@zhivex-ai/openai 0.15.0 → 0.16.1

package/dist/live-webrtc-browser.js

@@ -0,0 +1,235 @@
+import { ConfigurationError, withTimeoutSignal } from "@zhivex-ai/core/runtime";
+import { createLiveSession, isOpenAILiveModel } from "./live-session.js";
+import { liveAnswer, liveSdp, liveSessionId } from "./live-webrtc-protocol.js";
+/** Experimental browser transport over the shared RealtimeSession lifecycle. */
+export async function experimentalConnectOpenAILiveWebRTC(options) {
+    const modelId = options.modelId ?? "gpt-live-1";
+    if (!isOpenAILiveModel(modelId))
+        throw new ConfigurationError("A supported GPT-Live-1 model is required.");
+    if (typeof options.exchangeSdp !== "function" || typeof options.releaseSession !== "function")
+        throw new ConfigurationError("Application SDP exchange and session cleanup callbacks are required.");
+    if (!options.mediaStream?.getAudioTracks().length)
+        throw new ConfigurationError("WebRTC requires an application-provided audio track.");
+    if (options.closeTimeoutMs !== undefined && (!Number.isSafeInteger(options.closeTimeoutMs) || options.closeTimeoutMs <= 0))
+        throw new ConfigurationError("Invalid close timeout.");
+    if (!options.peerConnectionFactory && typeof globalThis.RTCPeerConnection !== "function")
+        throw new ConfigurationError("This browser does not support RTCPeerConnection.");
+    const deadline = withTimeoutSignal({ timeoutMs: options.timeoutMs ?? 15_000, abortSignal: options.signal });
+    let pc;
+    try {
+        deadline.signal.throwIfAborted();
+        pc = (options.peerConnectionFactory ?? (() => new RTCPeerConnection({ iceServers: [] })))();
+    }
+    catch (error) {
+        deadline.cleanup();
+        throw error;
+    }
+    const tracks = [];
+    const enabled = [];
+    let channel;
+    let closed = false, acknowledged = false, sessionId;
+    let failure;
+    let receiver;
+    const queue = [];
+    let queuedBytes = 0;
+    let released;
+    let rejectInterrupted;
+    const interrupted = new Promise((_, reject) => { rejectInterrupted = reject; });
+    void interrupted.catch(() => { });
+    const bounded = (promise) => Promise.race([promise, interrupted]);
+    const report = (error) => { try {
+        options.onError?.(error);
+    }
+    catch { /* Teardown must proceed. */ } };
+    const release = () => {
+        if (acknowledged || !sessionId)
+            return Promise.resolve();
+        if (!released) {
+            // An aborted handshake signal must not cancel the backend's cleanup request.
+            released = (async () => {
+                let timer;
+                try {
+                    await Promise.race([options.releaseSession({ sessionId: sessionId }), new Promise((_, reject) => {
+                            timer = setTimeout(() => reject(new Error("Backend session cleanup timed out; finalization is unconfirmed.")), 5000);
+                        })]);
+                }
+                catch (error) {
+                    report(error instanceof Error ? error : new Error("Backend session cleanup failed; finalization is unconfirmed."));
+                }
+                finally {
+                    if (timer)
+                        clearTimeout(timer);
+                }
+            })();
+        }
+        return released;
+    };
+    const cleanup = () => {
+        if (closed)
+            return;
+        closed = true;
+        deadline.cleanup();
+        deadline.signal.removeEventListener("abort", abortSetup);
+        options.signal?.removeEventListener("abort", abortCaller);
+        pc.ontrack = null;
+        pc.onconnectionstatechange = null;
+        if (channel) {
+            channel.onmessage = null;
+            channel.onerror = null;
+            channel.onclose = null;
+            channel.close();
+        }
+        for (const track of tracks)
+            track.stop();
+        pc.close();
+        queue.length = 0;
+        queuedBytes = 0;
+        receiver?.reject(failure ?? new Error("WebRTC connection closed."));
+        receiver = undefined;
+        rejectInterrupted(failure ?? new Error("WebRTC connection closed."));
+    };
+    const fail = (error) => {
+        if (closed || acknowledged)
+            return;
+        failure = error;
+        cleanup();
+        deadline.abort(error);
+        void release();
+        report(error);
+    };
+    const abortSetup = () => fail(deadline.signal.reason instanceof Error ? deadline.signal.reason : new Error("WebRTC connection aborted; finalization is unconfirmed."));
+    const abortCaller = () => fail(options.signal?.reason instanceof Error ? options.signal.reason : new Error("WebRTC session aborted; finalization is unconfirmed."));
+    const connection = {
+        async sendJson(payload) {
+            if (closed || channel?.readyState !== "open")
+                throw new ConfigurationError("WebRTC data channel is not open.");
+            const json = JSON.stringify(payload), bytes = new TextEncoder().encode(json).length;
+            if (bytes > 64 * 1024 || bytes + channel.bufferedAmount > 256 * 1024)
+                throw new ConfigurationError("WebRTC outgoing buffer limit exceeded.");
+            channel.send(json);
+        },
+        async recvJson() {
+            if (failure)
+                throw failure;
+            if (queue.length) {
+                const entry = queue.shift();
+                queuedBytes -= entry.bytes;
+                return entry.payload;
+            }
+            if (closed)
+                return undefined;
+            if (receiver)
+                throw new ConfigurationError("Concurrent WebRTC receives are unsupported.");
