import type { Message } from 'sip.js/lib/api/message';
import { OutgoingInviteRequest } from 'sip.js/lib/core';
import { Inviter } from 'sip.js/lib/api';
import CallSession from '../CallSession';
import type { Phone, AvailablePhoneOptions } from './Phone';
import WebRTCClient from '../../web-rtc-client';
import Emitter from '../../utils/Emitter';
import { WazoSession } from '../types';
export declare const ON_USER_AGENT = "onUserAgent";
export declare const ON_REGISTERED = "onRegistered";
export declare const ON_UNREGISTERED = "onUnRegistered";
export declare const ON_PROGRESS = "onProgress";
export declare const ON_CALL_ACCEPTED = "onCallAccepted";
export declare const ON_CALL_ANSWERED = "onCallAnswered";
export declare const ON_CALL_INCOMING = "onCallIncoming";
export declare const ON_CALL_OUTGOING = "onCallOutgoing";
export declare const ON_CALL_MUTED = "onCallMuted";
export declare const ON_CALL_UNMUTED = "onCallUnmuted";
export declare const ON_CALL_RESUMED = "onCallResumed";
export declare const ON_CALL_HELD = "onCallHeld";
export declare const ON_CALL_UNHELD = "onCallUnHeld";
export declare const ON_CAMERA_DISABLED = "onCameraDisabled";
export declare const ON_CAMERA_RESUMED = "onCameraResumed";
export declare const ON_CALL_CANCELED = "onCallCanceled";
export declare const ON_CALL_FAILED = "onCallFailed";
export declare const ON_CALL_REJECTED = "onCallRejected";
export declare const ON_CALL_ENDED = "onCallEnded";
export declare const ON_CALL_ENDING = "onCallEnding";
export declare const ON_MESSAGE = "onMessage";
export declare const ON_REINVITE = "reinvite";
export declare const ON_TRACK = "onTrack";
export declare const ON_AUDIO_STREAM = "onAudioStream";
export declare const ON_VIDEO_STREAM = "onVideoStream";
export declare const ON_REMOVE_STREAM = "onRemoveStream";
export declare const ON_SHARE_SCREEN_STARTED = "onScreenShareStarted";
export declare const ON_SHARE_SCREEN_ENDING = "onScreenShareEnding";
export declare const ON_SHARE_SCREEN_ENDED = "onScreenShareEnded";
export declare const ON_TERMINATE_SOUND = "terminateSound";
export declare const ON_PLAY_RING_SOUND = "playRingingSound";
export declare const ON_PLAY_INBOUND_CALL_SIGNAL_SOUND = "playInboundCallSignalSound";
export declare const ON_PLAY_HANGUP_SOUND = "playHangupSound";
export declare const ON_PLAY_PROGRESS_SOUND = "playProgressSound";
export declare const ON_VIDEO_INPUT_CHANGE = "videoInputChange";
export declare const ON_CALL_ERROR = "onCallError";
export declare const ON_MESSAGE_TRACK_UPDATED = "onTrackUpdated";
export declare const ON_NETWORK_STATS = "onNetworkStats";
export declare const ON_CHAT = "phone/ON_CHAT";
export declare const ON_SIGNAL = "phone/ON_SIGNAL";
export declare const ON_ICE_DISCONNECTED = "onIceDisconnected";
export declare const ON_ICE_RECONNECTING = "onIceReconnecting";
export declare const ON_ICE_RECONNECTED = "onIceReconnected";
export declare const ON_EARLY_MEDIA = "onEarlyMedia";
export declare const ON_MEDIA_CONNECTED = "onMediaConnected";
export declare const MESSAGE_TYPE_CHAT = "message/TYPE_CHAT";
export declare const MESSAGE_TYPE_SIGNAL = "message/TYPE_SIGNAL";
export declare const events: string[];
export default class WebRTCPhone extends Emitter implements Phone {
    client: WebRTCClient;
    allowVideo: boolean;
    callSessions: Record<string, CallSession>;
    incomingSessions: string[];
    currentSipSession?: WazoSession;
    currentCallSession?: CallSession;
    audioOutputDeviceId?: string;
    audioRingDeviceId?: string;
    audioOutputVolume: number;
    audioRingVolume: number;
    ringingEnabled: boolean;
    acceptedSessions: Record<string, any>;
    rejectedSessions: Record<string, any>;
    ignoredSessions: Record<string, any>;
    currentScreenShare: Record<string, any> | null;
    lastScreenShare: Record<string, any> | null;
    shouldSendReinvite: boolean;
    fallbackToFirstSipSession: boolean;
