@guava-ai/guava-sdk 0.37.0 → 0.39.0

package/dist/src/webrtc/index.d.ts

@@ -0,0 +1,74 @@
+import { type ConnectingTone, type Tone } from "./tones.ts";
+export { type ConnectingTone, JingleTone, RingbackTone, LeavingTone, type Tone } from "./tones.ts";
+export type CallState = "not-started" | "starting" | "waiting-for-permission" | "connecting" | "connected" | "ended";
+export type CallErrorCode = "MICROPHONE_ACCESS_FAILED" | "CONNECTION_FAILED" | "AUTHENTICATION_FAILED" | "PROTOCOL_ERROR" | "SERVER_ERROR" | "REMOTE_HANGUP" | "CONNECTION_TIMEOUT";
+export declare class CallError extends Error {
+    readonly code: CallErrorCode;
+    constructor(code: CallErrorCode, message: string, options?: ErrorOptions);
+}
+export declare class Call {
+    private _state;
+    private _error?;
+    private readonly _stateHandlers;
+    private readonly _remoteStreamHandlers;
+    private readonly _messageHandlers;
+    private readonly _webrtcCode;
+    private readonly _sockUrl;
+    private readonly _remoteAudio;
+    private readonly _connectingTone?;
+    private readonly _leavingTone?;
+    private readonly _connectionTimeoutMs?;
+    private readonly _customPayload;
+    private _connectionTimer?;
+    private _sock?;
+    private readonly _chatSockUrl;
+    private readonly _chatSessionId;
+    private _chatSock?;
+    private _chatReady;
+    private _pendingChatMessages;
+    private _localStream?;
+    private _peerConnection?;
+    private _remoteStream?;
+    private _iceServers?;
+    private _pendingStart?;
+    constructor({ webrtcCode, baseUrl, connectingTone, leavingTone, connectionTimeoutMs, customPayload, }: {
+        /** Code used to authenticate with the signaling server. */
+        webrtcCode: string;
+        /** Base URL of the Guava server. Defaults to `https://app.goguava.ai/`. */
+        baseUrl?: string;
+        /** Played while the call is in the "connecting" state. Defaults to {@link JingleTone}. */
+        connectingTone?: ConnectingTone | "disabled";
+        /** Played when the call ends. Defaults to {@link LeavingTone}. */
+        leavingTone?: Tone | "disabled";
+        /** Max time in the "connecting" state before the call fails with CONNECTION_TIMEOUT. Defaults to 30 seconds. */
+        connectionTimeoutMs?: number | "disabled";
+        /** A custom payload that is forwarded to the agent at the start of the call. */
+        customPayload?: Record<string, unknown>;
+    });
+    get state(): CallState;
+    get error(): CallError | undefined;
+    get remoteStream(): MediaStream | undefined;
+    onStateChange(handler: (state: CallState) => void): () => void;
+    onRemoteStream(handler: (stream: MediaStream) => void): () => void;
+    /**
+     * Registers a handler for messages the agent sends with `call.send_webrtc_message()`.
+     * @param handler - Called with each message's payload.
+     * @returns A function that unregisters the handler.
+     */
+    onMessage(handler: (payload: Record<string, unknown>) => void): () => void;
+    private _setState;
+    start(): Promise<void>;
+    /**
+     * Sends a message to the agent, delivered to its `on_webrtc_message` handler.
+     * Messages sent before the message channel is ready are queued.
+     * @param payload - A JSON-serializable object.
+     */
+    sendMessage(payload: Record<string, unknown>): void;
+    private _openChatSocket;
+    hangup(): Promise<void>;
+    private _send;
+    private _onMessage;
+    private _connectPeer;
+    private _end;
+    private _teardown;
+}