+            return new Promise((resolve, reject) => { receiver = { resolve, reject }; });
+        },
+        async close() { cleanup(); await release(); }
+    };
+    deadline.signal.addEventListener("abort", abortSetup, { once: true });
+    try {
+        pc.ontrack = event => { try {
+            options.onTrack?.(event);
+        }
+        catch {
+            fail(new Error("Application track callback failed; finalization is unconfirmed."));
+        } };
+        pc.onconnectionstatechange = () => { if (["failed", "closed", "disconnected"].includes(pc.connectionState))
+            fail(new Error("WebRTC connection lost; final usage is unconfirmed.")); };
+        for (const source of options.mediaStream.getAudioTracks()) {
+            const track = source.clone();
+            tracks.push(track);
+            enabled.push(source.enabled);
+            track.enabled = false;
+            pc.addTrack(track, options.mediaStream);
+        }
+        channel = pc.createDataChannel("oai-events");
+        channel.onerror = () => fail(new Error("WebRTC data channel failed; finalization is unconfirmed."));
+        channel.onclose = () => fail(new Error("WebRTC data channel closed before session.closed; final usage is unconfirmed."));
+        channel.onmessage = event => {
+            if (closed)
+                return;
+            try {
+                if (typeof event.data !== "string")
+                    throw new Error("WebRTC events must be JSON text.");
+                const bytes = new TextEncoder().encode(event.data).length;
+                if (bytes > 64 * 1024)
+                    throw new Error("WebRTC event exceeds 64 KiB.");
+                const payload = JSON.parse(event.data);
+                if (!payload || Array.isArray(payload) || typeof payload.type !== "string")
+                    throw new Error("Invalid WebRTC event.");
+                if (payload.type === "session.started" && payload.session?.id !== sessionId)
+                    throw new Error("WebRTC session ID mismatch.");
+                if (payload.type === "session.closed") {
+                    if (payload.session?.id !== sessionId || !Number.isFinite(payload.usage?.seconds) || payload.usage.seconds < 0)
+                        throw new Error("Invalid WebRTC finalization event; final usage is unconfirmed.");
+                    acknowledged = true;
+                }
+                if (receiver) {
+                    const waiter = receiver;
+                    receiver = undefined;
+                    waiter.resolve(payload);
+                }
+                else {
+                    if (queue.length >= 128 || queuedBytes + bytes > 256 * 1024)
+                        throw new Error("WebRTC incoming queue limit exceeded.");
+                    queue.push({ payload, bytes });
+                    queuedBytes += bytes;
+                }
+            }
+            catch (error) {
+                fail(error instanceof Error ? error : new Error("Invalid WebRTC event."));
+            }
+        };
+        const offer = await bounded(pc.createOffer());
+        if (closed)
+            throw failure;
+        await bounded(pc.setLocalDescription(offer));
+        if (pc.iceGatheringState !== "complete")
+            await new Promise((resolve, reject) => {
+                const finish = (error) => { pc.removeEventListener("icegatheringstatechange", check); deadline.signal.removeEventListener("abort", abort); error ? reject(error) : resolve(); };
+                const check = () => { if (pc.iceGatheringState === "complete")
+                    finish(); };
+                const abort = () => finish(deadline.signal.reason);
+                pc.addEventListener("icegatheringstatechange", check);
+                deadline.signal.addEventListener("abort", abort, { once: true });
+                check();
+                if (deadline.signal.aborted)
+                    abort();
+            });
+        if (closed)
+            throw failure;
+        const exchange = options.exchangeSdp({ offerSdp: liveSdp(pc.localDescription?.sdp), signal: deadline.signal }).then(raw => {
+            sessionId = liveSessionId(raw?.session?.id);
+            const answer = liveAnswer(raw);
+            if (closed) {
+                void release();
+                throw failure ?? new Error("WebRTC closed during signaling.");
+            }
+            return answer;
+        });
+        const answer = await bounded(exchange);
+        await bounded(pc.setRemoteDescription({ type: "answer", sdp: answer.transport.sdp }));
+        if (closed)
+            throw failure;
+        const session = createLiveSession(modelId, { delegation: { type: "client" }, providerOptions: { closeTimeoutMs: options.closeTimeoutMs } }, connection, { started: true, webrtc: true, timeoutMs: options.timeoutMs });
+        await bounded(session.initialize());
+        if (closed)
+            throw failure;
+        deadline.cleanup();
+        deadline.signal.removeEventListener("abort", abortSetup);
+        options.signal?.addEventListener("abort", abortCaller, { once: true });
+        if (options.signal?.aborted) {
+            abortCaller();
+            throw failure;
+        }
+        tracks.forEach((track, index) => { track.enabled = enabled[index]; });
+        return Object.assign(session, { sessionId: answer.session.id, peerConnection: pc });
+    }
+    catch (error) {
+        failure ??= error instanceof Error ? error : new Error("WebRTC setup failed.");
+        deadline.abort(failure);
+        cleanup();
+        await release();
+        throw failure;
+    }
+}
+//# sourceMappingURL=live-webrtc-browser.js.map

