixRoutesOptions>;
+    telephonyMedia?: VoiceSurfaceConfig<import("./telephonyMediaRoutes").VoiceTelephonyMediaRoutesOptions>;
+    telephonyWebhook?: VoiceSurfaceConfig<import("./telephonyOutcome").VoiceTelephonyWebhookRoutesOptions>;
+    telephonyWebhookSecurity?: VoiceSurfaceConfig<import("../telephony/security").VoiceTelephonyWebhookSecurityRoutesOptions>;
+    toolContract?: VoiceSurfaceConfig<import("./toolContract").VoiceToolContractRoutesOptions>;
+    traceDelivery?: VoiceSurfaceConfig<import("./traceDeliveryRoutes").VoiceTraceDeliveryRoutesOptions>;
+    traceTimeline?: VoiceSurfaceConfig<import("./traceTimeline").VoiceTraceTimelineRoutesOptions>;
+    turnLatency?: VoiceSurfaceConfig<import("./turnLatency").VoiceTurnLatencyRoutesOptions>;
+    turnQuality?: VoiceSurfaceConfig<import("./turnQuality").VoiceTurnQualityRoutesOptions>;
+} & VoiceRouteConfig<TContext, TSession, TResult>;
+export type VoiceSessionRecordingConfig = {
+    channels?: ReadonlyArray<"assistant" | "user">;
+    maxBytesPerChannel?: number;
+    store: import("./recordingStore").VoiceRecordingStore;
+    userInputFormat?: AudioFormat;
+};
+export type CreateVoiceSessionOptions<TContext = unknown, TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown> = {
+    costTelemetry?: VoiceCostTelemetryConfig<TContext, TSession, TResult>;
+    id: string;
+    context: TContext;
+    socket: VoiceSocket;
+    greeting?: string | ((input: {
+        session: TSession;
+    }) => string | Promise<string>);
+    /** Spoken on a RESUME (reconnect / restart) of a call that already had at
+     *  least one committed turn — a short re-orientation ("Sorry, you cut out —
+     *  you were telling me about X; go on") so the caller knows they're back,
+     *  since the in-flight assistant audio didn't survive. Receives the session so
+     *  it can reference the last turn. No resume greeting fires if unset. */
+    resumeGreeting?: string | ((input: {
+        session: TSession;
+    }) => string | Promise<string>);
+    /** Spoken when the STT-health watchdog detects the stream has gone DEAF
+     *  mid-call (continuous speech energy, no transcripts landing — see
+     *  STT_HEALTH_STALE_MS). A short re-prompt ("Sorry, I think I missed that — go
+     *  ahead?") so the caller repeats into the freshly reconnected stream instead
+     *  of talking into a silently dead call. Cooldown-guarded to fire at most once
+     *  per stale episode. Receives the session. Unset = silent reconnect only. */
+    sttRecoveryLine?: string | ((input: {
+        session: TSession;
+    }) => string | Promise<string>);
+    /** Last-resort GRACEFUL terminal close for a wedged call. If no caller-side
+     *  progress (committed turn / user partial) lands for `afterMs` on a live call
+     *  — STT permanently deaf, or the caller left — the assistant speaks `line` and
+     *  the session COMPLETES (disposition "completed") so onComplete still saves and
+     *  the call ends with a real goodbye instead of dead air + "abandoned". Reset by
+     *  real progress (committed turn / user partial / (re)connect), NOT by the
+     *  assistant's own speech, so STT recovery re-prompts can't defer it forever. */
+    stuckCallClose?: {
+        afterMs: number;
+        /** Whether this fault-driven close invokes the route completion hook.
