@wazo/sdk 0.43.15 → 0.44.0

package/lib/domain/Phone/WebRTCPhone.js

@@ -1,1492 +0,0 @@
-"use strict";
-var __awaiter = (this && this.__awaiter) || function (thisArg, _arguments, P, generator) {
-    function adopt(value) { return value instanceof P ? value : new P(function (resolve) { resolve(value); }); }
-    return new (P || (P = Promise))(function (resolve, reject) {
-        function fulfilled(value) { try { step(generator.next(value)); } catch (e) { reject(e); } }
-        function rejected(value) { try { step(generator["throw"](value)); } catch (e) { reject(e); } }
-        function step(result) { result.done ? resolve(result.value) : adopt(result.value).then(fulfilled, rejected); }
-        step((generator = generator.apply(thisArg, _arguments || [])).next());
-    });
-};
-var __importDefault = (this && this.__importDefault) || function (mod) {
-    return (mod && mod.__esModule) ? mod : { "default": mod };
-};
-Object.defineProperty(exports, "__esModule", { value: true });
-exports.events = exports.MESSAGE_TYPE_SIGNAL = exports.MESSAGE_TYPE_CHAT = exports.ON_EARLY_MEDIA = exports.ON_DISCONNECTED = exports.ON_SIGNAL = exports.ON_CHAT = exports.ON_NETWORK_STATS = exports.ON_MESSAGE_TRACK_UPDATED = exports.ON_CALL_ERROR = exports.ON_VIDEO_INPUT_CHANGE = exports.ON_PLAY_PROGRESS_SOUND = exports.ON_PLAY_HANGUP_SOUND = exports.ON_PLAY_INBOUND_CALL_SIGNAL_SOUND = exports.ON_PLAY_RING_SOUND = exports.ON_TERMINATE_SOUND = exports.ON_SHARE_SCREEN_ENDED = exports.ON_SHARE_SCREEN_ENDING = exports.ON_SHARE_SCREEN_STARTED = exports.ON_REMOVE_STREAM = exports.ON_VIDEO_STREAM = exports.ON_AUDIO_STREAM = exports.ON_TRACK = exports.ON_REINVITE = exports.ON_MESSAGE = exports.ON_CALL_ENDING = exports.ON_CALL_ENDED = exports.ON_CALL_REJECTED = exports.ON_CALL_FAILED = exports.ON_CALL_CANCELED = exports.ON_CAMERA_RESUMED = exports.ON_CAMERA_DISABLED = exports.ON_CALL_UNHELD = exports.ON_CALL_HELD = exports.ON_CALL_RESUMED = exports.ON_CALL_UNMUTED = exports.ON_CALL_MUTED = exports.ON_CALL_OUTGOING = exports.ON_CALL_INCOMING = exports.ON_CALL_ANSWERED = exports.ON_CALL_ACCEPTED = exports.ON_PROGRESS = exports.ON_UNREGISTERED = exports.ON_REGISTERED = exports.ON_USER_AGENT = void 0;
-const session_state_1 = require("sip.js/lib/api/session-state");
-const api_1 = require("sip.js/lib/api");
-const CallSession_1 = __importDefault(require("../CallSession"));
-const Emitter_1 = __importDefault(require("../../utils/Emitter"));
-const IssueReporter_1 = __importDefault(require("../../service/IssueReporter"));
-exports.ON_USER_AGENT = 'onUserAgent';
-exports.ON_REGISTERED = 'onRegistered';
-exports.ON_UNREGISTERED = 'onUnRegistered';
-exports.ON_PROGRESS = 'onProgress';
-exports.ON_CALL_ACCEPTED = 'onCallAccepted';
-exports.ON_CALL_ANSWERED = 'onCallAnswered';
-exports.ON_CALL_INCOMING = 'onCallIncoming';
-exports.ON_CALL_OUTGOING = 'onCallOutgoing';
-exports.ON_CALL_MUTED = 'onCallMuted';
-exports.ON_CALL_UNMUTED = 'onCallUnmuted';
-exports.ON_CALL_RESUMED = 'onCallResumed';
-exports.ON_CALL_HELD = 'onCallHeld';
-exports.ON_CALL_UNHELD = 'onCallUnHeld';
-exports.ON_CAMERA_DISABLED = 'onCameraDisabled';
-exports.ON_CAMERA_RESUMED = 'onCameraResumed';
-exports.ON_CALL_CANCELED = 'onCallCanceled';
-exports.ON_CALL_FAILED = 'onCallFailed';
-exports.ON_CALL_REJECTED = 'onCallRejected';
-exports.ON_CALL_ENDED = 'onCallEnded';
-exports.ON_CALL_ENDING = 'onCallEnding';
-exports.ON_MESSAGE = 'onMessage';
-exports.ON_REINVITE = 'reinvite';
-exports.ON_TRACK = 'onTrack';
-exports.ON_AUDIO_STREAM = 'onAudioStream';
-exports.ON_VIDEO_STREAM = 'onVideoStream';
-exports.ON_REMOVE_STREAM = 'onRemoveStream';
-exports.ON_SHARE_SCREEN_STARTED = 'onScreenShareStarted';
-exports.ON_SHARE_SCREEN_ENDING = 'onScreenShareEnding';
-exports.ON_SHARE_SCREEN_ENDED = 'onScreenShareEnded';
-exports.ON_TERMINATE_SOUND = 'terminateSound';
-exports.ON_PLAY_RING_SOUND = 'playRingingSound';
-exports.ON_PLAY_INBOUND_CALL_SIGNAL_SOUND = 'playInboundCallSignalSound';
-exports.ON_PLAY_HANGUP_SOUND = 'playHangupSound';
-exports.ON_PLAY_PROGRESS_SOUND = 'playProgressSound';
-exports.ON_VIDEO_INPUT_CHANGE = 'videoInputChange';
-exports.ON_CALL_ERROR = 'onCallError';
-exports.ON_MESSAGE_TRACK_UPDATED = 'onTrackUpdated';
-exports.ON_NETWORK_STATS = 'onNetworkStats';
-exports.ON_CHAT = 'phone/ON_CHAT';
-exports.ON_SIGNAL = 'phone/ON_SIGNAL';
-exports.ON_DISCONNECTED = 'onDisconnected';
-exports.ON_EARLY_MEDIA = 'onEarlyMedia';
-exports.MESSAGE_TYPE_CHAT = 'message/TYPE_CHAT';
-exports.MESSAGE_TYPE_SIGNAL = 'message/TYPE_SIGNAL';
-exports.events = [exports.ON_USER_AGENT, exports.ON_REGISTERED, exports.ON_UNREGISTERED, exports.ON_PROGRESS, exports.ON_CALL_ACCEPTED, exports.ON_CALL_ANSWERED, exports.ON_CALL_INCOMING, exports.ON_CALL_OUTGOING, exports.ON_CALL_MUTED, exports.ON_CALL_UNMUTED, exports.ON_CALL_RESUMED, exports.ON_CALL_HELD, exports.ON_CALL_UNHELD, exports.ON_CAMERA_DISABLED, exports.ON_CALL_FAILED, exports.ON_CALL_ENDED, exports.ON_CALL_REJECTED, exports.ON_MESSAGE, exports.ON_REINVITE, exports.ON_TRACK, exports.ON_AUDIO_STREAM, exports.ON_VIDEO_STREAM, exports.ON_REMOVE_STREAM, exports.ON_SHARE_SCREEN_ENDED, exports.ON_TERMINATE_SOUND, exports.ON_PLAY_RING_SOUND, exports.ON_PLAY_INBOUND_CALL_SIGNAL_SOUND, exports.ON_PLAY_HANGUP_SOUND, exports.ON_PLAY_PROGRESS_SOUND, exports.ON_VIDEO_INPUT_CHANGE, exports.ON_SHARE_SCREEN_STARTED, exports.ON_CALL_ERROR, exports.ON_CHAT, exports.ON_SIGNAL, exports.ON_NETWORK_STATS, exports.ON_DISCONNECTED, exports.ON_EARLY_MEDIA];
-const logger = IssueReporter_1.default.loggerFor('webrtc-phone');
-class WebRTCPhone extends Emitter_1.default {
-    constructor(client, audioOutputDeviceId, allowVideo = false, audioRingDeviceId) {
-        super();
-        this.client = client;
-        this.allowVideo = allowVideo;
-        this.callSessions = {};
-        this.audioOutputDeviceId = audioOutputDeviceId;
-        this.audioRingDeviceId = audioRingDeviceId || audioOutputDeviceId;
-        this.audioOutputVolume = 1;
-        this.audioRingVolume = 1;
-        this.incomingSessions = [];
-        this.ringingEnabled = true;
-        this.shouldSendReinvite = false;
-        this.bindClientEvents();
-        this.acceptedSessions = {};
-        this.rejectedSessions = {};
-        this.ignoredSessions = {};
-    }
-    register() {
-        logger.info('WebRTC register', { client: !!this.client });
-        if (!this.client) {
-            return Promise.resolve();
-        }
-        return this.client.register().then(() => {
-            logger.info('WebRTC register, registered');
-            return this.bindClientEvents();
-        }).catch(error => {
-            // Avoid exception on `t.server.scheme` in sip transport when losing the webrtc socket connection
-            console.error('register error', error, error.message, error.stack);
-            logger.error('WebRTC register error', {
-                message: error.message,
-                stack: error.stack,
-            });
-        });
-    }
-    unregister() {
-        if (!this.client || !this.client.isRegistered()) {
-            return null;
-        }
-        return this.client.unregister();
-    }
-    stop() {
-        if (!this.client) {
-            return;
-        }
-        logger.info('WebRTC phone stop');
-        this.client.stop();
-    }
-    removeIncomingSessions(id) {
-        this.incomingSessions = this.incomingSessions.filter(sessionId => sessionId !== id);
-    }
-    isWebRTC() {
-        return true;
-    }
-    sendReinvite(callSession_1) {
-        return __awaiter(this, arguments, void 0, function* (callSession, constraints = null, conference = false, audioOnly = false, iceRestart = false) {
-            const sipSession = this.findSipSession(callSession);
-            logger.info('WebRTC phone - send reinvite', {
-                sessionId: this.getSipSessionId(sipSession),
-                constraints,
-                audioOnly,
-                conference,