package/dist/live-webrtc-browser.js.map

@@ -0,0 +1 @@
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package/dist/live-webrtc-protocol.d.ts

@@ -0,0 +1,15 @@
+/** Experimental, secret-free response returned to the browser. */
+export interface OpenAILiveWebRTCAnswer {
+    session: {
+        id: string;
+    };
+    transport: {
+        type: "webrtc";
+        sdp: string;
+    };
+}
+export declare const LIVE_SDP_MAX_BYTES: number;
+export declare function liveSdp(value: unknown): string;
+export declare function liveSessionId(value: unknown): string;
+export declare function liveAnswer(value: unknown): OpenAILiveWebRTCAnswer;
+//# sourceMappingURL=live-webrtc-protocol.d.ts.map

package/dist/live-webrtc-protocol.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"live-webrtc-protocol.d.ts","sourceRoot":"","sources":["../src/live-webrtc-protocol.ts"],"names":[],"mappings":"AAEA,kEAAkE;AAClE,MAAM,WAAW,sBAAsB;IACrC,OAAO,EAAE;QAAE,EAAE,EAAE,MAAM,CAAA;KAAE,CAAC;IACxB,SAAS,EAAE;QAAE,IAAI,EAAE,QAAQ,CAAC;QAAC,GAAG,EAAE,MAAM,CAAA;KAAE,CAAC;CAC5C;AACD,eAAO,MAAM,kBAAkB,QAAY,CAAC;AAC5C,wBAAgB,OAAO,CAAC,KAAK,EAAE,OAAO,GAAG,MAAM,CAK9C;AACD,wBAAgB,aAAa,CAAC,KAAK,EAAE,OAAO,GAAG,MAAM,CAGpD;AACD,wBAAgB,UAAU,CAAC,KAAK,EAAE,OAAO,GAAG,sBAAsB,CAKjE"}

