@tixae-labs/web-sdk/dist/voice-class.d.ts

Javascript Web SDK for doing WebRTC AI Voice Calls with Convocore.

import type { InitCallOptionsType, InitWebRtcCall } from "../../../../src/app/Types/firebase";
import { TypedWebCall } from "./voice-only-events";
export declare const AppIceServers: RTCIceServer[];
export type InitCallEvent = InitWebRtcCall;
export default class WebCall extends TypedWebCall {
    private agentId;
    private region;
    private convoId;
    private sessionId;
    private options;
    private mediaStream;
    private remoteStream;
    private audioElement;
    private peerConnection;
    private webSocket;
    isMuted: boolean;
    status: "idle" | "connecting" | "connected";
    private audioContext;
    private analyser;
    private volumeAnimationFrame;
    private baseUrl;
    constructor(options?: InitCallOptionsType);
    /**
     * Initialize the API with the given agent ID.
     * @param agentId - Your unique agent ID (or any string)
     */
    init(input: Partial<InitWebRtcCall>): Promise<{
        agentId: string | null;
        region: string | null;
        convoId: string | null;
        sessionId: string | null;
        options: InitCallOptionsType;
    }>;
    /**
     * Start the WebRTC call flow:
     * 1) Open a WebSocket to your signaling server
     * 2) Create a RTCPeerConnection
     * 3) Get local microphone and attach it
     * 4) Handle remote track (attach to hidden <audio>)
     * 5) Exchange offers/answers/candidates
     * 6) Trigger events as needed
     */
    startCall(): Promise<void>;
    /**
     * Toggle the local microphone tracks (mute/unmute).
     */
    toggleMute(): void;
    /**
     * End the call:
     * 1) Close peer connection
     * 2) Close WS
     * 3) Stop local tracks
     * 4) Remove hidden <audio>
     * 5) Cancel animation frames
     */
    endCall(): void;
}