workano-js-sdk/dist/lib/simple/Phone.d.ts

Workano Communications SDK - A modern JavaScript SDK for WebRTC and VoIP integration.

import { OutgoingInviteRequest } from 'sip.js/lib/core';
import type SipLine from '../domain/SipLine';
import type Session from '../domain/Session';
import type CallSession from '../domain/CallSession';
import AdHocAPIConference from '../domain/AdHocAPIConference';
import WebRTCPhone from '../domain/Phone/WebRTCPhone';
import WebRTCClient from '../web-rtc-client';
import Emitter from '../utils/Emitter';
import { WazoSession, WebRtcConfig } from '../domain/types';
export declare class Phone extends Emitter {
    client: WebRTCClient;
    phone: WebRTCPhone | null | undefined;
    session: Session;
    sipLine: SipLine | null | undefined;
    SessionState: Record<string, any>;
    ON_USER_AGENT: string;
    ON_REGISTERED: string;
    ON_UNREGISTERED: string;
    ON_PROGRESS: string;
    ON_CALL_ACCEPTED: string;
    ON_CALL_ANSWERED: string;
    ON_CALL_INCOMING: string;
    ON_CALL_OUTGOING: string;
    ON_CALL_MUTED: string;
    ON_CALL_UNMUTED: string;
    ON_CALL_RESUMED: string;
    ON_CALL_HELD: string;
    ON_CALL_UNHELD: string;
    ON_CAMERA_DISABLED: string;
    ON_CAMERA_RESUMED: string;
    ON_CALL_CANCELED: string;
    ON_CALL_FAILED: string;
    ON_CALL_REJECTED: string;
    ON_CALL_ENDED: string;
    ON_CALL_ENDING: string;
    ON_MESSAGE: string;
    ON_REINVITE: string;
    ON_TRACK: string;
    ON_AUDIO_STREAM: string;
    ON_VIDEO_STREAM: string;
    ON_REMOVE_STREAM: string;
    ON_SHARE_SCREEN_STARTED: string;
    ON_SHARE_SCREEN_ENDING: string;
    ON_SHARE_SCREEN_ENDED: string;
    ON_TERMINATE_SOUND: string;
    ON_PLAY_RING_SOUND: string;
    ON_PLAY_INBOUND_CALL_SIGNAL_SOUND: string;
    ON_PLAY_HANGUP_SOUND: string;
    ON_PLAY_PROGRESS_SOUND: string;
    ON_VIDEO_INPUT_CHANGE: string;
    ON_CALL_ERROR: string;
    ON_MESSAGE_TRACK_UPDATED: string;
    ON_NETWORK_STATS: string;
    ON_CHAT: string;
    ON_SIGNAL: string;
    ON_DISCONNECTED: string;
    ON_EARLY_MEDIA: string;
    MESSAGE_TYPE_CHAT: string;
    MESSAGE_TYPE_SIGNAL: string;
    constructor();
    connect(rawOptions?: Partial<WebRtcConfig>, sipLine?: SipLine | null | undefined): Promise<void>;
    connectWithCredentials(server: string, sipLine: SipLine, displayName: string, rawOptions?: Partial<WebRtcConfig>): void;
    disconnect(): Promise<void>;
    call(extension: string, withCamera?: boolean, rawSipLine?: SipLine | null | undefined, audioOnly?: boolean, conference?: boolean): Promise<CallSession | null | undefined>;
    hangup(callSession: CallSession): Promise<boolean>;
    accept(callSession: CallSession, cameraEnabled?: boolean): Promise<string | null>;
    startConference(host: CallSession, otherCalls: CallSession[]): Promise<AdHocAPIConference>;
    mute(callSession: CallSession, withApi?: boolean): void;
    unmute(callSession: CallSession, withApi?: boolean): void;
    muteViaAPI(callSession: CallSession): void;
    unmuteViaAPI(callSession: CallSession): void;
    hold(callSession: CallSession): Promise<void | OutgoingInviteRequest> | null | undefined;
    unhold(callSession: CallSession): Promise<MediaStream | void | null | undefined>;
    resume(callSession: CallSession): Promise<MediaStream | null | void>;
    reject(callSession: CallSession): Promise<void> | undefined;
    transfer(callSession: CallSession, target: string): void;
    atxfer(callSession: CallSession): Record<string, any> | null;
    reinvite(callSession: CallSession, constraints?: (Record<string, any> | null), conference?: boolean): Promise<OutgoingInviteRequest | void | null>;
    getStats(callSession: CallSession): Promise<RTCStatsReport | null | undefined>;
    startNetworkMonitoring(callSession: CallSession, interval?: number): void | null;
    stopNetworkMonitoring(callSession: CallSession): void | null;
    getSipSessionId(sipSession: WazoSession): string | null | undefined;
    sendMessage(body: string, sipSession?: WazoSession, contentType?: string): void;
    sendChat(content: string, sipSession?: WazoSession): void;
    sendSignal(content: any, sipSession?: WazoSession): void;
    turnCameraOff(callSession: CallSession): void;
    turnCameraOn(callSession: CallSession): void;
    startScreenSharing(constraints: Record<string, any>, callSession?: CallSession): Promise<MediaStream | null>;
    stopScreenSharing(callSession?: CallSession, restoreLocalStream?: boolean): Promise<OutgoingInviteRequest | void | null>;
    sendDTMF(tone: string, callSession: CallSession): void;
    getLocalStream(callSession: CallSession): MediaStream | null | undefined;
    hasLocalVideo(callSession: CallSession): boolean;
    hasALocalVideoTrack(callSession: CallSession): boolean;
    getLocalMediaStream(callSession: CallSession): MediaStream | null | undefined;
    getLocalVideoStream(callSession: CallSession): MediaStream | null;
    getRemoteStream(callSession: CallSession): MediaStream | null;
    getRemoteVideoStream(callSession: CallSession): MediaStream | null;
    isVideoRemotelyHeld(callSession: CallSession): boolean;
    getRemoteStreamForCall(callSession: CallSession): MediaStream | null;
    getRemoteStreamsForCall(callSession: CallSession): MediaStream | null;
    getRemoteVideoStreamForCall(callSession: CallSession): MediaStream | null;
    getRemoteVideoStreamFromPc(callSession: CallSession): MediaStream | null;
    hasVideo(callSession: CallSession): boolean;
    hasAVideoTrack(callSession: CallSession): boolean;
    getCurrentSipSession(): WazoSession | null;
    getPrimaryWebRtcLine(): SipLine | null;
    getOutputDevice(): string | null;
    getPrimaryLine(): SipLine | null;
    getLineById(lineId: string): SipLine | null;
    getSipLines(): SipLine[];
    hasSfu(): boolean;
    checkSfu(): void;
    enableLogger(): void;
    _transferEvents(): void;
    _logConnector(level: any, className: string, label: any, content: string): void;
}
declare const _default: Phone;
export default _default;
//# sourceMappingURL=Phone.d.ts.map