@openai/agents-realtime 0.0.1

package/dist/openaiRealtimeWebRtc.js

@@ -0,0 +1,245 @@
+/// <reference lib="dom" />
+import { isBrowserEnvironment } from '@openai/agents-core/_shims';
+import { UserError } from '@openai/agents-core';
+import logger from "./logger.js";
+import { OpenAIRealtimeBase, } from "./openaiRealtimeBase.js";
+import { parseRealtimeEvent } from "./openaiRealtimeEvents.js";
+import { HEADERS } from "./utils.js";
+/**
+ * Transport layer that's handling the connection between the client and OpenAI's Realtime API
+ * via WebRTC. While this transport layer is designed to be used within a RealtimeSession, it can
+ * also be used standalone if you want to have a direct connection to the Realtime API.
+ *
+ * Unless you specify a `mediaStream` or `audioElement` option, the transport layer will
+ * automatically configure the microphone and audio output to be used by the session.
+ */
+export class OpenAIRealtimeWebRTC extends OpenAIRealtimeBase {
+    options;
+    #url;
+    #state = {
+        status: 'disconnected',
+        peerConnection: undefined,
+        dataChannel: undefined,
+    };
+    #useInsecureApiKey;
+    #ongoingResponse = false;
+    #muted = false;
+    constructor(options = {}) {
+        if (typeof RTCPeerConnection === 'undefined') {
+            throw new Error('WebRTC is not supported in this environment');
+        }
+        super(options);
+        this.options = options;
+        this.#url = options.baseUrl ?? `https://api.openai.com/v1/realtime`;
+        this.#useInsecureApiKey = options.useInsecureApiKey ?? false;
+    }
+    /**
+     * The current status of the WebRTC connection.
+     */
+    get status() {
+        return this.#state.status;
+    }
+    /**
+     * The current connection state of the WebRTC connection including the peer connection and data
+     * channel.
+     */
+    get connectionState() {
+        return this.#state;
+    }
+    /**
+     * Whether the session is muted.
+     */
+    get muted() {
+        return this.#muted;
+    }
+    /**
+     * Connect to the Realtime API. This will establish the connection to the OpenAI Realtime API
+     * via WebRTC.
+     *
+     * If you are using a browser, the transport layer will also automatically configure the
+     * microphone and audio output to be used by the session.
+     *
+     * @param options - The options for the connection.
+     */
+    async connect(options) {
+        if (this.#state.status === 'connected') {
+            return;
+        }
+        if (this.#state.status === 'connecting') {
+            logger.warn('Realtime connection already in progress. Please await original promise');
+        }
+        const model = options.model ?? this.currentModel;
+        this.currentModel = model;
+        const baseUrl = options.url ?? this.#url;
+        const apiKey = await this._getApiKey(options);
+        const isClientKey = typeof apiKey === 'string' && apiKey.startsWith('ek_');
+        if (isBrowserEnvironment() && !this.#useInsecureApiKey && !isClientKey) {
+            throw new UserError('Using the WebRTC connection in a browser environment requires an insecure API key. Please use a WebSocket connection instead or set the useInsecureApiKey option to true.');
+        }
+        // eslint-disable-next-line no-async-promise-executor
+        return new Promise(async (resolve, reject) => {
+            try {
+                const userSessionConfig = {
+                    ...(options.initialSessionConfig || {}),
+                    model: this.currentModel,
+                };
+                const connectionUrl = new URL(baseUrl);
+                const peerConnection = new RTCPeerConnection();
+                const dataChannel = peerConnection.createDataChannel('oai-events');
+                this.#state = {
+                    status: 'connecting',
+                    peerConnection,
+                    dataChannel,
+                };
+                this.emit('connection_change', this.#state.status);
+                dataChannel.addEventListener('open', () => {
+                    this.#state = {
+                        status: 'connected',
+                        peerConnection,
+                        dataChannel,
+                    };
+                    // Sending the session config again here once the channel is connected to ensure
+                    // that the session config is sent to the server before the first response is received
+                    // Setting it on connection should work but the config is not being validated on the
+                    // server. This triggers a validation error if the config is not valid.