-                iceRestart,
-            });
-            if (!sipSession) {
-                return;
-            }
-            const shouldScreenShare = constraints === null || constraints === void 0 ? void 0 : constraints.screen;
-            const isUpgrade = shouldScreenShare || (constraints === null || constraints === void 0 ? void 0 : constraints.video);
-            // Downgrade
-            if (constraints && !isUpgrade) {
-                // No reinvite needed
-                this._downgradeToAudio(callSession);
-                return;
-            }
-            if (isUpgrade) {
-                const shouldReinvite = yield this._upgradeToVideo(callSession, constraints, conference);
-                if (!shouldReinvite) {
-                    return;
-                }
-            }
-            return this.client.reinvite(sipSession, constraints, conference, audioOnly, iceRestart);
-        });
-    }
-    getUserAgent() {
-        var _a, _b;
-        return ((_b = (_a = this.client) === null || _a === void 0 ? void 0 : _a.config) === null || _b === void 0 ? void 0 : _b.userAgentString) || 'webrtc-phone';
-    }
-    startHeartbeat() {
-        logger.info('WebRTC phone - start heartbeat', {
-            client: !!this.client,
-            hasHeartbeat: this.client.hasHeartbeat(),
-        });
-        if (!this.client || this.client.hasHeartbeat()) {
-            return;
-        }
-        this.client.startHeartbeat();
-    }
-    stopHeartbeat() {
-        logger.info('WebRTC phone - stopHeartbeat', {
-            client: !!this.client,
-        });
-        if (!this.client) {
-            return;
-        }
-        this.client.stopHeartbeat();
-    }
-    setOnHeartbeatTimeout(cb) {
-        this.client.setOnHeartbeatTimeout(cb);
-    }
-    setOnHeartbeatCallback(cb) {
-        this.client.setOnHeartbeatCallback(cb);
-    }
-    reconnect() {
-        logger.info('WebRTC phone - reconnect', {
-            client: !!this.client,
-        });
-        if (!this.client) {
-            return;
-        }
-        this.client.attemptReconnection();
-    }
-    getOptions() {
-        return {
-            accept: true,
-            decline: true,
-            mute: true,
-            hold: true,
-            transfer: true,
-            sendKey: true,
-            addParticipant: false,
-            record: true,
-            merge: true,
-        };
-    }
-    onConnect() {
-        if (!this.client) {
-            return Promise.reject(new Error('No webrtc client'));
-        }
-        return this.client.onConnect();
-    }
-    onDisconnect() {
-        if (!this.client) {
-            return Promise.reject(new Error('No webrtc client'));
-        }
-        return this.client.onDisconnect();
-    }
-    _downgradeToAudio(callSession, withMessage = true) {
-        const sipSession = this.findSipSession(callSession);
-        if (!sipSession) {
-            return false;
-        }
-        logger.info('Downgrade to audio', {
-            id: callSession ? callSession.getId() : null,
-            withMessage,
-        });
-        this.client.downgradeToAudio(sipSession);
-        if (callSession) {
-            callSession.cameraEnabled = false;
-            this._updateCallSession(callSession);
-        }
-        if (withMessage) {
-            this._sendReinviteMessage(callSession, false);
-        }
-        return true;
-    }
-    // Returns true if we need to send a re-INVITE request
-    _upgradeToVideo(callSession, constraints, isConference) {
-        return __awaiter(this, void 0, void 0, function* () {
-            const sipSession = this.findSipSession(callSession);
-            if (!sipSession) {
-                return Promise.resolve(false);
-            }
-            logger.info('Upgrade to video', {
-                id: callSession ? callSession.getId() : null,
-                constraints,
-                isConference,
-            });
-            const shouldScreenShare = constraints && !!constraints.screen;
-            const desktop = constraints && constraints.desktop;
-            const options = sipSession.sessionDescriptionHandlerOptionsReInvite;
-            const wasAudioOnly = options === null || options === void 0 ? void 0 : options.audioOnly;
-            const newStream = yield this.client.upgradeToVideo(sipSession, constraints, isConference);
-            if (callSession) {
-                callSession.cameraEnabled = true;
-                this._updateCallSession(callSession);
-            }
-            // If no stream is returned, it means we have to reinvite
-            if (!newStream) {
-                return true;
-            }
-            this._sendReinviteMessage(callSession, true);
-            if (shouldScreenShare) {
-                this._onScreenSharing(newStream, sipSession, callSession, false, desktop);
-            }
-            // We have to reinvite to change the direction on the bundle when upgrading from an audioOnly conference
-            if (isConference && wasAudioOnly) {
-                return true;
-            }
-            // No reinvite needed here
-            return false;
-        });
-    }
-    _sendReinviteMessage(callSession, isUpgrade) {
-        const sipSession = this.findSipSession(callSession);
-        logger.info('Sending reinvite message', {
-            id: callSession ? callSession.getId() : null,
-            isUpgrade,
-        });
-        // Have to send the message after a delay due to latency to update the remote peer
-        setTimeout(() => {
-            this.sendMessage(sipSession, JSON.stringify({
-                type: exports.MESSAGE_TYPE_SIGNAL,
-                content: {
-                    type: exports.ON_MESSAGE_TRACK_UPDATED,
-                    update: isUpgrade ? 'upgrade' : 'downgrade',
-                    sipCallId: sipSession && this.getSipSessionId(sipSession),
-                    callId: callSession ? callSession.callId : null,
-                    number: callSession ? callSession.number : null,
-                    callerNumber: callSession ? callSession.callerNumber : null,
-                },
-            }));
-        }, 2500);
-    }
-    _bindEvents(sipSession) {
-        const sipSessionId = this.getSipSessionId(sipSession);
-        if (sipSession instanceof api_1.Invitation) {
-            // Monkey patch to know when canceled with the CANCEL message
-            const onCancel = sipSession._onCancel.bind(sipSession);
-            sipSession._onCancel = (message) => {
-                logger.trace('on sip session canceled', {
-                    callId: message.callId,
-                    sipSessionId,
-                });
-                onCancel(message);
-                const elsewhere = message.data.indexOf('cause=26') !== -1 && message.data.indexOf('completed elsewhere') !== -1;
-                const callSession = this._createCallSession(sipSession);
-                if (sipSessionId) {
-                    delete this.callSessions[sipSessionId];
-                }
-                this.eventEmitter.emit(exports.ON_CALL_CANCELED, callSession, elsewhere);
-            };
-        }
-        sipSession.stateChange.addListener((newState) => {
-            switch (newState) {
-                case session_state_1.SessionState.Terminating:
-                    logger.info('WebRTC phone - call terminating', {
-                        sipId: this.getSipSessionId(sipSession),
-                    });
-                    this.eventEmitter.emit(exports.ON_CALL_ENDING, this._createCallSession(sipSession));
-                    break;
-                case session_state_1.SessionState.Terminated:
-                    {
-                        logger.info('WebRTC phone - call terminated', {
-                            sipId: this.getSipSessionId(sipSession),
-                        });
-                        // Should be called before `_onCallTerminated` or the callCount will not decrement...