    constructor(client: WebRTCClient, audioOutputDeviceId: string | undefined, allowVideo?: boolean, audioRingDeviceId?: string);
    register(): Promise<void>;
    unregister(): Promise<void> | null;
    stop(): void;
    removeIncomingSessions(id: string): void;
    isWebRTC(): boolean;
    sendReinvite(callSession?: CallSession, constraints?: Record<string, any> | null | undefined, conference?: boolean, audioOnly?: boolean, iceRestart?: boolean): Promise<OutgoingInviteRequest | void>;
    getUserAgent(): string;
    startHeartbeat(): void;
    stopHeartbeat(): void;
    setOnHeartbeatTimeout(cb: (...args: Array<any>) => any): void;
    setOnHeartbeatCallback(cb: (...args: Array<any>) => any): void;
    reconnect(): void;
    getOptions(): AvailablePhoneOptions;
    onConnect(): Promise<void>;
    onDisconnect(): Promise<void>;
    _downgradeToAudio(callSession?: CallSession, withMessage?: boolean): boolean;
    _upgradeToVideo(callSession: CallSession | undefined, constraints: Record<string, any>, isConference: boolean): Promise<boolean>;
    _sendReinviteMessage(callSession: CallSession | undefined, isUpgrade: boolean): void;
    _bindEvents(sipSession: WazoSession): void;
    startScreenSharing(constraints: Record<string, any>, callSession?: CallSession): Promise<MediaStream>;
    stopScreenSharing(restoreLocalStream?: boolean, callSession?: CallSession): Promise<OutgoingInviteRequest | void | null>;
    _onScreenSharing(screenStream: MediaStream, sipSession: WazoSession, callSession: CallSession | null | undefined, hadVideo: boolean, desktop?: boolean | null | undefined): void;
    _onCallAccepted(sipSession: WazoSession, cameraEnabled: boolean): CallSession;
    changeAudioDevice(id: string): Promise<void>;
    createAudioElementFor(sessionId: string): Promise<HTMLAudioElement>;
    changeRingDevice(id: string): void;
    changeAudioVolume(volume: number): void;
    changeRingVolume(volume: number): void;
    changeAudioInputDevice(id: string, force?: boolean): Promise<MediaStream | null>;
    changeVideoInputDevice(id: string): Promise<void | MediaStream | null | undefined>;
    changeSessionVideoInputDevice(id: string | null | undefined): Promise<void | MediaStream> | undefined;
    _onCallTerminated(sipSession: WazoSession): boolean;
    setActiveSipSession(callSession: CallSession): void;
    hasAnActiveCall(): boolean;
    hasActiveRemoteVideoStream(): boolean;
    callCount(): number;
    isCurrentCallSipSession(callSession: CallSession): boolean;
    hasVideo(callSession: CallSession): boolean;
    hasAVideoTrack(callSession: CallSession): boolean;
    getLocalStreamForCall(callSession: CallSession): MediaStream | null | undefined;
    getRemoteStreamForCall(callSession: CallSession): MediaStream | null | undefined;
    getRemoteStreamsForCall(callSession: CallSession): MediaStream | null | undefined;
    accept(callSession: CallSession, cameraEnabled?: boolean): Promise<string | null>;
    isAccepted(callSession: CallSession): boolean;
    reject(callSession: CallSession): Promise<void>;
    ignore(callSession: CallSession): Promise<void>;
    atxfer(callSession: CallSession): Record<string, any> | null | undefined;
    hold(callSession: CallSession, withEvent?: boolean): Promise<OutgoingInviteRequest | void> | null | undefined;
    unhold(callSession: CallSession, withEvent?: boolean): Promise<MediaStream | void | null>;
    resume(callSession?: CallSession, withEvent?: boolean): Promise<MediaStream | void | null>;
    holdSipSession(sipSession: WazoSession, callSession?: CallSession, withEvent?: boolean): Promise<OutgoingInviteRequest | void>;
    unholdSipSession(sipSession: WazoSession, callSession: CallSession | null | undefined, withEvent?: boolean): Promise<MediaStream | void | null>;
    mute(callSession?: CallSession, withEvent?: boolean): void;