+                    this.updateSessionConfig(userSessionConfig);
+                    this.emit('connection_change', this.#state.status);
+                    this._onOpen();
+                    resolve();
+                });
+                dataChannel.addEventListener('error', (event) => {
+                    this.close();
+                    this._onError(event);
+                    reject(event);
+                });
+                dataChannel.addEventListener('message', (event) => {
+                    this._onMessage(event);
+                    const { data: parsed, isGeneric } = parseRealtimeEvent(event);
+                    if (!parsed || isGeneric) {
+                        return;
+                    }
+                    if (parsed.type === 'response.created') {
+                        this.#ongoingResponse = true;
+                    }
+                    else if (parsed.type === 'response.done') {
+                        this.#ongoingResponse = false;
+                    }
+                    if (parsed.type === 'session.created') {
+                        this._tracingConfig = parsed.session.tracing;
+                        // Trying to turn on tracing after the session is created
+                        this._updateTracingConfig(userSessionConfig.tracing ?? 'auto');
+                    }
+                });
+                // set up audio playback
+                const audioElement = this.options.audioElement ?? document.createElement('audio');
+                audioElement.autoplay = true;
+                peerConnection.ontrack = (event) => {
+                    audioElement.srcObject = event.streams[0];
+                };
+                // get microphone stream
+                const stream = this.options.mediaStream ??
+                    (await navigator.mediaDevices.getUserMedia({
+                        audio: true,
+                    }));
+                peerConnection.addTrack(stream.getAudioTracks()[0]);
+                const offer = await peerConnection.createOffer();
+                await peerConnection.setLocalDescription(offer);
+                if (!offer.sdp) {
+                    throw new Error('Failed to create offer');
+                }
+                const sessionConfig = {
+                    ...this._getMergedSessionConfig(userSessionConfig),
+                    model: this.currentModel,
+                };
+                const data = new FormData();
+                data.append('sdp', offer.sdp);
+                data.append('session', JSON.stringify(sessionConfig));
+                const sdpResponse = await fetch(connectionUrl, {
+                    method: 'POST',
+                    body: data,
+                    headers: {
+                        Authorization: `Bearer ${apiKey}`,
+                        'X-OpenAI-Agents-SDK': HEADERS['X-OpenAI-Agents-SDK'],
+                    },
+                });
+                const answer = {
+                    type: 'answer',
+                    sdp: await sdpResponse.text(),
+                };
+                await peerConnection.setRemoteDescription(answer);
+            }
+            catch (error) {
+                this.close();
+                this._onError(error);
+                reject(error);
+            }
+        });
+    }
+    /**
+     * Send an event to the Realtime API. This will stringify the event and send it directly to the
+     * API. This can be used if you want to take control over the connection and send events manually.
+     *
+     * @param event - The event to send.
+     */
+    sendEvent(event) {
+        if (!this.#state.dataChannel ||
+            this.#state.dataChannel.readyState !== 'open') {
+            throw new Error('WebRTC data channel is not connected. Make sure you call `connect()` before sending events.');
+        }
+        this.#state.dataChannel.send(JSON.stringify(event));
+    }
+    /**
+     * Mute or unmute the session.
+     * @param muted - Whether to mute the session.
+     */
+    mute(muted) {
+        this.#muted = muted;
+        if (this.#state.peerConnection) {
+            const peerConnection = this.#state.peerConnection;
+            peerConnection.getSenders().forEach((sender) => {
+                if (sender.track) {
+                    sender.track.enabled = !muted;
+                }
+            });
+        }
+    }
+    /**
+     * Close the connection to the Realtime API and disconnects the underlying WebRTC connection.
+     */
+    close() {
+        if (this.#state.dataChannel) {
+            this.#state.dataChannel.close();
+        }
+        if (this.#state.peerConnection) {
+            const peerConnection = this.#state.peerConnection;
+            peerConnection.getSenders().forEach((sender) => {
+                sender.track?.stop();
+            });
+            peerConnection.close();
+        }
+        if (this.#state.status !== 'disconnected') {
+            this.#state = {
+                status: 'disconnected',
+                peerConnection: undefined,
+                dataChannel: undefined,
+            };
+            this.emit('connection_change', this.#state.status);
+            this._onClose();
+        }
+    }
+    /**
+     * Interrupt the current response if one is ongoing and clear the audio buffer so that the agent
+     * stops talking.
+     */
+    interrupt() {
+        if (this.#ongoingResponse) {
+            this.sendEvent({
+                type: 'response.cancel',
+            });
+        }
+        this.sendEvent({
+            type: 'output_audio_buffer.clear',
+        });
+    }
+}
+//# sourceMappingURL=openaiRealtimeWebRtc.js.map

package/dist/openaiRealtimeWebRtc.js.map

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package/dist/openaiRealtimeWebRtc.mjs

@@ -0,0 +1,245 @@
+/// <reference lib="dom" />
+import { isBrowserEnvironment } from '@openai/agents-core/_shims';
+import { UserError } from '@openai/agents-core';
+import logger from "./logger.mjs";
+import { OpenAIRealtimeBase, } from "./openaiRealtimeBase.mjs";
+import { parseRealtimeEvent } from "./openaiRealtimeEvents.mjs";
+import { HEADERS } from "./utils.mjs";
+/**
+ * Transport layer that's handling the connection between the client and OpenAI's Realtime API
+ * via WebRTC. While this transport layer is designed to be used within a RealtimeSession, it can
+ * also be used standalone if you want to have a direct connection to the Realtime API.