-                        const callSession = this._createCallSession(sipSession);
-                        callSession.endTime = new Date();
-                        const wasCurrentSession = this._onCallTerminated(sipSession);
-                        return this.eventEmitter.emit(exports.ON_CALL_ENDED, callSession, wasCurrentSession);
-                    }
-                default:
-                    break;
-            }
-        });
-        if (!sipSession.sessionDescriptionHandler) {
-            return;
-        }
-        // Video events
-        const sdh = sipSession.sessionDescriptionHandler;
-        const peerConnection = sdh.peerConnection;
-        if (!peerConnection) {
-            logger.info('null peer connection');
-            return;
-        }
-        peerConnection.ontrack = (rawEvent) => {
-            const event = rawEvent;
-            const [stream] = event.streams;
-            const kind = event && event.track && event.track.kind;
-            logger.info('on track stream called on the peer connection', {
-                callId: this.getSipSessionId(sipSession),
-                streamId: stream ? stream.id : null,
-                tracks: stream ? stream.getTracks() : null,
-                kind,
-            });
-            if (kind === 'audio') {
-                return this.eventEmitter.emit(exports.ON_AUDIO_STREAM, stream);
-            }
-            return this.eventEmitter.emit(exports.ON_VIDEO_STREAM, stream, event.track.id, event, sipSession);
-        };
-        peerConnection.onremovestream = (event) => {
-            logger.info('on remove stream called on the peer connection', {
-                id: event.stream.id,
-                tracks: event.stream.getTracks(),
-            });
-            this.eventEmitter.emit(exports.ON_REMOVE_STREAM, event.stream);
-        };
-    }
-    startScreenSharing(constraints, callSession) {
-        return __awaiter(this, void 0, void 0, function* () {
-            logger.info('WebRTC - start screen sharing', {
-                constraints,
-                id: callSession ? callSession.getId() : null,
-            });
-            const screenShareStream = yield this.client.getStreamFromConstraints(constraints);
-            if (!screenShareStream) {
-                throw new Error(`Can't create media stream for screensharing with: ${JSON.stringify(constraints)}`);
-            }
-            const screenTrack = screenShareStream.getVideoTracks()[0];
-            const sipSession = this.currentSipSession;
-            const pc = sipSession && this.client.getPeerConnection(this.getSipSessionId(sipSession));
-            const sender = pc && pc.getSenders && pc.getSenders().find((s) => s && s.track && s.track.kind === 'video');
-            const localStream = sipSession && this.client.getLocalStream(this.getSipSessionId(sipSession));
-            const videoTrack = localStream ? localStream.getTracks().find(track => track.kind === 'video') : null;
-            const hadVideo = !!videoTrack;
-            // Stop local video tracks
-            if (videoTrack && localStream) {
-                videoTrack.enabled = false;
-                videoTrack.stop();
-                localStream.removeTrack(videoTrack);
-            }
-            if (localStream) {
-                localStream.addTrack(screenTrack);
-            }
-            if (sender) {
-                sender.replaceTrack(screenTrack);
-            }
-            this._onScreenSharing(screenShareStream, sipSession, callSession, hadVideo, constraints.desktop);
-            return screenShareStream;
-        });
-    }
-    stopScreenSharing() {
-        return __awaiter(this, arguments, void 0, function* (restoreLocalStream = true, callSession) {
-            if (!this.currentScreenShare) {
-                return;
-            }
-            logger.info('WebRTC phone - stop screen sharing', {
-                restoreLocalStream,
-            });
-            let reinvited = null;
-            try {
-                if (this.currentScreenShare.stream) {
-                    this.currentScreenShare.stream.getTracks()
-                        .filter((track) => track.enabled && track.kind === 'video')
-                        .forEach((track) => track.stop());
-                }
-                if (restoreLocalStream) {
-                    const targetCallSession = callSession || this.currentCallSession;
-                    const conference = this.isConference(targetCallSession);
-                    // When stopping screenshare and we had video before that, we have to re-upgrade
-                    // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
-                    // We have to downgrade to audio.
-                    const screenshareStopped = this.currentScreenShare.hadVideo;
-                    const constraints = {
-                        audio: false,
-                        video: screenshareStopped,
-                    };
-                    // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
-                    this._downgradeToAudio(targetCallSession, false);
-                    // Wait for the track to be removed, unless the video sender won't have a null track in `_upgradeToVideo`.
-                    yield new Promise(resolve => setTimeout(resolve, 500));
-                    reinvited = yield this.sendReinvite(targetCallSession, constraints, conference);
-                }
-            }
-            catch (e) {
-                console.warn(e);
-            }
-            const sipSession = this.currentSipSession;
-            this.eventEmitter.emit(exports.ON_SHARE_SCREEN_ENDED, callSession && sipSession ? this._createCallSession(sipSession, callSession, {
-                screensharing: false,
-            }) : null);
-            this.currentScreenShare = null;
-            return reinvited;
-        });
-    }
-    _onScreenSharing(screenStream, sipSession, callSession, hadVideo, desktop) {
-        const screenTrack = screenStream.getVideoTracks()[0];
-        const sipSessionId = this.getSipSessionId(sipSession);
-        const pc = this.client.getPeerConnection(sipSessionId);
-        const sender = pc && pc.getSenders && pc.getSenders().find((s) => s && s.track && s.track.kind === 'video');
-        logger.info('WebRTC phone - on screensharing', {
-            hadVideo,
-            id: this.getSipSessionId(sipSession),
-            screenTrack,
-        });
-        if (screenTrack) {
-            screenTrack.onended = () => {
-                logger.info('WebRTC phone - on screenshare ended', {
-                    hadVideo,
-                    id: this.getSipSessionId(sipSession),
-                    screenTrack,
-                });
-                this.eventEmitter.emit(exports.ON_SHARE_SCREEN_ENDING, this._createCallSession(sipSession, callSession));
-            };
-        }
-        this.currentScreenShare = {
-            stream: screenStream,
-            hadVideo,
-            sender,
-            sipSessionId,
-            desktop,
-        };
-        this.client.setLocalMediaStream(this.getSipSessionId(sipSession), screenStream);
-        this.eventEmitter.emit(exports.ON_SHARE_SCREEN_STARTED, this._createCallSession(sipSession, callSession, {
-            screensharing: true,
-        }), screenStream);
-    }
-    _onCallAccepted(sipSession, cameraEnabled) {
-        logger.info('WebRTC phone - on call accepted', {
-            sipId: this.getSipSessionId(sipSession),
-            cameraEnabled,
-        });
-        const callSession = this._createAcceptedCallSession(sipSession, cameraEnabled);
-        if (this.currentSipSession && !this.isCurrentCallSipSession(callSession)) {
-            logger.info('WebRTC phone - on call accepted with another ongoing call, holding it', {
-                sipId: this.getSipSessionId(sipSession),
-                cameraEnabled,
-            });
-            this.holdSipSession(this.currentSipSession, this.currentCallSession, true);
-        }
-        this.currentSipSession = sipSession;
-        this.currentCallSession = callSession;
-        this.eventEmitter.emit(exports.ON_TERMINATE_SOUND, callSession, 'call accepted');
-        const sipSessionId = this.getSipSessionId(sipSession);
-        if (sipSessionId) {
-            this.removeIncomingSessions(sipSessionId);
-        }
-        callSession.answerTime = new Date();