    unmute(callSession?: CallSession, withEvent?: boolean): void;
    isAudioMuted(callSession: CallSession): boolean;
    turnCameraOn(callSession?: CallSession): void;
    turnCameraOff(callSession?: CallSession): void;
    sendKey(callSession: CallSession | undefined, tone: string): void;
    makeCall(number: string, line: any, cameraEnabled?: boolean, audioOnly?: boolean, conference?: boolean, options?: {}): Promise<CallSession | null | undefined>;
    transfer(callSession: CallSession, target: string): void;
    indirectTransfer(source: CallSession, destination: CallSession): Promise<boolean>;
    initiateCTIIndirectTransfer(): Promise<null>;
    cancelCTIIndirectTransfer(): void;
    confirmCTIIndirectTransfer(): void;
    hangup(callSession?: CallSession): Promise<boolean>;
    getStats(callSession?: CallSession): Promise<RTCStatsReport | null | undefined>;
    startNetworkMonitoring(callSession?: CallSession, interval?: number): void | null;
    stopNetworkMonitoring(callSession?: CallSession): void | null;
    forceCancel(sipSession: Inviter): void;
    endCurrentCall(callSession: CallSession): void;
    onConnectionMade(): void;
    isConference(callSession: CallSession | null | undefined): boolean;
    close(force?: boolean): Promise<void>;
    isRegistered(): boolean;
    enableRinging(): Promise<void> | void;
    disableRinging(): Promise<void> | void;
    getCurrentCallSession(): CallSession | null | undefined;
    hasIncomingCallSession(): boolean;
    getIncomingCallSession(): CallSession | null | undefined;
    sendMessage(sipSession: (WazoSession | null) | undefined, body: string, contentType?: string): void;
    getLocalStream(callSession: CallSession): MediaStream | null | undefined;
    getLocalVideoStream(callSession: CallSession): MediaStream | null | undefined;
    getLocalMediaStream(callSession: CallSession): MediaStream | null | undefined;
    hasLocalVideo(callSession: CallSession): boolean;
    hasALocalVideoTrack(callSession: CallSession): boolean;
    getRemoteStream(callSession: CallSession): MediaStream | null | undefined;
    getRemoteVideoStream(callSession: CallSession): MediaStream | null | undefined;
    getRemoteVideoStreamFromPc(callSession: CallSession): MediaStream | null | undefined;
    hasRemoteVideo(callSession: CallSession): boolean;
    setMediaConstraints(media: MediaStreamConstraints): void;
    isVideoRemotelyHeld(callSession: CallSession): boolean;
    bindClientEvents(): void;
    findSipSession(callSession?: CallSession): WazoSession | null | undefined;
    getCallSession(sipSessionId: string): CallSession | null | undefined;
    getSipSessionId(sipSession: WazoSession | null | undefined): string;
    getSipSessionIdFromCallSession(callSession: CallSession): string | null;
    _updateCallSession(callSession: CallSession): void;
    _onMessage(message: Message & {
        method?: string;
        body?: string;
    }): void;
    _createIncomingCallSession(sipSession: WazoSession, cameraEnabled: boolean, fromSession?: CallSession | null | undefined, autoAnswer?: boolean): CallSession;
    _createOutgoingCallSession(sipSession: WazoSession, cameraEnabled: boolean, fromSession?: CallSession): CallSession;
    _createAcceptedCallSession(sipSession: WazoSession, cameraEnabled?: boolean, fromSession?: CallSession): CallSession;
    _createMutedCallSession(sipSession: WazoSession, fromSession?: CallSession): CallSession;
    _createUnmutedCallSession(sipSession: WazoSession, fromSession?: CallSession): CallSession;
    _createCameraResumedCallSession(sipSession: WazoSession, fromSession?: CallSession): CallSession;
    _createCameraDisabledCallSession(sipSession: WazoSession, fromSession?: CallSession): CallSession;
    _createCallSession(sipSession: WazoSession, fromSession?: CallSession | null | undefined, extra?: Record<string, any>): CallSession;
}
//# sourceMappingURL=WebRTCPhone.d.ts.map