+ *
+ * Unless you specify a `mediaStream` or `audioElement` option, the transport layer will
+ * automatically configure the microphone and audio output to be used by the session.
+ */
+export class OpenAIRealtimeWebRTC extends OpenAIRealtimeBase {
+    options;
+    #url;
+    #state = {
+        status: 'disconnected',
+        peerConnection: undefined,
+        dataChannel: undefined,
+    };
+    #useInsecureApiKey;
+    #ongoingResponse = false;
+    #muted = false;
+    constructor(options = {}) {
+        if (typeof RTCPeerConnection === 'undefined') {
+            throw new Error('WebRTC is not supported in this environment');
+        }
+        super(options);
+        this.options = options;
+        this.#url = options.baseUrl ?? `https://api.openai.com/v1/realtime`;
+        this.#useInsecureApiKey = options.useInsecureApiKey ?? false;
+    }
+    /**
+     * The current status of the WebRTC connection.
+     */
+    get status() {
+        return this.#state.status;
+    }
+    /**
+     * The current connection state of the WebRTC connection including the peer connection and data
+     * channel.
+     */
+    get connectionState() {
+        return this.#state;
+    }
+    /**
+     * Whether the session is muted.
+     */
+    get muted() {
+        return this.#muted;
+    }
+    /**
+     * Connect to the Realtime API. This will establish the connection to the OpenAI Realtime API
+     * via WebRTC.
+     *
+     * If you are using a browser, the transport layer will also automatically configure the
+     * microphone and audio output to be used by the session.
+     *
+     * @param options - The options for the connection.
+     */
+    async connect(options) {
+        if (this.#state.status === 'connected') {
+            return;
+        }
+        if (this.#state.status === 'connecting') {
+            logger.warn('Realtime connection already in progress. Please await original promise');
+        }
+        const model = options.model ?? this.currentModel;
+        this.currentModel = model;
+        const baseUrl = options.url ?? this.#url;
+        const apiKey = await this._getApiKey(options);
+        const isClientKey = typeof apiKey === 'string' && apiKey.startsWith('ek_');
+        if (isBrowserEnvironment() && !this.#useInsecureApiKey && !isClientKey) {
+            throw new UserError('Using the WebRTC connection in a browser environment requires an insecure API key. Please use a WebSocket connection instead or set the useInsecureApiKey option to true.');
+        }
+        // eslint-disable-next-line no-async-promise-executor
+        return new Promise(async (resolve, reject) => {
+            try {
+                const userSessionConfig = {
+                    ...(options.initialSessionConfig || {}),
+                    model: this.currentModel,
+                };
+                const connectionUrl = new URL(baseUrl);
+                const peerConnection = new RTCPeerConnection();
+                const dataChannel = peerConnection.createDataChannel('oai-events');
+                this.#state = {
+                    status: 'connecting',
+                    peerConnection,
+                    dataChannel,
+                };
+                this.emit('connection_change', this.#state.status);
+                dataChannel.addEventListener('open', () => {
+                    this.#state = {
+                        status: 'connected',
+                        peerConnection,
+                        dataChannel,
+                    };
+                    // Sending the session config again here once the channel is connected to ensure
+                    // that the session config is sent to the server before the first response is received
+                    // Setting it on connection should work but the config is not being validated on the
+                    // server. This triggers a validation error if the config is not valid.
+                    this.updateSessionConfig(userSessionConfig);
+                    this.emit('connection_change', this.#state.status);
+                    this._onOpen();
+                    resolve();
+                });
+                dataChannel.addEventListener('error', (event) => {
+                    this.close();
+                    this._onError(event);
+                    reject(event);
+                });
+                dataChannel.addEventListener('message', (event) => {
+                    this._onMessage(event);
+                    const { data: parsed, isGeneric } = parseRealtimeEvent(event);
+                    if (!parsed || isGeneric) {
+                        return;
+                    }
+                    if (parsed.type === 'response.created') {
+                        this.#ongoingResponse = true;
+                    }
+                    else if (parsed.type === 'response.done') {
+                        this.#ongoingResponse = false;
+                    }
+                    if (parsed.type === 'session.created') {
+                        this._tracingConfig = parsed.session.tracing;
+                        // Trying to turn on tracing after the session is created
+                        this._updateTracingConfig(userSessionConfig.tracing ?? 'auto');
+                    }
+                });
+                // set up audio playback
+                const audioElement = this.options.audioElement ?? document.createElement('audio');
+                audioElement.autoplay = true;
+                peerConnection.ontrack = (event) => {
+                    audioElement.srcObject = event.streams[0];
+                };
+                // get microphone stream
+                const stream = this.options.mediaStream ??