-        this._updateCallSession(callSession);
-        this.eventEmitter.emit(exports.ON_CALL_ACCEPTED, callSession, cameraEnabled);
-        return callSession;
-    }
-    changeAudioDevice(id) {
-        return __awaiter(this, void 0, void 0, function* () {
-            logger.info('WebRTC phone - change audio device', {
-                deviceId: id,
-            });
-            this.audioOutputDeviceId = id;
-            return this.client.changeAudioOutputDevice(id);
-        });
-    }
-    createAudioElementFor(sessionId) {
-        return this.client.createAudioElementFor(sessionId);
-    }
-    changeRingDevice(id) {
-        logger.info('WebRTC phone - changing ring device', {
-            id,
-        });
-        this.audioRingDeviceId = id;
-    }
-    // volume is a value between 0 and 1
-    changeAudioVolume(volume) {
-        logger.info('WebRTC phone - changing audio volume', {
-            volume,
-        });
-        this.audioOutputVolume = volume;
-        this.client.changeAudioOutputVolume(volume);
-    }
-    // volume is a value between 0 and 1
-    changeRingVolume(volume) {
-        logger.info('WebRTC phone - changing ring volume', {
-            volume,
-        });
-        this.audioRingVolume = volume;
-    }
-    changeAudioInputDevice(id, force) {
-        logger.info('WebRTC phone - changeAudio input device', {
-            deviceId: id,
-        });
-        return this.client.changeAudioInputDevice(id, this.currentSipSession, force);
-    }
-    changeVideoInputDevice(id) {
-        return __awaiter(this, void 0, void 0, function* () {
-            logger.info('WebRTC phone - change video input device', {
-                deviceId: id,
-            });
-            return this.client.changeVideoInputDevice(id, this.currentSipSession);
-        });
-    }
-    changeSessionVideoInputDevice(id) {
-        logger.info('WebRTC phone - change session video input device', {
-            deviceId: id,
-        });
-        return this.currentSipSession && this.client.changeSessionVideoInputDevice(id, this.currentSipSession);
-    }
-    _onCallTerminated(sipSession) {
-        const sipSessionId = this.getSipSessionId(sipSession);
-        logger.info('WebRTC phone - on call terminated', {
-            sipId: this.getSipSessionId(sipSession),
-            sipSessionId,
-            ids: Object.keys(this.callSessions),
-        });
-        const callSession = this._createCallSession(sipSession);
-        const isCurrentSession = this.isCurrentCallSipSession(callSession);
-        const isCurrentIncomingCall = callSession.is(this.getIncomingCallSession());
-        // If the terminated call was an incoming call, we have to re-trigger if it's was the first incoming call
-        // Otherwise, we have to re-trigger an incoming call event if another call is incoming
-        const shouldRetrigger = isCurrentSession ? this.incomingSessions.length : isCurrentIncomingCall;
-        // Terminate incoming sound only for the ringing call
-        // (so another terminated call won't stop the incoming call sound)
-        if (isCurrentIncomingCall) {
-            setTimeout(() => {
-                // Avoid race condition when the other is calling and hanging up immediately
-                this.eventEmitter.emit(exports.ON_TERMINATE_SOUND, callSession, 'call terminated');
-            }, 5);
-        }
-        if (sipSessionId) {
-            this.removeIncomingSessions(sipSessionId);
-            delete this.callSessions[sipSessionId];
-        }
-        if (isCurrentSession) {
-            this.currentSipSession = undefined;
-            this.currentCallSession = undefined;
-        }
-        const hasIncomingCallSession = this.hasIncomingCallSession();
-        // Re-trigger incoming call event for remaining incoming calls
-        logger.info('WebRTC phone - check to re-trigger incoming call', {
-            sipId: this.getSipSessionId(sipSession),
-            shouldRetrigger,
-            hasIncomingCallSession,
-        });
-        this.client.onCallEnded(sipSession);
-        if (hasIncomingCallSession && shouldRetrigger) {
-            const nextCallSession = this.getIncomingCallSession();
-            // Avoid race condition
-            setTimeout(() => {
-                if (this.ringingEnabled) {
-                    if (!this.currentCallSession) {
-                        this.eventEmitter.emit(exports.ON_PLAY_RING_SOUND, this.audioRingDeviceId, this.audioRingVolume, nextCallSession);
-                    }
-                    else {
-                        this.eventEmitter.emit(exports.ON_PLAY_INBOUND_CALL_SIGNAL_SOUND, this.audioOutputDeviceId, this.audioOutputVolume, nextCallSession);
-                    }
-                }
-                this.eventEmitter.emit(exports.ON_CALL_INCOMING, nextCallSession, nextCallSession === null || nextCallSession === void 0 ? void 0 : nextCallSession.cameraEnabled);
-            }, 100);
-        }
-        if (callSession.getId() in this.ignoredSessions) {
-            return false;
-        }
-        // @ts-ignore: does not exist
-        if (!sipSession.isCanceled) {
-            setTimeout(() => {
-                this.eventEmitter.emit(exports.ON_PLAY_HANGUP_SOUND, this.audioOutputDeviceId, this.audioOutputVolume, callSession);
-            }, 10);
-        }
-        return isCurrentSession;
-    }
-    setActiveSipSession(callSession) {
-        const sipSessionId = this.findSipSession(callSession);
-        if (!sipSessionId) {
-            return;
-        }
-        this.currentSipSession = sipSessionId;
-        this.currentCallSession = callSession;
-    }
-    hasAnActiveCall() {
-        return !!this.currentSipSession;
-    }
-    // /!\ In some case with react-native webrtc the session will have only one audio stream set
-    // Maybe due to https://github.com/react-native-webrtc/react-native-webrtc/issues/401
-    hasActiveRemoteVideoStream() {
-        const sipSession = this.currentSipSession;
-        if (!sipSession) {
-            return false;
-        }
-        const sdh = sipSession.sessionDescriptionHandler;
-        const peerConnection = sdh.peerConnection;
-        const remoteStream = peerConnection.getRemoteStreams().find((stream) => !!stream.getVideoTracks().length);
-        return remoteStream && remoteStream.getVideoTracks().some(track => !track.muted);
-    }
-    callCount() {
-        return Object.keys(this.callSessions).length;
-    }
-    isCurrentCallSipSession(callSession) {
-        if (!this.currentSipSession) {
-            return false;
-        }
-        return this.currentSipSession && this.getSipSessionId(this.currentSipSession) === callSession.getId();
-    }
-    hasVideo(callSession) {
-        return this.client.hasVideo(callSession.getId());
-    }
-    hasAVideoTrack(callSession) {
-        return this.client.hasAVideoTrack(callSession.getId());
-    }
-    // Deprecated
-    getLocalStreamForCall(callSession) {
-        logger.warn('WebRTCPhone.getLocalStreamForCall is deprecated, use WebRTCPhone.getLocalStream instead');
-        return this.getLocalStream(callSession);
-    }
-    // Deprecated
-    getRemoteStreamForCall(callSession) {
-        logger.warn('WebRTCPhone.getRemoteStreamForCall is deprecated, use WebRTCPhone.getRemoteStream instead');
-        return this.getRemoteVideoStream(callSession);
-    }
-    getRemoteStreamsForCall(callSession) {
-        logger.warn('WebRTCPhone.getRemoteStreamsForCall is deprecated, use WebRTCPhone.getRemoteStreams instead');
-        return this.getRemoteStream(callSession);
-    }
-    accept(callSession, cameraEnabled) {
-        logger.info('WebRTC phone - accept call', {
-            id: callSession ? callSession.getId() : 'n/a',
-            cameraEnabled,
-        });
-        if (this.currentSipSession && !this.isCurrentCallSipSession(callSession)) {
-            this.holdSipSession(this.currentSipSession, this.currentCallSession, true);
-        }
-        if (!callSession) {
-            logger.warn('no CallSession to accept.');
-            return Promise.resolve(null);
-        }
-        const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
-        if (sipSessionId && !(sipSessionId in this.callSessions)) {
-            logger.warn('Call session already ended or not found, can\'t accept it.', {
-                sipSessionId,
-            });
-            return Promise.resolve(null);
-        }