package/dist/live-webrtc-protocol.js

@@ -0,0 +1,21 @@
+import { ConfigurationError } from "@zhivex-ai/core/runtime";
+export const LIVE_SDP_MAX_BYTES = 64 * 1024;
+export function liveSdp(value) {
+    if (typeof value !== "string" || !value.startsWith("v=0\r\n") && !value.startsWith("v=0\n") || new TextEncoder().encode(value).length > LIVE_SDP_MAX_BYTES) {
+        throw new ConfigurationError("GPT-Live requires a valid SDP of at most 64 KiB.");
+    }
+    return value;
+}
+export function liveSessionId(value) {
+    if (typeof value !== "string" || !value || value === "." || value === ".." || value.length > 256 || /[\s\x00-\x1f\x7f]/.test(value))
+        throw new ConfigurationError("Invalid GPT-Live session ID.");
+    return value;
+}
+export function liveAnswer(value) {
+    const result = value;
+    if (result?.transport?.type !== "webrtc")
+        throw new ConfigurationError("Invalid GPT-Live WebRTC answer.");
+    // Whitelist fields: never serialize the provider's session configuration or secrets.
+    return { session: { id: liveSessionId(result.session?.id) }, transport: { type: "webrtc", sdp: liveSdp(result.transport.sdp) } };
+}
+//# sourceMappingURL=live-webrtc-protocol.js.map

package/dist/live-webrtc-protocol.js.map

@@ -0,0 +1 @@
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package/dist/live-webrtc-server.d.ts

@@ -0,0 +1,34 @@
+import { type RealtimeConnectionFactory, type RealtimeConnectOptions, type RealtimeSessionConfig } from "@zhivex-ai/core/provider";
+import { type OpenAILiveWebRTCAnswer } from "./live-webrtc-protocol.js";
+export type { OpenAILiveWebRTCAnswer } from "./live-webrtc-protocol.js";
+export interface OpenAILiveServerOptions {
+    /** Trusted server only. No environment lookup or browser credential flow. */
+    apiKey: string;
+    fetch?: typeof fetch;
+    safetyIdentifier?: string;
+}
+export interface OpenAILiveWebRTCCreateOptions extends OpenAILiveServerOptions, Pick<RealtimeConnectOptions, "signal" | "timeoutMs"> {
+    modelId?: string;
+    offerSdp: string;
+    session?: RealtimeSessionConfig;
+    /** Bind the known ID to an application creation lease before delivering the answer. */
+    onSessionCreated?: (input: {
+        sessionId: string;
+    }) => void | Promise<void>;
+}
+/** Experimental trusted-server SDP exchange. One request, no retries. */
+export declare function experimentalCreateOpenAILiveWebRTCSession(options: OpenAILiveWebRTCCreateOptions): Promise<OpenAILiveWebRTCAnswer>;
+export interface OpenAILiveAttachOptions extends OpenAILiveServerOptions, RealtimeConnectOptions {
+    /** Authorize this ID against the authenticated user's server-side session record first. */
+    sessionId: string;
+    modelId?: string;
+    session?: RealtimeSessionConfig;
+    connectionFactory?: RealtimeConnectionFactory;
+}
+/** Experimental sideband for runRealtimeDelegations. Does not start another paid session. */
+export declare function experimentalAttachOpenAILiveSession(options: OpenAILiveAttachOptions): Promise<import("@zhivex-ai/core/provider").CallbackRealtimeSession & {
+    sessionId: string;
+    /** Detach only; the primary session continues billing. close() ends the shared session. */
+    disconnect: () => Promise<void>;
+}>;
+//# sourceMappingURL=live-webrtc-server.d.ts.map

package/dist/live-webrtc-server.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"live-webrtc-server.d.ts","sourceRoot":"","sources":["../src/live-webrtc-server.ts"],"names":[],"mappings":"AAAA,OAAO,EAEL,KAAK,yBAAyB,EAAE,KAAK,sBAAsB,EAAE,KAAK,qBAAqB,EACxF,MAAM,0BAA0B,CAAC;AAGlC,OAAO,EAAsC,KAAK,sBAAsB,EAAE,MAAM,2BAA2B,CAAC;AAC5G,YAAY,EAAE,sBAAsB,EAAE,MAAM,2BAA2B,CAAC;AAExE,MAAM,WAAW,uBAAuB;IACtC,6EAA6E;IAC7E,MAAM,EAAE,MAAM,CAAC;IACf,KAAK,CAAC,EAAE,OAAO,KAAK,CAAC;IACrB,gBAAgB,CAAC,EAAE,MAAM,CAAC;CAC3B;AACD,MAAM,WAAW,6BAA8B,SAAQ,uBAAuB,EAAE,IAAI,CAAC,sBAAsB,EAAE,QAAQ,GAAG,WAAW,CAAC;IAClI,OAAO,CAAC,EAAE,MAAM,CAAC;IACjB,QAAQ,EAAE,MAAM,CAAC;IACjB,OAAO,CAAC,EAAE,qBAAqB,CAAC;IAChC,uFAAuF;IACvF,gBAAgB,CAAC,EAAE,CAAC,KAAK,EAAE;QAAE,SAAS,EAAE,MAAM,CAAA;KAAE,KAAK,IAAI,GAAG,OAAO,CAAC,IAAI,CAAC,CAAC;CAC3E;AAMD,yEAAyE;AACzE,wBAAsB,yCAAyC,CAAC,OAAO,EAAE,6BAA6B,GAAG,OAAO,CAAC,sBAAsB,CAAC,CAkCvI;AACD,MAAM,WAAW,uBAAwB,SAAQ,uBAAuB,EAAE,sBAAsB;IAC9F,2FAA2F;IAC3F,SAAS,EAAE,MAAM,CAAC;IAClB,OAAO,CAAC,EAAE,MAAM,CAAC;IACjB,OAAO,CAAC,EAAE,qBAAqB,CAAC;IAChC,iBAAiB,CAAC,EAAE,yBAAyB,CAAC;CAC/C;AACD,6FAA6F;AAC7F,wBAAsB,mCAAmC,CAAC,OAAO,EAAE,uBAAuB;;IActF,2FAA2F;;GAG9F"}