+         * Defaults to true for backward compatibility. */
+        invokeOnComplete?: boolean;
+        line?: string | ((input: {
+            session: TSession;
+        }) => string | Promise<string>);
+        reason?: string;
+    };
+    /** Gentle re-engagement before a silent call is given up on. When the caller
+     *  makes no progress (no committed turn / user partial) for `afterMs`, the
+     *  assistant speaks `line` ("still there?") instead of ending, and the close
+     *  deadline (stuckCallClose) is pushed out — up to `maxReprompts` times (default
+     *  1). After the budget is spent the normal close path runs. Caller speech
+     *  resets the budget. Assistant speech does NOT, so this can only extend a
+     *  bounded amount. `afterMs` should be < stuckCallClose.afterMs so the nudge
+     *  fires first. */
+    idleReprompt?: {
+        afterMs: number;
+        line: string | ((input: {
+            session: TSession;
+        }) => string | Promise<string>);
+        maxReprompts?: number;
+    };
+    /** Caller-driven in-call pause (VoiceSessionHandle.pause/resume). `maxMs`
+     *  bounds how long a session may sit paused before it closes gracefully
+     *  with reason "pause-timeout" (default 10 minutes). */
+    pause?: {
+        maxMs?: number;
+    };
+    stt?: STTAdapter;
+    realtime?: RealtimeAdapter;
+    realtimeInputFormat?: AudioFormat;
+    tts?: TTSAdapter;
+    languageStrategy?: VoiceLanguageStrategy;
+    lexicon?: VoiceLexiconEntry[];
+    sttFallback?: VoiceResolvedSTTFallbackConfig;
+    store: VoiceSessionStore<TSession>;
+    trace?: VoiceTraceEventStore;
+    recording?: VoiceSessionRecordingConfig;
+    callSilenceTimeoutMs?: number;
+    amd?: import("./amdDetector").VoiceAMDDetector<TContext, TSession, TResult>;
+    costAccountant?: import("./costAccounting").VoiceCostAccountant;
+    costTelephony?: {
+        provider?: string;
+    };
+    redact?: import("./redaction").VoiceTranscriptRedactor;
+    semanticTurnDetector?: import("./semanticTurn").VoiceSemanticTurnDetector;
+    bargeInDetector?: import("./bargeInDetector").VoiceBargeInDetector;
+    /**
+     * Pre-rendered filler phrases the runtime plays in the gap between
+     * user-turn-commit and real assistant audio (typically 800-1500ms). The
+     * caller hears something within ~150-300ms of stopping speaking, so the
+     * LLM/TTS latency feels like the bot thinking instead of dead air. Boardy's
+     * killer UX feature.
+     *
+     * Behavior:
+     *  - After a turn commits, a timer fires at `fillerDelayMs` (default
+     *    250ms). At that point, if the real assistant audio for this turn
+     *    hasn't started flowing yet, a random phrase is rendered via the
+     *    configured `tts` adapter and pushed to the socket.
+     *  - When the real assistant audio's first chunk arrives, any in-flight
+     *    filler is cancelled (`cancelActiveTTS` clears the carrier buffer).
+     *  - Cooldown protects against double-fillers per turn.
+     *
+     * Set `fillerPhrases: []` (or omit) to disable. Reasonable defaults if
+     * you enable: `["Hmm.", "Got it.", "Right.", "Mm-hm.", "Let me think.", "Okay."]`.
+     */
+    /**
+     * Minimum word count in an STT partial transcript before speech-gated
+     * barge-in cancels the in-flight assistant TTS. Default 1 (any non-empty
+     * partial triggers barge-in — backwards-compatible).
+     *
+     * Set to 2 (or higher) on phone routes where the caller's brief
+     * acknowledgements ("yeah", "uh-huh", "you", "am i") would otherwise
+     * cut the bot off mid-question. Each extra word added typically delays
+     * barge-in by ~100-200ms (one extra STT partial cycle) — cheap compared
+     * to losing the bot's response.
+     *
+     * Word splitting is whitespace-based. Punctuation is left attached.
+     */
+    bargeInMinPartialWords?: number;
+    /**
+     * When true, a pure listening cue ("mm-hm", "yeah", "right", "got it") spoken
+     * WHILE the assistant is talking does NOT barge-in — the assistant keeps going
+     * and the cue is dropped so it never becomes the caller's next turn. A bare
+     * "yeah" said AFTER the assistant finishes is a normal answer, unaffected.
+     * Default false (any in-speech words interrupt, the prior behavior).
+     */
+    backchannelBargeInGuard?: boolean;
+    fillerPhrases?: ReadonlyArray<string>;
+    /** Milliseconds after turn-commit before the filler fires. Default 250ms — short enough to feel instant, long enough to skip if the LLM is very fast. */
+    fillerDelayMs?: number;
+    /**
+     * Latency Theater — content-aware filler (Boardy parity move). When
+     * defined, the runtime calls `fillerFor({ userText, ... })` in parallel
+     * with the main LLM call to generate a brief acknowledgement of what the
+     * caller just said ("Freelance CFOs — interesting.", "Yeah, I hear you.").