-        if (callSession.getId() in this.acceptedSessions) {
-            logger.warn('CallSession already accepted.', {
-                id: callSession ? callSession.getId() : 'n/a',
-            });
-            return Promise.resolve(callSession.sipCallId);
-        }
-        this.shouldSendReinvite = false;
-        callSession.answerTime = new Date();
-        this._updateCallSession(callSession);
-        this.acceptedSessions[callSession.getId()] = true;
-        this.eventEmitter.emit(exports.ON_CALL_ANSWERED, callSession);
-        const sipSession = this.client.getSipSession(callSession.getId());
-        if (sipSession) {
-            logger.info('accept call, session found', { sipId: this.getSipSessionId(sipSession) });
-            return this.client.answer(sipSession, this.allowVideo ? cameraEnabled : false).then(() => {
-                return callSession.sipCallId;
-            }).catch(e => {
-                logger.error(e);
-                this.endCurrentCall(callSession);
-                throw e;
-            });
-        }
-        const error = 'No CallSession found to accept.';
-        logger.warn(error, {
-            id: callSession ? callSession.getId() : 'n/a',
-            ids: this.client.getSipSessionIds(),
-        });
-        if (callSession) {
-            this.eventEmitter.emit(exports.ON_CALL_ERROR, new Error(error), callSession);
-        }
-        return Promise.resolve(null);
-    }
-    isAccepted(callSession) {
-        return callSession && callSession.getId() in this.acceptedSessions;
-    }
-    reject(callSession) {
-        return __awaiter(this, void 0, void 0, function* () {
-            logger.info('reject WebRTC called', {
-                id: callSession.getId(),
-            });
-            this.eventEmitter.emit(exports.ON_TERMINATE_SOUND, callSession, 'call rejected locally');
-            if (!callSession || callSession.getId() in this.rejectedSessions) {
-                return;
-            }
-            this.shouldSendReinvite = false;
-            this.rejectedSessions[callSession.getId()] = true;
-            const sipSession = this.findSipSession(callSession);
-            if (sipSession) {
-                logger.info('WebRTC rejecting', {
-                    sipId: this.getSipSessionId(sipSession),
-                });
-                this.client.hangup(sipSession);
-            }
-        });
-    }
-    ignore(callSession) {
-        return __awaiter(this, void 0, void 0, function* () {
-            logger.info('WebRTC ignore', {
-                id: callSession.getId(),
-            });
-            // kill the ring
-            this.eventEmitter.emit(exports.ON_TERMINATE_SOUND, callSession, 'ignoring call');
-            this.ignoredSessions[callSession.getId()] = true;
-            callSession.ignore();
-        });
-    }
-    atxfer(callSession) {
-        const sipSession = this.findSipSession(callSession);
-        if (sipSession) {
-            logger.info('WebRTC atxfer', {
-                sipId: this.getSipSessionId(sipSession),
-            });
-            return this.client.atxfer(sipSession);
-        }
-    }
-    hold(callSession, withEvent = true) {
-        logger.info('WebRTC hold', {
-            id: callSession.getId(),
-        });
-        const sipSession = this.findSipSession(callSession);
-        return sipSession ? this.holdSipSession(sipSession, callSession, withEvent) : null;
-    }
-    // @Deprecated
-    unhold(callSession, withEvent = true) {
-        console.warn('Please note that `phone.unhold()` is being deprecated; `phone.resume()` is the preferred method');
-        return this.resume(callSession, withEvent);
-    }
-    resume(callSession, withEvent = true) {
-        logger.info('WebRTC resume called', {
-            id: callSession ? callSession.getId() : null,
-        });
-        const sipSession = this.findSipSession(callSession);
-        if (!sipSession) {
-            return new Promise((resolve, reject) => reject(new Error('No session to resume')));
-        }
-        logger.info('WebRTC resuming', {
-            sipId: this.getSipSessionId(sipSession),
-        });
-        // Hold current session if different from the current one (we don't want 2 sessions active at the same time).
-        if (this.currentSipSession && this.getSipSessionId(this.currentSipSession) !== this.getSipSessionId(sipSession)) {
-            logger.info('WebRTC hold call after resume', {
-                id: this.getSipSessionId(this.currentSipSession),
-            });
-            this.holdSipSession(this.currentSipSession, this.currentCallSession, withEvent);
-        }
-        const promise = this.unholdSipSession(sipSession, callSession, withEvent);
-        this.currentSipSession = sipSession;
-        if (callSession) {
-            this.currentCallSession = callSession;
-        }
-        return promise;
-    }
-    holdSipSession(sipSession_1, callSession_1) {
-        return __awaiter(this, arguments, void 0, function* (sipSession, callSession, withEvent = true) {
-            if (!sipSession) {
-                return new Promise((resolve, reject) => reject(new Error('No session to hold')));
-            }
-            const sessionId = this.getSipSessionId(sipSession);
-            const hasVideo = this.client.hasLocalVideo(sessionId);
-            logger.info('WebRTC hold sip session', {
-                sipId: this.getSipSessionId(sipSession),
-                hasVideo,
-            });
-            // Stop screenshare if needed
-            if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
-                this.lastScreenShare = this.currentScreenShare;
-                yield this.stopScreenSharing(false, callSession);
-            }
-            // Downgrade to audio if needed
-            if (hasVideo) {
-                yield this.client.downgradeToAudio(sipSession);
-            }
-            const isConference = !!callSession && callSession.isConference();
-            const promise = this.client.hold(sipSession, isConference, hasVideo);
-            if (withEvent) {
-                this.eventEmitter.emit(exports.ON_CALL_HELD, this._createCallSession(sipSession, callSession));
-            }
-            return promise;
-        });
-    }
-    unholdSipSession(sipSession_1, callSession_1) {
-        return __awaiter(this, arguments, void 0, function* (sipSession, callSession, withEvent = true) {
-            if (!sipSession) {
-                return new Promise((resolve, reject) => reject(new Error('No session to unhold')));
-            }
-            const isConference = !!callSession && callSession.isConference();
-            const sessionId = this.getSipSessionId(sipSession);
-            logger.info('WebRTC unhold sip session', {
-                sessionId,
-                isConference,
-            });
-            const { hasVideo, } = this.client.getHeldSession(sessionId) || {};
-            const wasScreensharing = this.lastScreenShare && this.lastScreenShare.sipSessionId === sessionId;
-            const wasDesktop = this.lastScreenShare && this.lastScreenShare.desktop;
-            const promise = this.client.unhold(sipSession, isConference);
-            if (hasVideo) {
-                const constraints = {
-                    audio: false,
-                    video: true,
-                    screen: wasScreensharing,
-                    desktop: wasDesktop,
-                };
-                yield this.client.upgradeToVideo(sipSession, constraints, isConference);
-            }
-            const onScreenSharing = (stream) => {
-                const hadVideo = this.lastScreenShare && this.lastScreenShare.hadVideo;
-                if (stream) {
-                    this._onScreenSharing(stream, sipSession, callSession, hadVideo);
-                }
-                this.lastScreenShare = null;
-            };
-            if (withEvent) {
-                // Calling webRtcClient.unhold will unmute the call; let's make sure the call session reflects that state
-                if (callSession) {
-                    callSession.muted = false;
-                }
-                const updatedCallSession = this._createCallSession(sipSession, callSession);
-                // Deprecated event
-                this.eventEmitter.emit(exports.ON_CALL_UNHELD, updatedCallSession);
-                this.eventEmitter.emit(exports.ON_CALL_RESUMED, updatedCallSession);