package/dist/src/webrtc/index.js

@@ -0,0 +1,398 @@
+"use strict";
+Object.defineProperty(exports, "__esModule", { value: true });
+exports.Call = exports.CallError = exports.LeavingTone = exports.RingbackTone = exports.JingleTone = void 0;
+const tones_ts_1 = require("./tones.js");
+var tones_ts_2 = require("./tones.js");
+Object.defineProperty(exports, "JingleTone", { enumerable: true, get: function () { return tones_ts_2.JingleTone; } });
+Object.defineProperty(exports, "RingbackTone", { enumerable: true, get: function () { return tones_ts_2.RingbackTone; } });
+Object.defineProperty(exports, "LeavingTone", { enumerable: true, get: function () { return tones_ts_2.LeavingTone; } });
+class CallError extends Error {
+    code;
+    constructor(code, message, options) {
+        super(message, options);
+        this.name = "CallError";
+        this.code = code;
+    }
+}
+exports.CallError = CallError;
+// Used if the server doesn't send ICE servers with the auth reply.
+const DEFAULT_ICE_SERVERS = [
+    { urls: ["stun:stun.l.google.com:19302", "stun:stun.sipgate.net:10000"] },
+];
+const SOCKET_PROTOCOLS = {
+    "http:": "ws:",
+    "https:": "wss:",
+    "ws:": "ws:",
+    "wss:": "wss:",
+};
+class Call {
+    _state = "not-started";
+    _error;
+    _stateHandlers = new Set();
+    _remoteStreamHandlers = new Set();
+    _messageHandlers = new Set();
+    _webrtcCode;
+    _sockUrl;
+    _remoteAudio;
+    _connectingTone;
+    _leavingTone;
+    _connectionTimeoutMs;
+    _customPayload;
+    _connectionTimer;
+    _sock;
+    _chatSockUrl;
+    // Links the chat socket to this call's session on the server.
+    _chatSessionId = crypto.randomUUID();
+    _chatSock;
+    _chatReady = false;
+    // Messages sent before the chat socket is ready.
+    _pendingChatMessages = [];
+    _localStream;
+    _peerConnection;
+    _remoteStream;
+    _iceServers;
+    // Settles the promise returned by start().
+    _pendingStart;
+    constructor({ webrtcCode, baseUrl = "https://app.goguava.ai/", connectingTone = new tones_ts_1.JingleTone(), leavingTone = new tones_ts_1.LeavingTone(), connectionTimeoutMs = 30_000, customPayload = {}, }) {
+        this._webrtcCode = webrtcCode;
+        this._customPayload = customPayload;
+        if (connectingTone !== "disabled") {
+            this._connectingTone = connectingTone;
+        }
+        if (leavingTone !== "disabled") {
+            this._leavingTone = leavingTone;
+        }
+        if (connectionTimeoutMs !== "disabled") {
+            this._connectionTimeoutMs = connectionTimeoutMs;
+        }
+        this._sockUrl = new URL("/webrtc/", baseUrl);
+        const sockProtocol = SOCKET_PROTOCOLS[this._sockUrl.protocol];
+        if (sockProtocol === undefined) {
+            throw new Error(`Unsupported baseUrl protocol: ${this._sockUrl.protocol}`);
+        }
+        this._sockUrl.protocol = sockProtocol;
+        this._chatSockUrl = new URL("/webrtc-chat/", this._sockUrl);
+        this._remoteAudio = document.createElement("audio");
+        this._remoteAudio.autoplay = true;
+    }
+    get state() {
+        return this._state;
+    }
+    // Set if the call ended due to a failure.
+    get error() {
+        return this._error;
+    }
+    // Set once the remote audio track arrives, which happens before the call reaches "connected".
+    get remoteStream() {
+        return this._remoteStream;
+    }
+    // Returns a function that unregisters the handler.
+    onStateChange(handler) {
+        this._stateHandlers.add(handler);
+        return () => {
+            this._stateHandlers.delete(handler);
+        };
+    }
+    // Fires whenever the remote stream is replaced, including the first time it arrives.
+    // Returns a function that unregisters the handler.
+    onRemoteStream(handler) {
+        this._remoteStreamHandlers.add(handler);
+        return () => {
+            this._remoteStreamHandlers.delete(handler);
+        };
+    }
+    /**
+     * Registers a handler for messages the agent sends with `call.send_webrtc_message()`.
+     * @param handler - Called with each message's payload.
+     * @returns A function that unregisters the handler.
+     */
+    onMessage(handler) {
+        this._messageHandlers.add(handler);
+        return () => {
+            this._messageHandlers.delete(handler);
+        };
+    }
+    _setState(state) {
+        if (state === this._state) {
+            return;
+        }
+        if (state === "ended") {
+            this._leavingTone?.play();
+        }
+        this._state = state;
+        if (state === "connecting") {
+            this._connectingTone?.start();
+            if (this._connectionTimeoutMs !== undefined) {
+                this._connectionTimer = setTimeout(() => {
+                    this._send({ type: "hangup" });
+                    this._end(new CallError("CONNECTION_TIMEOUT", "Timed out connecting the call."));
+                }, this._connectionTimeoutMs);
+            }
+        }
+        else {
+            this._connectingTone?.stop();
+            clearTimeout(this._connectionTimer);
+            this._connectionTimer = undefined;
+        }
+        for (const handler of this._stateHandlers) {
+            handler(state);
+        }
+    }
+    async start() {
+        if (this._state !== "not-started") {
+            throw new Error(`start() was already called.`);
+        }
+        this._setState("starting");
+        // Add the audio element.
+        document.body.appendChild(this._remoteAudio);