+     * The runtime races the promise against `fillerForTimeoutMs` (default
+     * 600ms). If `fillerFor` returns a non-empty string in time, it's spoken
+     * INSTEAD of a random `fillerPhrases` entry. On timeout or null return,
+     * the runtime falls back to a static random phrase, so a slow / failed
+     * acknowledgement call never costs you the filler entirely.
+     *
+     * Return `null` (or an empty string) to explicitly skip filler for this
+     * turn — useful when `userText` is so short ("yes", "no", "okay") that
+     * acknowledging it back sounds robotic. Return throws are caught and
+     * treated as null.
+     *
+     * Cost-aware: callers typically wire this to a cheap nano/haiku model
+     * (gpt-4.1-nano, claude-haiku-4-5) that returns 2–5 words.
+     */
+    fillerFor?: (input: {
+        sessionId: string;
+        turnId: string;
+        userText: string;
+    }) => Promise<string | null>;
+    /** Ceiling for the `fillerFor` call before we fall back to a static phrase. Default 600ms. */
+    fillerForTimeoutMs?: number;
+    /**
+     * Backchannel cues — short "mm-hm"/"right" acknowledgements played while the
+     * CALLER is mid-turn (a long answer) so they feel heard, the way a human
+     * listener interjects. Plays on the same non-turn TTS path as fillers, so it
+     * never registers as the assistant's turn or trips barge-in. Off unless
+     * `enabled` is set. Fires only while the assistant is silent.
+     */
+    backchannel?: import("./backchannel").VoiceBackchannelConfig;
+    /**
+     * Default spoken ack if the model returns ONLY tool calls (no text) and the
+     * turn isn't ending. Without this, the caller hears total silence after
+     * their turn and assumes the line dropped. Default is "Sorry, one moment."
+     * Set to "" to opt out entirely.
+     */
+    defaultSilentTurnAck?: string;
+    /**
+     * Hard timeout on a single `route.onTurn` call. If onTurn hasn't resolved
+     * in this many ms, it's rejected with a hard-timeout error which falls
+     * through to defaultSilentTurnAck. Default 45s — generous for normal
+     * conversational LLM calls (1-3s typical), but catches hangs where the
+     * model adapter's own timeout doesn't fire. Set to 0 to disable.
+     */
+    routeOnTurnTimeoutMs?: number;
+    assistantMode?: import("./assistantMode").VoiceAssistantMode;
+    modalities?: ReadonlyArray<"audio" | "text">;
+    prosody?: VoiceTTSProsody;
+    reconnect: Required<VoiceReconnectConfig>;
+    phraseHints?: VoicePhraseHint[];
+    sessionMetadata?: Record<string, unknown>;
+    scenarioId?: string;
+    sttLifecycle: VoiceSTTLifecycle;
+    turnDetection: VoiceResolvedTurnDetectionConfig;
+    audioConditioning?: VoiceResolvedAudioConditioningConfig;
+    normalizeNumbers?: boolean;
+    noiseSuppressor?: VoiceNoiseSuppressor;
+    noiseSuppressorFormat?: AudioFormat;
+    handoff?: VoiceHandoffConfig<TContext, TSession, TResult>;
+    liveOps?: VoiceLiveOpsRuntimeConfig;
+    route: VoiceNormalizedRouteConfig<TContext, TSession, TResult>;
+    logger?: VoiceLogger;
+};
+export type CreateVoiceSession = <TContext = unknown, TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown>(options: CreateVoiceSessionOptions<TContext, TSession, TResult>) => VoiceSessionHandle<TContext, TSession, TResult>;
+export type VoiceClientStartMessage = {
+    type: "start";
+    sessionId?: string;
+    scenarioId?: string;
+};
+export type VoiceClientEndTurnMessage = {
+    type: "end_turn";
+};
+export type VoiceClientCloseMessage = {
+    type: "close";
+    reason?: string;
+};
+export type VoiceClientCallControlMessage = {
+    type: "call_control";
+    requestId?: string;
+    action: "complete" | "escalate" | "no-answer" | "pause" | "resume" | "transfer" | "voicemail";
+    metadata?: Record<string, unknown>;
+    reason?: string;
+    target?: string;
+};
+export type VoiceServerCallControlAckMessage = {