-            }
-            return promise.then(() => {
-                const stream = callSession ? this.getLocalVideoStream(callSession) : null;
-                if (wasScreensharing) {
-                    onScreenSharing(stream);
-                }
-                return stream;
-            });
-        });
-    }
-    mute(callSession, withEvent = true) {
-        logger.info('WebRTC mute called', {
-            id: callSession ? callSession.getId() : null,
-        });
-        const sipSession = this.findSipSession(callSession);
-        if (!sipSession) {
-            return;
-        }
-        logger.info('WebRTC muting', {
-            sipId: this.getSipSessionId(sipSession),
-        });
-        this.client.mute(sipSession);
-        const newCallSession = this._createCallSession(sipSession, callSession, {
-            muted: true,
-        });
-        if (this.currentCallSession && newCallSession.is(this.currentCallSession)) {
-            this.currentCallSession = newCallSession;
-        }
-        if (withEvent) {
-            this.eventEmitter.emit(exports.ON_CALL_MUTED, newCallSession);
-        }
-    }
-    unmute(callSession, withEvent = true) {
-        logger.info('WebRTC unmute called', {
-            id: callSession ? callSession.getId() : null,
-        });
-        const sipSession = this.findSipSession(callSession);
-        if (!sipSession) {
-            return;
-        }
-        logger.info('WebRTC unmuting', {
-            sipId: this.getSipSessionId(sipSession),
-        });
-        this.client.unmute(sipSession);
-        const newCallSession = this._createCallSession(sipSession, callSession, {
-            muted: false,
-        });
-        if (this.currentCallSession && newCallSession.is(this.currentCallSession)) {
-            this.currentCallSession = newCallSession;
-        }
-        if (withEvent) {
-            this.eventEmitter.emit(exports.ON_CALL_UNMUTED, newCallSession);
-        }
-    }
-    isAudioMuted(callSession) {
-        const sipSession = this.findSipSession(callSession);
-        if (!sipSession) {
-            return true;
-        }
-        return this.client.isAudioMuted(sipSession);
-    }
-    turnCameraOn(callSession) {
-        const sipSession = this.findSipSession(callSession);
-        if (!sipSession) {
-            return;
-        }
-        logger.info('WebRTC turn camera on', {
-            sipId: this.getSipSessionId(sipSession),
-        });
-        this.client.toggleCameraOn(sipSession);
-        this.eventEmitter.emit(exports.ON_CAMERA_RESUMED, this._createCameraResumedCallSession(sipSession, callSession));
-    }
-    turnCameraOff(callSession) {
-        const sipSession = this.findSipSession(callSession);
-        if (!sipSession) {
-            return;
-        }
-        logger.info('WebRTC turn camera off', {
-            sipId: this.getSipSessionId(sipSession),
-        });
-        this.client.toggleCameraOff(sipSession);
-        this.eventEmitter.emit(exports.ON_CAMERA_DISABLED, this._createCameraDisabledCallSession(sipSession, callSession));
-    }
-    sendKey(callSession, tone) {
-        const sipSession = this.findSipSession(callSession);
-        if (!sipSession) {
-            return;
-        }
-        logger.info('WebRTC send key', {
-            sipId: this.getSipSessionId(sipSession),
-            tone,
-        });
-        this.client.sendDTMF(sipSession, tone);
-    }
-    // Should be async to match CTIPhone definition
-    // @TODO: line is not used here
-    makeCall(number_1, line_1, cameraEnabled_1) {
-        return __awaiter(this, arguments, void 0, function* (number, line, cameraEnabled, audioOnly = false, conference = false, options = {}) {
-            logger.info('make WebRTC call', {
-                number,
-                lineId: line ? line.id : null,
-                cameraEnabled,
-                audioOnly,
-                conference,
-                options,
-            });
-            if (!number) {
-                return Promise.resolve(null);
-            }
-            if (!this.client.isRegistered()) {
-                yield this.client.register();
-            }
-            if (this.currentSipSession) {
-                this.holdSipSession(this.currentSipSession, this.currentCallSession, true).catch(e => {
-                    logger.warn('Unable to hold current session when making another call', e);
-                });
-            }
-            let sipSession;
-            try {
-                sipSession = this.client.call(number, this.allowVideo ? cameraEnabled : false, audioOnly, conference, options);
-                this._bindEvents(sipSession);
-            }
-            catch (error) {
-                logger.warn('make WebRTC call, error', {
-                    message: error.message,
-                    stack: error.stack,
-                });
-                return Promise.resolve(null);
-            }
-            const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
-            callSession.setIsConference(conference);
-            callSession.creationTime = new Date();
-            this._updateCallSession(callSession);
-            this.eventEmitter.emit(exports.ON_PLAY_PROGRESS_SOUND, this.audioOutputDeviceId, this.audioOutputVolume);
-            this.currentSipSession = sipSession;
-            this.currentCallSession = callSession;
-            this.eventEmitter.emit(exports.ON_CALL_OUTGOING, callSession);
-            // If an invite promise exists, catch exceptions on it to trigger error like OverConstraintsError.
-            // @ts-ignore: does not exist
-            if (sipSession.invitePromise) {
-                // @ts-ignore: does not exist
-                sipSession.invitePromise.catch(error => {
-                    this.eventEmitter.emit(exports.ON_CALL_ERROR, error, callSession);
-                });
-            }
-            return Promise.resolve(callSession);
-        });
-    }
-    transfer(callSession, target) {
-        const sipSession = this.findSipSession(callSession);
-        if (!sipSession) {
-            return;
-        }
-        logger.info('WebRTC transfer', {
-            sipId: this.getSipSessionId(sipSession),
-            target,
-        });
-        this.client.transfer(sipSession, target);
-    }
-    indirectTransfer(source, destination) {
-        return __awaiter(this, void 0, void 0, function* () {
-            const sipSession = this.client.getSipSession(source.sipCallId);
-            const sipSessionTarget = this.client.getSipSession(destination.sipCallId);
-            logger.info('WebRTC indirect transfer', {
-                sipId: sipSession ? this.getSipSessionId(sipSession) : null,
-                target: sipSessionTarget ? this.getSipSessionId(sipSessionTarget) : null,
-            });
-            if (!sipSessionTarget) {
-                return Promise.reject();
-            }
-            return new Promise(resolve => {
-                const options = {
-                    requestDelegate: {
-                        onAccept: () => {
-                            // Wait a little so the REFER will take effect
-                            setTimeout(() => {
-                                this.hangup(destination).then(resolve);
-                            }, 200);
-                        },
-                    },
-                };
-                return sipSessionTarget.refer(sipSession, options);
-            });
-        });
-    }
-    initiateCTIIndirectTransfer() { return Promise.resolve(null); }
-    cancelCTIIndirectTransfer() { }
-    confirmCTIIndirectTransfer() { }
-    hangup(callSession) {
-        return __awaiter(this, void 0, void 0, function* () {
-            const sipSession = this.findSipSession(callSession);
-            if (!sipSession) {
-                console.error('Call is unknown to the WebRTC phone', callSession ? callSession.sipCallId : null, callSession ? callSession.callId : null);
-                return false;
-            }
-            const sipSessionId = this.getSipSessionId(sipSession);
-            logger.info('WebRTC hangup', {
-                sipId: this.getSipSessionId(sipSession),
-                sipSessionId,
-            });
-            yield this.client.hangup(sipSession);
-            if (callSession) {
-                // Removal of `this.callSessions` will be done in `_onCallTerminated`.