+        // We start by requesting device permissions first. The idea is that, after permission has been granted,
+        // it makes sence to play a ringback, but it doesn't make sense to do so during the permission prompt.
+        const permissionTimer = setTimeout(() => {
+            this._setState("waiting-for-permission");
+        }, 500);
+        try {
+            this._localStream = await navigator.mediaDevices.getUserMedia({ audio: true, video: false });
+        }
+        catch (error) {
+            const callError = new CallError("MICROPHONE_ACCESS_FAILED", "Failed to access microphone.", {
+                cause: error,
+            });
+            this._end(callError);
+            throw callError;
+        }
+        finally {
+            clearTimeout(permissionTimer);
+        }
+        // hangup() was called during the permission prompt.
+        if (this.state === "ended") {
+            this._teardown();
+            return;
+        }
+        // Now open and authenticate the signaling socket.
+        this._setState("connecting"); // UI can start playing rinback tones here.
+        const started = new Promise((resolve, reject) => {
+            this._pendingStart = { resolve, reject };
+        });
+        this._sock = new WebSocket(this._sockUrl);
+        this._sock.onopen = () => {
+            this._send({
+                type: "auth",
+                body: JSON.stringify({
+                    webrtc_code: this._webrtcCode,
+                    chat_session_id: this._chatSessionId,
+                }),
+            });
+        };
+        this._sock.onmessage = (event) => {
+            this._onMessage(JSON.parse(event.data));
+        };
+        this._sock.onclose = () => {
+            this._end(new CallError("CONNECTION_FAILED", "Signaling connection closed."));
+        };
+        await started;
+    }
+    /**
+     * Sends a message to the agent, delivered to its `on_webrtc_message` handler.
+     * Messages sent before the message channel is ready are queued.
+     * @param payload - A JSON-serializable object.
+     */
+    sendMessage(payload) {
+        if (this._state === "not-started" || this._state === "ended") {
+            throw new Error(`sendMessage() called while the call is ${this._state}.`);
+        }
+        const message = JSON.stringify({ type: "webrtc-message", payload });
+        if (this._chatReady && this._chatSock !== undefined) {
+            this._chatSock.send(message);
+        }
+        else {
+            this._pendingChatMessages.push(message);
+        }
+    }
+    _openChatSocket() {
+        this._chatSock = new WebSocket(this._chatSockUrl);
+        this._chatSock.onopen = () => {
+            this._chatSock?.send(JSON.stringify({ webrtc_code: this._webrtcCode, chat_session_id: this._chatSessionId }));
+        };
+        this._chatSock.onmessage = (event) => {
+            const message = JSON.parse(event.data);
+            // The server sends "ready" once the call session is found.
+            if (message.type === "ready" && !this._chatReady) {
+                this._chatReady = true;
+                for (const pending of this._pendingChatMessages) {
+                    this._chatSock?.send(pending);
+                }
+                this._pendingChatMessages = [];
+            }
+            else if (message.command_type === "send-webrtc-message") {
+                for (const handler of this._messageHandlers) {
+                    handler(message.payload ?? {});
+                }
+            }
+        };
+    }
+    async hangup() {
+        if (this._state === "not-started") {
+            throw new Error(`hangup() called before start().`);
+        }
+        if (this._state === "ended") {
+            return;
+        }
+        this._send({ type: "hangup" });
+        this._end();
+    }
+    _send(message) {
+        if (this._sock !== undefined && this._sock.readyState === WebSocket.OPEN) {
+            this._sock.send(JSON.stringify(message));
+        }
+    }
+    _onMessage(message) {
+        switch (message.type) {
+            case "auth":
+                if (message.body !== "OK") {
+                    this._end(new CallError("AUTHENTICATION_FAILED", message.reason || "Authentication failed."));
+                    return;
+                }
+                this._iceServers = message.ice_servers;
+                this._openChatSocket();
+                this._send({ type: "connect", body: { custom_payload: this._customPayload } });
+                void this._connectPeer();
+                break;
+            case "desc":
+                if (!this._peerConnection) {
+                    this._end(new CallError("PROTOCOL_ERROR", "Received desc before the peer connection was created."));
+                    return;
+                }
+                this._peerConnection.setRemoteDescription(message.body).catch((error) => {
+                    this._end(new CallError("CONNECTION_FAILED", "Failed to set remote description.", {
+                        cause: error,
+                    }));
+                });
+                break;
+            case "candidate":
+                if (!this._peerConnection) {
+                    this._end(new CallError("PROTOCOL_ERROR", "Received candidate before the peer connection was created."));
+                    return;
+                }
+                this._peerConnection.addIceCandidate(message.body).catch((error) => {
+                    console.warn("Failed to add ICE candidate:", error);
+                });
+                break;