+    type: "call_control_ack";
+    action: VoiceClientCallControlMessage["action"];
+    message?: string;
+    ok: boolean;
+    requestId: string;
+};
+export type VoiceClientPingMessage = {
+    type: "ping";
+};
+export type VoiceClientMessage = VoiceClientStartMessage | VoiceClientEndTurnMessage | VoiceClientCloseMessage | VoiceClientCallControlMessage | VoiceClientPingMessage;
+export type VoiceServerSessionMessage = {
+    type: "session";
+    paused?: boolean;
+    pauseExpiresAt?: number;
+    sessionId: string;
+    status: VoiceSessionStatus;
+    scenarioId?: string;
+    sessionMetadata?: Record<string, unknown>;
+};
+export type VoiceServerReplayMessage<TResult = unknown> = {
+    type: "replay";
+    assistantTexts: string[];
+    call?: VoiceCallLifecycleState;
+    partial: string;
+    scenarioId?: string;
+    sessionId: string;
+    sessionMetadata?: Record<string, unknown>;
+    status: VoiceSessionStatus;
+    turns: VoiceTurnRecord<TResult>[];
+};
+export type VoiceServerPartialMessage = {
+    type: "partial";
+    transcript: Transcript;
+};
+export type VoiceServerFinalMessage = {
+    type: "final";
+    transcript: Transcript;
+};
+export type VoiceServerTurnMessage<TResult = unknown> = {
+    type: "turn";
+    turn: VoiceTurnRecord<TResult>;
+};
+export type VoiceServerAssistantMessage = {
+    type: "assistant";
+    text: string;
+    turnId?: string;
+};
+export type VoiceServerAssistantDeltaMessage = {
+    type: "assistant_delta";
+    delta: string;
+    turnId?: string;
+};
+export type VoiceServerAudioMessage = {
+    type: "audio";
+    chunkBase64: string;
+    format: AudioFormat;
+    receivedAt: number;
+    turnId?: string;
+};
+/** Requests a live assistant-audio playback-rate change in browser clients. */
+export type VoiceServerPlaybackRateMessage = {
+    type: "playback_rate";
+    rate: number;
+};
+export type VoiceServerCompleteMessage = {
+    type: "complete";
+    sessionId: string;
+};
+export type VoiceServerCallLifecycleMessage = {
+    type: "call_lifecycle";
+    event: VoiceCallLifecycleEvent;
+    sessionId: string;
+};
+export type VoiceServerErrorMessage = {
+    type: "error";
+    message: string;
+    recoverable?: boolean;
+};
+export type VoiceServerPongMessage = {
+    type: "pong";
+};
+export type VoiceServerConnectionMessage = {
+    type: "connection";
+    reconnect: VoiceReconnectClientState;
+};
+export type VoiceServerMessage<TResult = unknown> = VoiceServerSessionMessage | VoiceServerReplayMessage<TResult> | VoiceServerPartialMessage | VoiceServerFinalMessage | VoiceServerTurnMessage<TResult> | VoiceServerAssistantMessage | VoiceServerAssistantDeltaMessage | VoiceServerAudioMessage | VoiceServerPlaybackRateMessage | VoiceServerCallControlAckMessage | VoiceServerCallLifecycleMessage | VoiceServerCompleteMessage | VoiceServerErrorMessage | VoiceServerPongMessage | VoiceServerConnectionMessage;
+export type VoiceConnectionOptions = {
+    browserMedia?: false | VoiceBrowserMediaReporterOptions;
+    protocols?: string[];
+    scenarioId?: string;
+    reconnect?: boolean;
+    reconnectReportPath?: string;
+    maxReconnectAttempts?: number;
+    /** Cap on the exponential reconnect backoff (ms). The delay doubles from 500ms
+     *  up to this ceiling each attempt, so the total retry window is roughly
+     *  maxReconnectAttempts spread across it. Default 8000 — with the default 15
+     *  attempts that's a ~95s window, enough to ride out a server redeploy without
+     *  the caller losing the call. */
+    reconnectMaxDelayMs?: number;
+    /** A reconnected socket must remain open this long before its retry budget is
+     *  reset. This prevents a server that accepts and immediately drops sockets
+     *  from creating an unbounded reconnect loop. Default 30000. */
+    reconnectResetAfterMs?: number;
+    /** Refresh application-owned admission state before each reconnect attempt.