-                this.endCurrentCall(callSession);
-            }
-            else if (sipSessionId) {
-                delete this.callSessions[sipSessionId];
-            }
-            this.shouldSendReinvite = false;
-            return true;
-        });
-    }
-    getStats(callSession) {
-        return __awaiter(this, void 0, void 0, function* () {
-            const sipSession = this.findSipSession(callSession);
-            return sipSession ? this.client.getStats(sipSession) : null;
-        });
-    }
-    startNetworkMonitoring(callSession, interval = 1000) {
-        const sipSession = this.findSipSession(callSession);
-        return sipSession ? this.client.startNetworkMonitoring(sipSession, interval) : null;
-    }
-    stopNetworkMonitoring(callSession) {
-        const sipSession = this.findSipSession(callSession);
-        return sipSession ? this.client.stopNetworkMonitoring(sipSession) : null;
-    }
-    forceCancel(sipSession) {
-        // @ts-ignore: private
-        if (!sipSession || !sipSession.outgoingInviteRequest) {
-            return;
-        }
-        try {
-            // @ts-ignore: private
-            sipSession.outgoingInviteRequest.cancel();
-        }
-        catch (e) {
-            logger.info('force cancel, error', e);
-        }
-    }
-    endCurrentCall(callSession) {
-        const isCurrent = this.isCurrentCallSipSession(callSession);
-        logger.info('Ending current call', {
-            id: callSession ? callSession.getId() : null,
-            isCurrent,
-        });
-        if (isCurrent) {
-            this.currentSipSession = undefined;
-            this.currentCallSession = undefined;
-        }
-        this.eventEmitter.emit(exports.ON_TERMINATE_SOUND, callSession, 'locally ended');
-    }
-    onConnectionMade() { }
-    isConference(callSession) {
-        if (!callSession) {
-            return false;
-        }
-        return this.client && this.client.isConference(callSession.sipCallId);
-    }
-    close() {
-        return __awaiter(this, arguments, void 0, function* (force = false) {
-            logger.info('WebRTC close');
-            try {
-                yield Promise.race([
-                    this.unregister(),
-                    new Promise((resolve, reject) => setTimeout(() => reject(new Error('Unregister, timed out')), 3000)),
-                ]);
-            }
-            catch (e) {
-                logger.error('WebRTC close, unregister error', e);
-            }
-            yield this.client.close(force);
-            this.unbind();
-            this.incomingSessions = [];
-            this.currentSipSession = undefined;
-            this.currentCallSession = undefined;
-            this.shouldSendReinvite = false;
-            this.rejectedSessions = {};
-        });
-    }
-    isRegistered() {
-        return this.client && this.client.isRegistered();
-    }
-    enableRinging() {
-        logger.info('WebRTC enable ringing');
-        this.ringingEnabled = true;
-    }
-    disableRinging() {
-        logger.info('WebRTC disable ringing');
-        this.ringingEnabled = false;
-    }
-    getCurrentCallSession() {
-        if (!this.currentSipSession) {
-            return null;
-        }
-        return this._createCallSession(this.currentSipSession, this.currentCallSession);
-    }
-    hasIncomingCallSession() {
-        return this.incomingSessions.length > 0;
-    }
-    getIncomingCallSession() {
-        if (!this.hasIncomingCallSession()) {
-            return null;
-        }
-        const sessionId = this.incomingSessions[0];
-        const sipSession = this.client.getSipSession(sessionId);
-        if (!sipSession) {
-            return null;
-        }
-        return this._createCallSession(sipSession);
-    }
-    // eslint-disable-next-line @typescript-eslint/default-param-last
-    sendMessage(sipSession = null, body, contentType = 'text/plain') {
-        return this.client.sendMessage(sipSession, body, contentType);
-    }
-    getLocalStream(callSession) {
-        return callSession ? this.client.getLocalStream(callSession.sipCallId) : null;
-    }
-    getLocalVideoStream(callSession) {
-        return callSession ? this.client.getLocalVideoStream(callSession.sipCallId) : null;
-    }
-    // Deprecated
-    getLocalMediaStream(callSession) {
-        logger.warn('WebRTCPhone.getLocalMediaStream is deprecated, use WebRTCPhone.getLocalStream instead');
-        return this.getLocalStream(callSession);
-    }
-    hasLocalVideo(callSession) {
-        return callSession ? this.client.hasLocalVideo(callSession.sipCallId) : false;
-    }
-    hasALocalVideoTrack(callSession) {
-        return callSession ? this.client.hasALocalVideoTrack(callSession.sipCallId) : false;
-    }
-    getRemoteStream(callSession) {
-        return callSession ? this.client.getRemoteStream(callSession.sipCallId) : null;
-    }
-    getRemoteVideoStream(callSession) {
-        return callSession ? this.client.getRemoteVideoStream(callSession.sipCallId) : null;
-    }
-    getRemoteVideoStreamFromPc(callSession) {
-        return callSession ? this.client.getRemoteVideoStreamFromPc(callSession.sipCallId) : null;
-    }
-    hasRemoteVideo(callSession) {
-        return callSession ? this.client.hasRemoteVideo(callSession.sipCallId) : false;
-    }
-    setMediaConstraints(media) {
-        this.client.setMediaConstraints(media);
-    }
-    isVideoRemotelyHeld(callSession) {
-        return callSession ? this.client.isVideoRemotelyHeld(callSession.sipCallId) : false;
-    }
-    bindClientEvents() {
-        this.client.unbind();
-        this.client.on(this.client.INVITE, (sipSession, wantsToDoVideo) => {
-            const autoAnswer = sipSession.request.getHeader('Answer-Mode') === 'Auto';
-            const withVideo = this.allowVideo ? wantsToDoVideo : false;
-            logger.info('WebRTC invite received', {
-                sipId: this.getSipSessionId(sipSession),
-                withVideo,
-                autoAnswer,
-            });
-            const callSession = this._createIncomingCallSession(sipSession, withVideo, null, autoAnswer);
-            this.incomingSessions.push(callSession.getId());
-            this._bindEvents(sipSession);
-            this.client.storeSipSession(sipSession);
-            callSession.creationTime = new Date();
-            this._updateCallSession(callSession);
-            this.eventEmitter.emit(exports.ON_CALL_INCOMING, callSession, wantsToDoVideo);
-            if (!this.currentSipSession) {
-                if (this.ringingEnabled) {
-                    this.eventEmitter.emit(exports.ON_PLAY_RING_SOUND, this.audioRingDeviceId, this.audioRingVolume, callSession);
-                }
-            }
-            else {
-                this.eventEmitter.emit(exports.ON_PLAY_INBOUND_CALL_SIGNAL_SOUND, this.audioOutputDeviceId, this.audioOutputVolume, callSession);
-            }
-        });
-        this.client.on(this.client.ON_REINVITE, (...args) => {
-            logger.info('WebRTC on reinvite', {
-                sessionId: args[0].id,
-                inviteId: args[1].id,
-                updatedCalleeName: args[2],
-                updatedNumber: args[3],
-            });
-            const sipSession = args[0];
-            // Give some time for the stream to be mounted.
-            // if `_createCallSession` is called too soon `hasVideo` will return false because the stream doesn't exists yet
-            setTimeout(() => {
-                // Update callSession
-                this._createCallSession(sipSession, this.callSessions[this.getSipSessionId(sipSession)]);
-                this.eventEmitter.emit.apply(this.eventEmitter, [this.client.ON_REINVITE, ...args]);
-            }, 2000);
-        });
-        this.client.on(this.client.ACCEPTED, (sipSession) => __awaiter(this, void 0, void 0, function* () {
-            const sessionId = this.getSipSessionId(sipSession);
-            const hasSession = (sessionId in this.callSessions);
-            logger.info('WebRTC call accepted', {
-                sipId: sessionId,
-                hasSession,
-            });
-            if (!hasSession) {
-                logger.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', {
-                    sessionId,
-                });
-                return;
-            }
-            this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
-            if (this.audioOutputDeviceId) {
-                yield this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
-            }
-        }));
-        this.client.on('ended', () => { });
-        this.client.on(this.client.ON_ERROR, e => {
-            logger.error('WebRTC error', e);
-            this.eventEmitter.emit(exports.ON_CALL_ERROR, e);
-        });
-        this.client.on(this.client.REJECTED, session => {
-            logger.info('WebRTC call rejected', { id: this.getSipSessionId(session) });
-            const callSession = this._createCallSession(session);
-            // `_createCallSession` add the CallSession in `this.callSessions`.