+                    (await navigator.mediaDevices.getUserMedia({
+                        audio: true,
+                    }));
+                peerConnection.addTrack(stream.getAudioTracks()[0]);
+                const offer = await peerConnection.createOffer();
+                await peerConnection.setLocalDescription(offer);
+                if (!offer.sdp) {
+                    throw new Error('Failed to create offer');
+                }
+                const sessionConfig = {
+                    ...this._getMergedSessionConfig(userSessionConfig),
+                    model: this.currentModel,
+                };
+                const data = new FormData();
+                data.append('sdp', offer.sdp);
+                data.append('session', JSON.stringify(sessionConfig));
+                const sdpResponse = await fetch(connectionUrl, {
+                    method: 'POST',
+                    body: data,
+                    headers: {
+                        Authorization: `Bearer ${apiKey}`,
+                        'X-OpenAI-Agents-SDK': HEADERS['X-OpenAI-Agents-SDK'],
+                    },
+                });
+                const answer = {
+                    type: 'answer',
+                    sdp: await sdpResponse.text(),
+                };
+                await peerConnection.setRemoteDescription(answer);
+            }
+            catch (error) {
+                this.close();
+                this._onError(error);
+                reject(error);
+            }
+        });
+    }
+    /**
+     * Send an event to the Realtime API. This will stringify the event and send it directly to the
+     * API. This can be used if you want to take control over the connection and send events manually.
+     *
+     * @param event - The event to send.
+     */
+    sendEvent(event) {
+        if (!this.#state.dataChannel ||
+            this.#state.dataChannel.readyState !== 'open') {
+            throw new Error('WebRTC data channel is not connected. Make sure you call `connect()` before sending events.');
+        }
+        this.#state.dataChannel.send(JSON.stringify(event));
+    }
+    /**
+     * Mute or unmute the session.
+     * @param muted - Whether to mute the session.
+     */
+    mute(muted) {
+        this.#muted = muted;
+        if (this.#state.peerConnection) {
+            const peerConnection = this.#state.peerConnection;
+            peerConnection.getSenders().forEach((sender) => {
+                if (sender.track) {
+                    sender.track.enabled = !muted;
+                }
+            });
+        }
+    }
+    /**
+     * Close the connection to the Realtime API and disconnects the underlying WebRTC connection.
+     */
+    close() {
+        if (this.#state.dataChannel) {
+            this.#state.dataChannel.close();
+        }
+        if (this.#state.peerConnection) {
+            const peerConnection = this.#state.peerConnection;
+            peerConnection.getSenders().forEach((sender) => {
+                sender.track?.stop();
+            });
+            peerConnection.close();
+        }
+        if (this.#state.status !== 'disconnected') {
+            this.#state = {
+                status: 'disconnected',
+                peerConnection: undefined,
+                dataChannel: undefined,
+            };
+            this.emit('connection_change', this.#state.status);
+            this._onClose();
+        }
+    }
+    /**
+     * Interrupt the current response if one is ongoing and clear the audio buffer so that the agent
+     * stops talking.
+     */
+    interrupt() {
+        if (this.#ongoingResponse) {
+            this.sendEvent({
+                type: 'response.cancel',
+            });
+        }
+        this.sendEvent({
+            type: 'output_audio_buffer.clear',
+        });
+    }
+}
+//# sourceMappingURL=openaiRealtimeWebRtc.mjs.map

package/dist/openaiRealtimeWebRtc.mjs.map

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package/dist/openaiRealtimeWebsocket.d.ts

@@ -0,0 +1,126 @@
+import { WebSocket } from '@openai/agents-realtime/_shims';
+import { RealtimeTransportLayerConnectOptions, RealtimeTransportLayer } from './transportLayer';
+import { RealtimeClientMessage } from './clientMessages';
+import { OpenAIRealtimeBase, OpenAIRealtimeBaseOptions } from './openaiRealtimeBase';
+import { TransportLayerAudio } from './transportLayerEvents';
+/**
+ * The connection state of the WebSocket connection.