+     *  Throwing keeps the transport disconnected and consumes one bounded retry
+     *  attempt instead of opening a socket with stale credentials. */
+    prepareReconnect?: (input: {
+        attempt: number;
+        path: string;
+        scenarioId: string | null;
+        sessionId: string;
+        signal: AbortSignal;
+    }) => Promise<void> | void;
+    /** Maximum time allowed for one prepareReconnect hook. A hook that exceeds
+     * this deadline is aborted and consumes the attempt. Default 10000. */
+    prepareReconnectTimeoutMs?: number;
+    pingInterval?: number;
+    /** Additional query values sent on every initial or reconnecting socket. */
+    query?: Record<string, string>;
+    sessionId?: string;
+};
+export type VoiceBrowserMediaReportPayload = {
+    at: number;
+    continuity?: MediaWebRTCStreamContinuityReport;
+    report: MediaWebRTCStatsReport;
+    scenarioId?: string | null;
+    sessionId?: string | null;
+};
+export type VoiceBrowserMediaReporterOptions = Omit<MediaWebRTCStatsReportInput, "peerConnection"> & {
+    fetch?: typeof fetch;
+    getPeerConnection?: (() => MediaWebRTCStatsCollector | null | undefined) | (() => Promise<MediaWebRTCStatsCollector | null | undefined>);
+    getScenarioId?: () => string | null | undefined;
+    getSessionId?: () => string | null | undefined;
+    intervalMs?: number;
+    continuity?: false | Omit<MediaWebRTCStreamContinuityInput, "previousStats" | "stats">;
+    onError?: (error: unknown) => void;
+    onReport?: (payload: VoiceBrowserMediaReportPayload) => void;
+    path?: string;
+    peerConnection?: MediaWebRTCStatsCollector;
+};
+export type VoiceCaptureOptions = {
+    channelCount?: 1 | 2;
+    onAudio?: (audio: Uint8Array | ArrayBuffer, sendAudio: (audio: Uint8Array | ArrayBuffer) => void) => void;
+    onLevel?: (level: number) => void;
+    sampleRateHz?: number;
+    /**
+     * A pre-acquired microphone MediaStream. When set, capture uses it instead of
+     * calling getUserMedia — so a host that requested permission UP FRONT (before
+     * connecting, so the prompt can't interrupt the assistant's greeting) can hand
+     * the SAME stream in rather than release-and-reacquire (a second getUserMedia
+     * + a track stop can trigger an audio-device change that suspends playback and
+     * cuts the greeting). Capture owns it after handoff and stops it on close.
+     */
+    stream?: MediaStream;
+};
+export type VoiceControllerOptions = {
+    preset?: VoiceRuntimePreset;
+    connection?: VoiceConnectionOptions;
+    capture?: VoiceCaptureOptions;
+    autoStopOnComplete?: boolean;
+};
+export type VoiceBargeInOptions = {
+    enabled?: boolean;
+    interruptOnPartial?: boolean;
+    interruptThreshold?: number;
+    monitor?: VoiceBargeInMonitor;
+};
+export type VoiceBargeInTriggerReason = "input-level" | "manual-audio" | "manual-interrupt" | "partial-transcript";
+export type VoiceBargeInMonitorEvent = {
+    at: number;
+    id: string;
+    latencyMs?: number;
+    playbackStopLatencyMs?: number;
+    reason: VoiceBargeInTriggerReason;
+    sessionId?: string | null;
+    status: "requested" | "stopped" | "skipped";
+    thresholdMs?: number;
+};
+export type VoiceBargeInMonitorSnapshot = {
+    averageLatencyMs?: number;
+    events: VoiceBargeInMonitorEvent[];
+    failed: number;
+    lastEvent?: VoiceBargeInMonitorEvent;
+    passed: number;
+    status: "empty" | "fail" | "pass" | "warn";
+    thresholdMs: number;
+    total: number;
+};
+export type VoiceBargeInMonitor = {
+    getSnapshot: () => VoiceBargeInMonitorSnapshot;
+    recordRequested: (input: {
+        reason: VoiceBargeInTriggerReason;
+        sessionId?: string | null;