package/dist/live-webrtc-server.js

@@ -0,0 +1,79 @@
+import { ConfigurationError, ProviderHTTPError, assertTrustedEndpoint, openWebSocketConnection, readJsonWithLimit, withTimeoutSignal } from "@zhivex-ai/core/provider";
+import { liveAudioCallbacks } from "./live-audio.js";
+import { createLiveSession, isOpenAILiveModel, resolveLiveWebRTCConfig } from "./live-session.js";
+import { liveAnswer, liveSdp, liveSessionId } from "./live-webrtc-protocol.js";
+const headers = (options) => {
+    if (typeof options.apiKey !== "string" || !options.apiKey.trim() || /[\r\n]/.test(options.apiKey))
+        throw new ConfigurationError("A server API key is required.");
+    if (options.safetyIdentifier !== undefined && (!/^[\x21-\x7e]{1,256}$/.test(options.safetyIdentifier)))
+        throw new ConfigurationError("Invalid safety identifier.");
+    return { authorization: `Bearer ${options.apiKey}`, ...(options.safetyIdentifier ? { "OpenAI-Safety-Identifier": options.safetyIdentifier } : {}) };
+};
+/** Experimental trusted-server SDP exchange. One request, no retries. */
+export async function experimentalCreateOpenAILiveWebRTCSession(options) {
+    const modelId = options.modelId ?? "gpt-live-1";
+    if (!isOpenAILiveModel(modelId))
+        throw new ConfigurationError("A supported GPT-Live-1 model is required.");
+    const config = resolveLiveWebRTCConfig(options.session);
+    const requestHeaders = headers(options);
+    const body = JSON.stringify({ session: {
+            model: modelId, delegation: { type: "client" }, store: config.providerOptions?.store ?? false,
+            ...(config.instructions !== undefined ? { instructions: config.instructions } : {}),
+            audio: { output: { voice: config.voice ?? "marin" } },
+            ...(config.providerOptions?.input !== undefined ? { input: config.providerOptions.input } : {}),
+            // The browser may close/mute, but cannot append instructions or backend results.
+            client: { data_channel: {
+                    allowed_client_events: ["session.close", "session.input_audio.mute", "session.input_audio.unmute"],
+                    allowed_server_events: ["session.started", "session.closed", "session.input_transcript.delta", "session.output_transcript.delta", "session.usage.updated", "error"].map(type => ({ type }))
+                } }
+        }, transport: { type: "webrtc", sdp: liveSdp(options.offerSdp) } });
+    if (new TextEncoder().encode(body).length > 256 * 1024)
+        throw new ConfigurationError("GPT-Live session request exceeds 256 KiB.");
+    const deadline = withTimeoutSignal({ timeoutMs: options.timeoutMs ?? 15_000, abortSignal: options.signal });
+    try {
+        deadline.signal.throwIfAborted();
+        const response = await (options.fetch ?? fetch)("https://api.openai.com/v1/live/sessions", {
+            method: "POST", headers: { ...requestHeaders, "content-type": "application/json" }, body, signal: deadline.signal, redirect: "error"
+        });
+        if (!response.ok) {
+            await response.body?.cancel();
+            throw new ProviderHTTPError(`OpenAI Live session creation failed (${response.status}).`, response.status);
+        }
+        const raw = await readJsonWithLimit(response, { maxBytes: 128 * 1024, provider: "openai", endpoint: "live/sessions", abort: deadline.abort });
+        // Record a valid ID even if the remaining response is malformed or the caller aborted.
+        const sessionId = liveSessionId(raw?.session?.id);
+        await options.onSessionCreated?.({ sessionId });
+        deadline.signal.throwIfAborted();
+        return liveAnswer(raw);
+    }
+    finally {
+        deadline.cleanup();
+    }
+}
+/** Experimental sideband for runRealtimeDelegations. Does not start another paid session. */
+export async function experimentalAttachOpenAILiveSession(options) {
+    const modelId = options.modelId ?? "gpt-live-1";
+    if (!isOpenAILiveModel(modelId))
+        throw new ConfigurationError("A supported GPT-Live-1 model is required.");
+    const config = resolveLiveWebRTCConfig(options.session);
+    const url = assertTrustedEndpoint(`wss://api.openai.com/v1/live/sessions/${encodeURIComponent(liveSessionId(options.sessionId))}/attach`, {
+        label: "OpenAI Live sideband", protocols: ["wss"], allowedHosts: ["api.openai.com"]
+    });
+    const connection = await (options.connectionFactory ?? openWebSocketConnection)(url.toString(), headers(options), { signal: options.signal, timeoutMs: options.timeoutMs ?? 15_000, maxIncomingFrameBytes: options.maxIncomingFrameBytes, subprotocols: options.subprotocols });
+    // Attachment observes an already-running session: no session.start or started wait.
+    const reflected = { ...config, outputAudioMediaType: "audio/pcm", outputSampleRateHz: 24000 };
+    const session = createLiveSession(modelId, reflected, connection, { expectedSessionId: options.sessionId, started: true, attached: true, parseOutputAudio: liveAudioCallbacks(reflected).parseOutputAudio });
+    try {
+        await session.initialize();
+    }
+    catch (error) {
+        await connection.close();
+        throw error;
+    }
+    return Object.assign(session, {
+        sessionId: options.sessionId,
+        /** Detach only; the primary session continues billing. close() ends the shared session. */
+        disconnect: () => connection.close()
+    });
+}
+//# sourceMappingURL=live-webrtc-server.js.map