-            // so we have to remove it or `callCount` will always return 1 even if the call was rejected
-            delete this.callSessions[this.getSipSessionId(session)];
-            callSession.endTime = new Date();
-            this.eventEmitter.emit(exports.ON_CALL_REJECTED, callSession);
-        });
-        this.client.on(this.client.UNREGISTERED, () => {
-            logger.info('WebRTC unregistered');
-            this.eventEmitter.emit(exports.ON_UNREGISTERED);
-        });
-        this.client.on(this.client.ON_DISCONNECTED, () => {
-            logger.info('WebRTC disconnected');
-            this.eventEmitter.emit(exports.ON_DISCONNECTED);
-        });
-        this.client.on(this.client.ON_PROGRESS, session => {
-            logger.info('WebRTC progress (180)');
-            const callSession = this._createCallSession(session, null, {
-                incoming: false,
-                ringing: true,
-                callId: this.getSipSessionId(session),
-            });
-            this.eventEmitter.emit(exports.ON_PROGRESS, callSession, this.audioOutputDeviceId, this.audioOutputVolume);
-        });
-        this.client.on(this.client.ON_EARLY_MEDIA, session => {
-            logger.info('WebRTC early media');
-            const callSession = this._createCallSession(session, null, {
-                incoming: false,
-                ringing: true,
-                callId: this.getSipSessionId(session),
-            });
-            this.eventEmitter.emit(exports.ON_EARLY_MEDIA, callSession, this.audioOutputDeviceId, this.audioOutputVolume);
-        });
-        this.client.on(this.client.REGISTERED, () => {
-            var _a, _b;
-            logger.info('WebRTC registered', {
-                shouldSendReinvite: this.shouldSendReinvite,
-                state: this.currentSipSession ? this.currentSipSession.state : null,
-                currentSipSession: !!this.currentSipSession,
-            });
-            this.stopHeartbeat();
-            this.eventEmitter.emit(exports.ON_REGISTERED);
-            // If the phone registered with a current callSession (eg: when switching network):
-            // send a reinvite to renegociate ICE with new IP
-            if (this.shouldSendReinvite && this.currentSipSession && this.currentSipSession.state === session_state_1.SessionState.Established) {
-                this.shouldSendReinvite = false;
-                const pc = (_b = (_a = this.currentSipSession) === null || _a === void 0 ? void 0 : _a.sessionDescriptionHandler) === null || _b === void 0 ? void 0 : _b.peerConnection;
-                const isConference = pc ? pc.sfu : false;
-                const hasVideo = this.currentCallSession && this.currentCallSession.cameraEnabled;
-                try {
-                    // Send reinvite with iceRestart
-                    this.sendReinvite(this.currentCallSession, null, isConference, !hasVideo, true);
-                }
-                catch (e) {
-                    logger.error('WebRTC reinvite after register, error', {
-                        message: e.message,
-                        stack: e.stack,
-                    });
-                }
-            }
-        });
-        this.client.on(this.client.CONNECTED, () => {
-            logger.info('WebRTC client connected');
-            this.stopHeartbeat();
-        });
-        this.client.on(this.client.DISCONNECTED, () => {
-            logger.info('WebRTC client disconnected');
-            this.eventEmitter.emit(exports.ON_UNREGISTERED);
-            // Do not trigger heartbeat if already running
-            if (!this.client.hasHeartbeat()) {
-                this.startHeartbeat();
-            }
-            // Tell to send reinvite when reconnecting
-            this.shouldSendReinvite = true;
-        });
-        this.client.on(this.client.ON_TRACK, (session, event) => {
-            this.eventEmitter.emit(exports.ON_TRACK, session, event);
-        });
-        this.client.on('onVideoInputChange', stream => {
-            this.eventEmitter.emit(exports.ON_VIDEO_INPUT_CHANGE, stream);
-        });
-        this.client.on(this.client.MESSAGE, (message) => {
-            this._onMessage(message);
-            this.eventEmitter.emit(exports.ON_MESSAGE, message);
-        });
-        this.client.on(this.client.ON_NETWORK_STATS, (session, stats, previousStats) => {
-            const callSession = this.getCallSession(this.getSipSessionId(session));
-            this.eventEmitter.emit(exports.ON_NETWORK_STATS, callSession, stats, previousStats);
-        });
-        // Used when upgrading directly in screenshare mode
-        this.client.on(this.client.ON_SCREEN_SHARING_REINVITE, (sipSession, response, desktop) => {
-            const sipSessionId = this.getSipSessionId(sipSession);
-            const localStream = this.client.getLocalStream(sipSessionId);
-            const callSession = this.callSessions[sipSessionId];
-            logger.info('Updrading directly in screensharing mode', {
-                sipSessionId,
-                tracks: (localStream === null || localStream === void 0 ? void 0 : localStream.getTracks()) || null,
-            });
-            this._onScreenSharing(localStream, sipSession, callSession, false, desktop);
-        });
-    }
-    // Find a corresponding sipSession from a CallSession
-    findSipSession(callSession) {
-        const keys = this.client.getSipSessionIds();
-        const keyIndex = keys.findIndex(sessionId => callSession && callSession.isId(sessionId));
-        if (keyIndex === -1) {
-            const currentSipSessionId = this.currentSipSession ? this.getSipSessionId(this.currentSipSession) : this.client.getSipSessionIds()[0];
-            return currentSipSessionId ? this.client.getSipSession(currentSipSessionId) : null;
-        }
-        return this.client.getSipSession(keys[keyIndex]);
-    }
-    getCallSession(sipSessionId) {
-        return this.callSessions[sipSessionId];
-    }
-    getSipSessionId(sipSession) {
-        if (!sipSession) {
-            return '';
-        }
-        return this.client.getSipSessionId(sipSession);
-    }
-    getSipSessionIdFromCallSession(callSession) {
-        const sipSession = this.findSipSession(callSession);
-        return sipSession ? this.getSipSessionId(sipSession) : null;
-    }
-    _updateCallSession(callSession) {
-        const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
-        if (sipSessionId) {
-            this.callSessions[sipSessionId] = callSession;
-        }
-    }
-    _onMessage(message) {
-        if (!message || message.method !== 'MESSAGE') {
-            return;
-        }
-        let body;
-        try {
-            body = JSON.parse(message.body || '');
-        }
-        catch (e) {
-            return;
-        }
-        const { type, content, } = body;
-        switch (type) {
-            case exports.MESSAGE_TYPE_CHAT:
-                this.eventEmitter.emit(exports.ON_CHAT, content);
-                break;
-            case exports.MESSAGE_TYPE_SIGNAL:
-                {
-                    this.eventEmitter.emit(exports.ON_SIGNAL, content);
-                    break;
-                }
-            default:
-        }
-    }
-    _createIncomingCallSession(sipSession, cameraEnabled, fromSession, autoAnswer = false) {
-        return this._createCallSession(sipSession, fromSession, {
-            incoming: true,
-            ringing: true,
-            cameraEnabled,
-            autoAnswer,
-        });
-    }
-    _createOutgoingCallSession(sipSession, cameraEnabled, fromSession) {
-        return this._createCallSession(sipSession, fromSession, {
-            incoming: false,
-            ringing: true,
-            cameraEnabled,
-        });
-    }
-    _createAcceptedCallSession(sipSession, cameraEnabled, fromSession) {
-        return this._createCallSession(sipSession, fromSession, {
-            cameraEnabled: cameraEnabled !== undefined ? cameraEnabled : false,
-        });
-    }
-    _createMutedCallSession(sipSession, fromSession) {
-        return this._createCallSession(sipSession, fromSession, {
-            muted: true,
-        });
-    }
-    _createUnmutedCallSession(sipSession, fromSession) {
-        return this._createCallSession(sipSession, fromSession, {
-            muted: false,
-        });
-    }
-    _createCameraResumedCallSession(sipSession, fromSession) {
-        return this._createCallSession(sipSession, fromSession, {
-            videoMuted: false,
-        });
-    }
-    _createCameraDisabledCallSession(sipSession, fromSession) {
-        return this._createCallSession(sipSession, fromSession, {
-            videoMuted: true,
-        });
-    }
-    _createCallSession(sipSession, fromSession, extra = {}) {
-        // eslint-disable-next-line
-        const identity = sipSession ? sipSession.remoteIdentity || sipSession.assertedIdentity : null;
-        // @ts-ignore: private
-        const number = identity ? identity.uri._normal.user : null;
-        const { state, } = sipSession || {};
-        const sessionId = this.getSipSessionId(sipSession);
-        fromSession = fromSession || this.callSessions[sessionId];
-        const callSession = new CallSession_1.default(Object.assign({ callId: fromSession && fromSession.callId, sipCallId: sessionId, sipStatus: state, displayName: identity ? identity.displayName || number : number, 
-            // @ts-ignore: date / number
-            startTime: fromSession ? fromSession.startTime : new Date(), creationTime: fromSession ? fromSession.creationTime : null, answerTime: fromSession ? fromSession.answerTime : null, endTime: fromSession ? fromSession.endTime : null, answered: state === session_state_1.SessionState.Established, paused: this.client.isCallHeld(sipSession), isCaller: fromSession ? fromSession.isCaller : 'incoming' in extra ? !extra.incoming : false, cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId), number, ringing: false, muted: fromSession ? fromSession.isMuted() : this.client.isAudioMuted(sipSession), videoMuted: fromSession ? fromSession.isVideoMuted() : false, recording: fromSession ? fromSession.isRecording() : false, recordingPaused: false, sipSession, conference: !!fromSession && fromSession.isConference() }, extra));
-        this.callSessions[sessionId] = callSession;
-        return callSession;
-    }
-}
-exports.default = WebRTCPhone;