+    }) => VoiceBargeInMonitorEvent;
+    recordSkipped: (input: {
+        reason: VoiceBargeInTriggerReason;
+        sessionId?: string | null;
+    }) => VoiceBargeInMonitorEvent;
+    recordStopped: (input: {
+        latencyMs?: number;
+        playbackStopLatencyMs?: number;
+        reason: VoiceBargeInTriggerReason;
+        sessionId?: string | null;
+    }) => VoiceBargeInMonitorEvent;
+    subscribe: (subscriber: () => void) => () => void;
+};
+export type VoiceAudioPlayerOptions = {
+    autoStart?: boolean;
+    createAudioContext?: () => AudioContext;
+    lookaheadMs?: number;
+    /**
+     * Playback speed multiplier for the assistant's speech. 1 = normal. Clamped
+     * to [0.5, 2]. Pitch shifts with the rate (Web Audio playbackRate), so keep
+     * UI ranges modest (≈0.85–1.25) to stay natural. Can be changed live via
+     * setPlaybackRate — already-scheduled chunks keep their rate; new chunks
+     * adopt the new one.
+     */
+    playbackRate?: number;
+    volume?: number;
+};
+export type VoiceDuplexControllerOptions = VoiceControllerOptions & {
+    audioPlayer?: VoiceAudioPlayerOptions;
+    bargeIn?: VoiceBargeInOptions;
+};
+export type VoiceSTTRoutingGoal = "best" | "low-cost";
+export type VoiceSTTRoutingCorrectionMode = "generic" | "none" | "risky-turn";
+export type VoiceSTTRoutingStrategy = {
+    benchmarkSessionTarget: "deepgram-corrected" | "deepgram-flux";
+    correctionMode: VoiceSTTRoutingCorrectionMode;
+    goal: VoiceSTTRoutingGoal;
+    notes: string[];
+    preset: VoiceRuntimePreset;
+    sttLifecycle: VoiceSTTLifecycle;
+};
+export type VoiceHTMXRenderInput<TResult = unknown, TSession extends VoiceSessionRecord = VoiceSessionRecord> = {
+    assistantTexts: string[];
+    partial: string;
+    scenarioId?: string;
+    result?: TResult;
+    session?: TSession;
+    sessionId?: string;
+    status: VoiceSessionStatus | "idle";
+    turnCount: number;
+    turns: VoiceTurnRecord<TResult>[];
+};
+export type VoiceHTMXRenderConfig<TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown> = {
+    metrics?: (input: VoiceHTMXRenderInput<TResult, TSession>) => string;
+    status?: (input: VoiceHTMXRenderInput<TResult, TSession>) => string;
+    turns?: (input: VoiceHTMXRenderInput<TResult, TSession>) => string;
+    assistant?: (input: VoiceHTMXRenderInput<TResult, TSession>) => string;
+    result?: (input: VoiceHTMXRenderInput<TResult, TSession>) => string;
+    emptyState?: (kind: keyof VoiceHTMXTargets, input: VoiceHTMXRenderInput<TResult, TSession>) => string;
+};
+export type VoiceHTMXRenderer<TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown> = (input: VoiceHTMXRenderInput<TResult, TSession>) => string;
+export type VoiceHTMXTargets = {
+    assistant: string;
+    metrics: string;
+    result: string;
+    status: string;
+    turns: string;
+};
+export type VoiceHTMXOptions<TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown> = VoiceHTMXRenderConfig<TSession, TResult> & {
+    bootstrapRoute?: string;
+    route?: string;
+    targets?: Partial<VoiceHTMXTargets>;
+};
+export type VoiceHTMXConfig<TSession extends VoiceSessionRecord = VoiceSessionRecord, TResult = unknown> = VoiceHTMXRenderer<TSession, TResult> | VoiceHTMXOptions<TSession, TResult>;
+export type VoiceStreamState<TResult = unknown> = {
+    call: VoiceCallLifecycleState | null;
+    paused: boolean;
+    pauseExpiresAt?: number;
+    sessionMetadata: Record<string, unknown> | null;
+    sessionId: string | null;
+    scenarioId: string | null;
+    status: VoiceSessionStatus | "idle";
+    reconnect: VoiceReconnectClientState;
+    partial: string;
+    turns: VoiceTurnRecord<TResult>[];
