msg91-webrtc-call/dist/index.d.ts

**msg91-webrtc-call** is a lightweight JavaScript SDK that enables you to easily add peer-to-peer WebRTC audio/video calling functionality to your web applications using the MSG91 infrastructure.

import { User } from "./type/user";
import { CALL_EVENT, WebRTC_EVENT, MessageArray, CALL_STATUS, CallData, CallInfo, AnsweredPayload, UnavailablePayload, RejoinedPayload, EndedPayload, MessagePayload, SilenceStatePayload, ConnectedPayload, UnmutePayload, MutePayload, ErrorPayload, WebRTCEventMap, CallEventMap } from "./type/call";
import { EventEmitter } from "./util/event-emitter";
declare class WebRTC extends EventEmitter {
    private user;
    private callId;
    private status;
    private userToken;
    private callToken?;
    private callManager;
    constructor(userToken: string);
    on<K extends WebRTC_EVENT>(event: K, listener: WebRTCEventMap[K]): {
        remove: () => void;
    };
    close(): void;
    call(callToken: string): Promise<void>;
    rejoinCall(callId: string): Promise<void>;
    sendUserContext(data: {
        context?: any;
        message?: any;
    }): Promise<void>;
    /**
     *
     * @param status Set global silence for incoming calls
     */
    silence(status?: boolean): void;
}
declare class Call extends EventEmitter {
    protected user: User;
    protected id: string;
    protected existingCall: boolean;
    private from;
    private to;
    private producerTransport;
    private consumerTransport;
    private producer;
    private routerRtpCapabilities;
    private type;
    private device;
    protected userToken: string;
    private status;
    private mediaStream;
    private audioTrack;
    private rtlayerEvents;
    constructor(user: User, userToken: string, data: CallData);
    on<K extends CALL_EVENT>(event: K, listener: CallEventMap[K]): {
        remove: () => void;
    };
    protected addEvent(callback: Function): void;
    protected setStatus(status: CALL_STATUS): Promise<void>;
    getStatus(): CALL_STATUS;
    getMediaStream(): MediaStream;
    protected isExistingCall(): boolean;
    mute(): void;
    unmute(): void;
    hang(data?: any): void;
    getInfo(): CallInfo;
    protected setupCall(): Promise<void>;
}
declare class IncomingCall extends Call {
    private isUnavailable;
    private isSilent;
    constructor(user: User, userToken: string, data: CallData);
    private rejoin;
    accept(): void;
    silence(status?: boolean): void;
    getInfo(): CallInfo;
    reject(): void;
}
declare class OutgoingCall extends Call {
    constructor(user: User, userToken: string, data: CallData);
    private rejoin;
    cancel(): void;
    sendMessage(message: MessageArray, isContext: boolean): Promise<void>;
}
declare const _default: (userToken: string, env?: "dev" | "test" | "prod") => WebRTC | undefined;
export default _default;
export { CALL_EVENT, WebRTC_EVENT };
export type { IncomingCall, OutgoingCall };
export type { AnsweredPayload, UnavailablePayload, ConnectedPayload, RejoinedPayload, EndedPayload, MessagePayload, SilenceStatePayload, MutePayload, UnmutePayload, ErrorPayload };