msg91-webrtc-call/dist/type/call.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 type { IncomingCall, OutgoingCall } from "..";
import { User } from "./user";
export type Message = {
    type: "text";
    content: string;
} | {
    type: "image";
    content: string;
} | {
    type: "button";
    options: {
        title: string;
    }[];
};
export type MessageArray = Message[];
export declare enum CALL_EVENT {
    ENDED = "ended",
    ANSWERED = "answered",
    REJOINED = "rejoined",
    UNAVAILABLE = "unavailable",
    ERROR = "error",
    CONNECTED = "connected",
    MUTE = "mute",
    UNMUTE = "unmute",
    MESSAGE = "message",
    SILENCE_STATE = "silence"
}
export interface CallEventMap {
    [CALL_EVENT.ANSWERED]: (data: AnsweredPayload) => void;
    [CALL_EVENT.UNAVAILABLE]: (data: UnavailablePayload) => void;
    [CALL_EVENT.REJOINED]: (data: RejoinedPayload) => void;
    [CALL_EVENT.ENDED]: (data: EndedPayload) => void;
    [CALL_EVENT.ERROR]: (data: ErrorPayload) => void;
    [CALL_EVENT.CONNECTED]: (data: ConnectedPayload) => void;
    [CALL_EVENT.SILENCE_STATE]: (data: SilenceStatePayload) => void;
    [CALL_EVENT.MESSAGE]: (data: MessagePayload) => void;
    [CALL_EVENT.MUTE]: (data: MutePayload) => void;
    [CALL_EVENT.UNMUTE]: (data: UnmutePayload) => void;
}
type CallPayload = Omit<CallData, 'status' | 'type' | 'summary' | 'producerTransport' | 'consumerTransport' | 'routerRtpCapabilities' | 'token'>;
export type AnsweredPayload = CallPayload & {
    answeredBy: User;
    answeredAt: Date;
    message: 'Call answered';
    type: 'call-answered';
};
export type UnavailablePayload = CallPayload & {
    answeredBy: User;
    answeredAt: Date;
    message: 'Call already answered';
    type: 'call-unavailable';
};
export type RejoinedPayload = CallPayload & Pick<CallData, 'summary' | 'type' | 'status'>;
export type ErrorPayload = {
    message: string;
    error: any;
};
export type ConnectedPayload = typeof MediaStream;
export type EndedPayload = CallPayload & {
    endedAt: Date;
    endedBy: User;
    type: 'call-ended';
    message: 'Call ended';
};
export type SilenceStatePayload = {
    silent: boolean;
};
export type MessagePayload = {
    message: {
        type: "text";
        content: string;
    };
    from: "bot" | "user";
};
export type MutePayload = {
    uid: string;
};
export type UnmutePayload = {
    uid: string;
};
export declare enum WebRTC_EVENT {
    CALL = "call",
    INCOMING_CALL = "incoming-call",
    OUTGOING_CALL = "outgoing-call",
    PLAY_RINGTONE = "play-ringtone",
    STOP_RINGTONE = "stop-ringtone"
}
export interface WebRTCEventMap {
    [WebRTC_EVENT.CALL]: (data: CallData) => void;
    [WebRTC_EVENT.INCOMING_CALL]: (data: IncomingCall) => void;
    [WebRTC_EVENT.OUTGOING_CALL]: (data: OutgoingCall) => void;
    [WebRTC_EVENT.PLAY_RINGTONE]: () => void;
    [WebRTC_EVENT.STOP_RINGTONE]: () => void;
}
export declare enum CALL_STATUS {
    IDLE = "idle",
    RINGING = "ringing",
    CONNECTED = "connected",
    ENDED = "ended"
}
export declare enum CALL_TYPE {
    INCOMING = "incoming-call",
    OUTGOING = "outgoing-call"
}
export declare enum USER_STATUS {
    IDLE = "idle",// User is available to take new call
    BUSY = "busy"
}
export declare enum RINGTONE {
    STOP = "stop",
    RING = "ring"
}
export declare enum CALL_MANAGER_EVENT {
    RINGTONE_STATUS_CHANGED = "ringtone-status-changed"
}
export interface CallData {
    id: string;
    from: User;
    to: User[];
    bot?: User;
    summary?: {
        startedAt: Date;
        answeredAt: Date;
        answeredBy: User;
    };
    status?: CALL_STATUS;
    producerTransport: any;
    consumerTransport: any;
    routerRtpCapabilities: any;
    type: CALL_TYPE;
    token?: string;
}
export type CallInfo = {
    id: string;
    from: User;
    to: User[];
    type: CALL_TYPE;
    silent?: boolean;
};
export {};