+    assistantTexts: string[];
+    assistantStreamingText: string;
+    assistantAudio: Array<{
+        chunk: Uint8Array;
+        format: AudioFormat;
+        receivedAt: number;
+        turnId?: string;
+    }>;
+    error: string | null;
+    isConnected: boolean;
+    /** Latest server-requested assistant playback rate, or null if unchanged. */
+    playbackRate: number | null;
+};
+export type VoiceStream<TResult = unknown> = {
+    call: VoiceCallLifecycleState | null;
+    paused: boolean;
+    pauseExpiresAt?: number;
+    callControl: (message: Omit<VoiceClientCallControlMessage, "requestId" | "type">) => Promise<void>;
+    close: (reason?: string) => void;
+    /** Release the transport without terminally completing the server session. */
+    disconnect: () => void;
+    start: (input?: {
+        scenarioId?: string;
+        sessionId?: string;
+    }) => Promise<void>;
+    endTurn: () => void;
+    error: string | null;
+    getServerSnapshot: () => VoiceStreamState<TResult>;
+    getSnapshot: () => VoiceStreamState<TResult>;
+    isConnected: boolean;
+    partial: string;
+    playbackRate: number | null;
+    reconnect: VoiceReconnectClientState;
+    sendAudio: (audio: Uint8Array | ArrayBuffer) => void;
+    simulateDisconnect: () => void;
+    sessionId: string | null;
+    sessionMetadata: Record<string, unknown> | null;
+    scenarioId: string | null;
+    status: VoiceSessionStatus | "idle";
+    subscribe: (subscriber: () => void) => () => void;
+    turns: VoiceTurnRecord<TResult>[];
+    assistantTexts: string[];
+    assistantStreamingText: string;
+    assistantAudio: Array<{
+        chunk: Uint8Array;
+        format: AudioFormat;
+        receivedAt: number;
+        turnId?: string;
+    }>;
+};
+export type VoiceControllerState<TResult = unknown> = VoiceStreamState<TResult> & {
+    isRecording: boolean;
+    recordingError: string | null;
+};
+export type VoiceAudioPlayerState = {
+    activeSourceCount: number;
+    error: string | null;
+    isActive: boolean;
+    isPlaying: boolean;
+    lastInterruptLatencyMs?: number;
+    lastPlaybackStopLatencyMs?: number;
+    processedChunkCount: number;
+    queuedChunkCount: number;
+};
+export type VoiceAudioPlayerSource = {
+    assistantAudio: VoiceStreamState["assistantAudio"];
+    subscribe: (subscriber: () => void) => () => void;
+};
+/** Post-call playback integrity, for verifying the assistant audio actually
+ *  played cleanly (no overlap, no dropped/stalled packets) — emit it as a trace
+ *  so a garbled call is provable, not guessed. `ok` is the all-clear roll-up. */
+export type VoiceAudioIntegrity = {
+    ok: boolean;
+    chunksReceived: number;
+    chunksScheduled: number;
+    scheduledDurationMs: number;
+    gapCount: number;
+    totalGapMs: number;
+    maxGapMs: number;
+    maxConcurrentPlayers: number;
+    errorCount: number;
+};
+export type VoiceAudioPlayer = {
+    close: () => Promise<void>;
+    error: string | null;
+    /** Instantaneous RMS amplitude (0..1) of the assistant's audio output — for
+     *  driving a visualizer from the actual voice. 0 when idle / no analyser. */
+    getOutputLevel: () => number;
+    getSnapshot: () => VoiceAudioPlayerState;
+    /** Post-call playback-integrity roll-up (overlap, gaps, drops). */
+    getIntegritySummary: () => VoiceAudioIntegrity;
+    activeSourceCount: number;
+    isActive: boolean;
+    isPlaying: boolean;
+    interrupt: () => Promise<void>;
+    lastInterruptLatencyMs?: number;
+    lastPlaybackStopLatencyMs?: number;
+    pause: () => Promise<void>;
+    playbackRate: number;
+    processedChunkCount: number;
+    queuedChunkCount: number;
+    setPlaybackRate: (rate: number) => void;