+ */
+export type WebSocketState = {
+    status: 'disconnected';
+    websocket: undefined;
+} | {
+    status: 'connecting';
+    websocket: WebSocket;
+} | {
+    status: 'connected';
+    websocket: WebSocket;
+};
+/**
+ * The options for the OpenAI Realtime WebSocket transport layer.
+ */
+export type OpenAIRealtimeWebSocketOptions = {
+    /**
+     * **Important**: Do not use this option unless you know what you are doing.
+     *
+     * Whether to use an insecure API key. This has to be set if you are trying to use a regular
+     * OpenAI API key instead of a client ephemeral key.
+     * @see https://platform.openai.com/docs/guides/realtime#creating-an-ephemeral-token
+     */
+    useInsecureApiKey?: boolean;
+} & OpenAIRealtimeBaseOptions;
+/**
+ * Transport layer that's handling the connection between the client and OpenAI's Realtime API
+ * via WebSockets. While this transport layer is designed to be used within a RealtimeSession, it
+ * can also be used standalone if you want to have a direct connection to the Realtime API.
+ */
+export declare class OpenAIRealtimeWebSocket extends OpenAIRealtimeBase implements RealtimeTransportLayer {
+    #private;
+    /**
+     * Timestamp maintained by the transport layer to aid with the calculation of the elapsed time
+     * since the response started to compute the right interruption time.
+     *
+     * Mostly internal but might be used by extended transport layers for their interruption
+     * calculation.
+     */
+    protected _firstAudioTimestamp: number | undefined;
+    protected _audioLengthMs: number;
+    constructor(options?: OpenAIRealtimeWebSocketOptions);
+    /**
+     * The current status of the WebSocket connection.
+     */
+    get status(): "connecting" | "connected" | "disconnected";
+    /**
+     * The current connection state of the WebSocket connection.
+     */
+    get connectionState(): WebSocketState;
+    /**
+     * Always returns `null` as the WebSocket transport layer does not handle muting. Instead,
+     * this should be handled by the client by not triggering the `sendAudio` method.
+     */
+    get muted(): null;
+    /**
+     * The current item ID of the ongoing response.
+     */
+    protected get currentItemId(): string | undefined;
+    /**
+     * Triggers the `audio` event that a client might listen to to receive the audio buffer.
+     * Protected for you to be able to override and disable emitting this event in case your extended
+     * transport layer handles audio internally.
+     *
+     * @param audioEvent - The audio event to emit.
+     */
+    protected _onAudio(audioEvent: TransportLayerAudio): void;
+    connect(options: RealtimeTransportLayerConnectOptions): Promise<void>;
+    /**
+     * Send an event to the Realtime API. This will stringify the event and send it directly to the
+     * API. This can be used if you want to take control over the connection and send events manually.
+     *
+     * @param event - The event to send.
+     */
+    sendEvent(event: RealtimeClientMessage): void;
+    /**
+     * Close the WebSocket connection.
+     *
+     * This will also reset any internal connection tracking used for interruption handling.
+     */
+    close(): void;
+    /**
+     * Will throw an error as the WebSocket transport layer does not support muting.
+     */
+    mute(_muted: boolean): never;
+    /**
+     * Send an audio buffer to the Realtime API. This is used for your client to send audio to the
+     * model to respond.
+     *
+     * @param audio - The audio buffer to send.
+     * @param options - The options for the audio buffer.
+     */
+    sendAudio(audio: ArrayBuffer, options?: {
+        commit?: boolean;
+    }): void;
+    /**
+     * Send a cancel response event to the Realtime API. This is used to cancel an ongoing
+     *  response that the model is currently generating.
+     */
+    _cancelResponse(): void;
+    /**
+     * Do NOT call this method directly. Call `interrupt()` instead for proper interruption handling.
+     *
+     * This method is used to send the right events to the API to inform the model that the user has
+     * interrupted the response. It might be overridden/extended by an extended transport layer. See
+     * the `TwilioRealtimeTransportLayer` for an example.
+     *
+     * @param elapsedTime - The elapsed time since the response started.
+     */
+    _interrupt(elapsedTime: number): void;
+    /**
+     * Interrupt the ongoing response. This method is triggered automatically by the client when
+     * voice activity detection (VAD) is enabled (default) as well as when an output guardrail got
+     * triggered.
+     *
+     * You can also call this method directly if you want to interrupt the conversation for example
+     * based on an event in the client.
+     */
+    interrupt(): void;
+}