package/dist/live-webrtc-server.js.map

@@ -0,0 +1 @@
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package/dist/live.d.ts

@@ -1,6 +1,5 @@
-import { CallbackRealtimeSession, type ModelCapabilities, type RealtimeConnectOptions, type RealtimeConnectionFactory, type RealtimeModel, type RealtimeSessionConfig } from "@zhivex-ai/core/provider";
-export declare const isOpenAILiveModel: (modelId: string) => boolean;
-export declare const liveCapabilities: ModelCapabilities;
+import { type RealtimeModel, type RealtimeConnectOptions, type RealtimeConnectionFactory, type RealtimeSessionConfig } from "@zhivex-ai/core/provider";
+export { isOpenAILiveModel } from "./live-session.js";
 /** Server WebSocket adapter. The application owns the delegated agent and its tools. */
 export declare class OpenAILiveModel implements RealtimeModel {
     readonly modelId: string;
@@ -10,9 +9,9 @@ export declare class OpenAILiveModel implements RealtimeModel {
     private readonly realtimeURL?;
     private readonly allowUnsafeEndpoints;
     readonly provider = "openai";
-    readonly capabilities: ModelCapabilities;
+    readonly capabilities: import("@zhivex-ai/core/provider").ModelCapabilities;
     constructor(modelId: string, baseURL: string, headers: (options: Record<string, unknown> | undefined) => Record<string, string>, connectionFactory: RealtimeConnectionFactory, realtimeURL?: string | undefined, allowUnsafeEndpoints?: boolean);
-    connect(config?: RealtimeSessionConfig, options?: RealtimeConnectOptions): Promise<CallbackRealtimeSession>;
+    connect(config?: RealtimeSessionConfig, options?: RealtimeConnectOptions): Promise<import("@zhivex-ai/core/provider").CallbackRealtimeSession>;
     createBrowserToken(): Promise<never>;
 }
 //# sourceMappingURL=live.d.ts.map

package/dist/live.d.ts.map

@@ -1 +1 @@
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