+    setVolume: (volume: number) => void;
+    start: () => Promise<void>;
+    subscribe: (subscriber: () => void) => () => void;
+    volume: number;
+};
+export type VoiceBargeInBinding = {
+    close: () => void;
+    handleLevel: (level: number) => void;
+    sendAudio: (audio: Uint8Array | ArrayBuffer) => void;
+};
+export type VoiceController<TResult = unknown> = {
+    bindHTMX: (options: VoiceHTMXBindingOptions) => () => void;
+    call: VoiceCallLifecycleState | null;
+    paused: boolean;
+    pauseExpiresAt?: number;
+    callControl: (message: Omit<VoiceClientCallControlMessage, "requestId" | "type">) => Promise<void>;
+    close: (reason?: string) => void;
+    /** Release the transport while leaving the server session resumable. */
+    disconnect: () => void;
+    endTurn: () => void;
+    start: (input?: {
+        scenarioId?: string;
+        sessionId?: string;
+    }) => Promise<void>;
+    error: string | null;
+    getServerSnapshot: () => VoiceControllerState<TResult>;
+    getSnapshot: () => VoiceControllerState<TResult>;
+    isConnected: boolean;
+    isRecording: boolean;
+    partial: string;
+    playbackRate: number | null;
+    reconnect: VoiceReconnectClientState;
+    recordingError: string | null;
+    sendAudio: (audio: Uint8Array | ArrayBuffer) => void;
+    simulateDisconnect: () => void;
+    sessionId: string | null;
+    sessionMetadata: Record<string, unknown> | null;
+    scenarioId: string | null;
+    startRecording: () => Promise<void>;
+    status: VoiceSessionStatus | "idle";
+    stopRecording: () => void;
+    subscribe: (subscriber: () => void) => () => void;
+    toggleRecording: () => Promise<void>;
+    turns: VoiceTurnRecord<TResult>[];
+    assistantTexts: string[];
+    assistantStreamingText: string;
+    assistantAudio: Array<{
+        chunk: Uint8Array;
+        format: AudioFormat;
+        receivedAt: number;
+        turnId?: string;
+    }>;
+};
+export type VoiceDuplexController<TResult = unknown> = VoiceController<TResult> & {
+    audioPlayer: VoiceAudioPlayer;
+    interruptAssistant: () => Promise<void>;
+};
+export type VoiceHTMXBindingOptions = {
+    element: Element | string;
+    eventName?: string;
+    route?: string;
+    sessionQueryParam?: string;
+};
+export type VoiceStoreAction<TResult = unknown> = {
+    type: "session";
+    paused?: boolean;
+    pauseExpiresAt?: number;
+    sessionId: string;
+    sessionMetadata?: Record<string, unknown>;
+    scenarioId?: string;
+    status: VoiceSessionStatus;
+} | {
+    type: "replay";
+    assistantTexts: string[];
+    call?: VoiceCallLifecycleState;
+    partial: string;
+    scenarioId?: string;
+    sessionId: string;
+    sessionMetadata?: Record<string, unknown>;
+    status: VoiceSessionStatus;
+    turns: VoiceTurnRecord<TResult>[];
+} | {
+    type: "call_lifecycle";
+    event: VoiceCallLifecycleEvent;
+    sessionId: string;
+} | {
+    type: "partial";
+    transcript: Transcript;
+} | {
+    type: "final";
+    transcript: Transcript;
+} | {
+    type: "turn";
+    turn: VoiceTurnRecord<TResult>;
+} | {
+    type: "assistant";
+    text: string;
+    turnId?: string;
+} | {
+    type: "assistant_delta";
+    delta: string;
+    turnId?: string;
+} | {
+    type: "audio";
+    chunk: Uint8Array;
+    format: AudioFormat;
+    receivedAt: number;
+    turnId?: string;
+} | {
+    type: "complete";
+    sessionId: string;
+} | {
+    type: "error";
+    message: string;
+    recoverable?: boolean;
+} | {
+    type: "playback_rate";
+    rate: number;
+} | {
+    type: "connected";
+} | {
+    type: "connection";
+    reconnect: VoiceReconnectClientState;
+} | {
+    type: "disconnected";
+};