+            case "hangup":
+                if (this._state === "connected") {
+                    this._end();
+                }
+                else {
+                    this._end(new CallError("REMOTE_HANGUP", "The remote hung up before the call connected."));
+                }
+                break;
+            case "error":
+                this._end(new CallError("SERVER_ERROR", message.body || "Unknown server error."));
+                break;
+            case "transfer":
+                this._end(new CallError("PROTOCOL_ERROR", "Transfers are not supported."));
+                break;
+            default:
+                console.warn("Unknown signaling message:", message);
+        }
+    }
+    async _connectPeer() {
+        if (!this._localStream) {
+            return;
+        }
+        this._peerConnection = new RTCPeerConnection({
+            iceServers: this._iceServers ?? DEFAULT_ICE_SERVERS,
+        });
+        this._peerConnection.onicecandidate = (event) => {
+            if (event.candidate) {
+                this._send({ type: "candidate", body: event.candidate });
+            }
+        };
+        this._peerConnection.ontrack = (event) => {
+            this._remoteStream = new MediaStream([event.track]);
+            this._remoteAudio.srcObject = this._remoteStream;
+            for (const handler of this._remoteStreamHandlers) {
+                handler(this._remoteStream);
+            }
+        };
+        this._peerConnection.onconnectionstatechange = () => {
+            if (this._peerConnection === undefined) {
+                return;
+            }
+            // "disconnected" is often transient, so wait for "failed" before ending the call.
+            if (this._peerConnection.connectionState === "connected") {
+                this._setState("connected");
+                if (this._pendingStart !== undefined) {
+                    this._pendingStart.resolve();
+                    this._pendingStart = undefined;
+                }
+            }
+            else if (this._peerConnection.connectionState === "failed") {
+                this._end(new CallError("CONNECTION_FAILED", "Peer connection failed."));
+            }
+        };
+        for (const track of this._localStream.getAudioTracks()) {
+            this._peerConnection.addTrack(track);
+        }
+        try {
+            await this._peerConnection.setLocalDescription();
+        }
+        catch (error) {
+            this._end(new CallError("CONNECTION_FAILED", "Failed to create offer.", { cause: error }));
+            return;
+        }
+        // The call may have ended while creating the offer.
+        if (this.state === "ended") {
+            return;
+        }
+        this._send({ type: "desc", body: this._peerConnection.localDescription });
+    }
+    _end(error) {
+        if (this._state === "ended") {
+            return;
+        }
+        this._error = error;
+        this._teardown();
+        this._setState("ended");
+        // If start() is still pending, it rejects on failure and resolves on hangup().
+        if (this._pendingStart !== undefined) {
+            if (error) {
+                this._pendingStart.reject(error);
+            }
+            else {
+                this._pendingStart.resolve();
+            }
+            this._pendingStart = undefined;
+        }
+    }
+    _teardown() {
+        // Close the signaling socket.
+        if (this._sock) {
+            // Don't fire close handlers for a socket we closed ourselves.
+            this._sock.onclose = null;
+            this._sock.close();
+            this._sock = undefined;
+        }
+        // Close the chat socket.
+        if (this._chatSock) {
+            this._chatSock.close();
+            this._chatSock = undefined;
+        }
+        this._chatReady = false;
+        this._pendingChatMessages = [];
+        // Close the peer connection.
+        if (this._peerConnection !== undefined) {
+            this._peerConnection.close();
+            this._peerConnection = undefined;
+        }
+        for (const track of this._localStream?.getTracks() ?? []) {
+            track.stop();
+        }
+        this._localStream = undefined;
+        this._remoteStream = undefined;
+        // Remove the audio element.
+        this._remoteAudio.srcObject = null;
+        this._remoteAudio.remove();
+    }
+}
+exports.Call = Call;
+//# sourceMappingURL=index.js.map

package/dist/src/webrtc/index.js.map

@@ -0,0 +1 @@
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package/dist/src/webrtc/tones.d.ts

@@ -0,0 +1,38 @@
+export interface ConnectingTone {
+    start(): void;
+    stop(): void;
+}
+export declare class RingbackTone implements ConnectingTone {
+    private context;
+    private intervalId;
+    private readonly volume;
+    private readonly onSeconds;
+    private readonly periodSeconds;
+    private readonly startDelay;
+    start(): void;
+    stop(): void;
+}
+export declare class JingleTone implements ConnectingTone {
+    private context;
+    private masterGain;
+    private intervalId;
+    private running;
+    private pulseCount;
+    private nextBarStart;
+    private readonly volume;
+    private readonly bpm;
+    private readonly beatsPerBar;
+    private readonly startDelay;
+    start(): void;
+    private noteToFrequency;
+    stop(): void;
+    private playPulse;
+    private playNote;
+}
+export interface Tone {
+    play(): void;
+}
+export declare class LeavingTone implements Tone {
+    play(): void;
+    private playNote;
+}