@wazo/sdk 0.43.15 → 0.44.0

package/src/domain/Phone/WebRTCPhone.ts

@@ -1,1830 +0,0 @@
-/* eslint-disable no-param-reassign */
-/* eslint-disable no-underscore-dangle */
-import type { Message } from 'sip.js/lib/api/message';
-import { SessionState } from 'sip.js/lib/api/session-state';
-import type { IncomingRequestMessage } from 'sip.js/lib/core/messages/incoming-request-message';
-import { OutgoingInviteRequest } from 'sip.js/lib/core';
-import { Inviter, Invitation, Session, SessionDescriptionHandlerOptions } from 'sip.js/lib/api';
-import { SessionDescriptionHandler } from 'sip.js/lib/platform/web';
-import CallSession from '../CallSession';
-import type { Phone, AvailablePhoneOptions } from './Phone';
-import WebRTCClient from '../../web-rtc-client';
-import Emitter from '../../utils/Emitter';
-import IssueReporter from '../../service/IssueReporter';
-import { PeerConnection, WazoSession } from '../types';
-
-export const ON_USER_AGENT = 'onUserAgent';
-export const ON_REGISTERED = 'onRegistered';
-export const ON_UNREGISTERED = 'onUnRegistered';
-export const ON_PROGRESS = 'onProgress';
-export const ON_CALL_ACCEPTED = 'onCallAccepted';
-export const ON_CALL_ANSWERED = 'onCallAnswered';
-export const ON_CALL_INCOMING = 'onCallIncoming';
-export const ON_CALL_OUTGOING = 'onCallOutgoing';
-export const ON_CALL_MUTED = 'onCallMuted';
-export const ON_CALL_UNMUTED = 'onCallUnmuted';
-export const ON_CALL_RESUMED = 'onCallResumed';
-export const ON_CALL_HELD = 'onCallHeld';
-export const ON_CALL_UNHELD = 'onCallUnHeld';
-export const ON_CAMERA_DISABLED = 'onCameraDisabled';
-export const ON_CAMERA_RESUMED = 'onCameraResumed';
-export const ON_CALL_CANCELED = 'onCallCanceled';
-export const ON_CALL_FAILED = 'onCallFailed';
-export const ON_CALL_REJECTED = 'onCallRejected';
-export const ON_CALL_ENDED = 'onCallEnded';
-export const ON_CALL_ENDING = 'onCallEnding';
-export const ON_MESSAGE = 'onMessage';
-export const ON_REINVITE = 'reinvite';
-export const ON_TRACK = 'onTrack';
-export const ON_AUDIO_STREAM = 'onAudioStream';
-export const ON_VIDEO_STREAM = 'onVideoStream';
-export const ON_REMOVE_STREAM = 'onRemoveStream';
-export const ON_SHARE_SCREEN_STARTED = 'onScreenShareStarted';
-export const ON_SHARE_SCREEN_ENDING = 'onScreenShareEnding';
-export const ON_SHARE_SCREEN_ENDED = 'onScreenShareEnded';
-export const ON_TERMINATE_SOUND = 'terminateSound';
-export const ON_PLAY_RING_SOUND = 'playRingingSound';
-export const ON_PLAY_INBOUND_CALL_SIGNAL_SOUND = 'playInboundCallSignalSound';
-export const ON_PLAY_HANGUP_SOUND = 'playHangupSound';
-export const ON_PLAY_PROGRESS_SOUND = 'playProgressSound';
-export const ON_VIDEO_INPUT_CHANGE = 'videoInputChange';
-export const ON_CALL_ERROR = 'onCallError';
-export const ON_MESSAGE_TRACK_UPDATED = 'onTrackUpdated';
-export const ON_NETWORK_STATS = 'onNetworkStats';
-export const ON_CHAT = 'phone/ON_CHAT';
-export const ON_SIGNAL = 'phone/ON_SIGNAL';
-export const ON_DISCONNECTED = 'onDisconnected';
-export const ON_EARLY_MEDIA = 'onEarlyMedia';
-export const MESSAGE_TYPE_CHAT = 'message/TYPE_CHAT';
-export const MESSAGE_TYPE_SIGNAL = 'message/TYPE_SIGNAL';
-export const events = [ON_USER_AGENT, ON_REGISTERED, ON_UNREGISTERED, ON_PROGRESS, ON_CALL_ACCEPTED, ON_CALL_ANSWERED, ON_CALL_INCOMING, ON_CALL_OUTGOING, ON_CALL_MUTED, ON_CALL_UNMUTED, ON_CALL_RESUMED, ON_CALL_HELD, ON_CALL_UNHELD, ON_CAMERA_DISABLED, ON_CALL_FAILED, ON_CALL_ENDED, ON_CALL_REJECTED, ON_MESSAGE, ON_REINVITE, ON_TRACK, ON_AUDIO_STREAM, ON_VIDEO_STREAM, ON_REMOVE_STREAM, ON_SHARE_SCREEN_ENDED, ON_TERMINATE_SOUND, ON_PLAY_RING_SOUND, ON_PLAY_INBOUND_CALL_SIGNAL_SOUND, ON_PLAY_HANGUP_SOUND, ON_PLAY_PROGRESS_SOUND, ON_VIDEO_INPUT_CHANGE, ON_SHARE_SCREEN_STARTED, ON_CALL_ERROR, ON_CHAT, ON_SIGNAL, ON_NETWORK_STATS, ON_DISCONNECTED, ON_EARLY_MEDIA];
-const logger = IssueReporter.loggerFor('webrtc-phone');
-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;
-
-  constructor(client: WebRTCClient, audioOutputDeviceId: string | undefined, allowVideo = false, audioRingDeviceId?: string) {
-    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: string) {
-    this.incomingSessions = this.incomingSessions.filter(sessionId => sessionId !== id);
-  }
-
-  isWebRTC() {
-    return true;
-  }
-
-  async sendReinvite(callSession?: CallSession, constraints: Record<string, any> | null | undefined = null, conference = false, audioOnly = false, iceRestart = false): Promise<OutgoingInviteRequest | void> {
-    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?.screen;
-    const isUpgrade = shouldScreenShare || constraints?.video;
-
-    // Downgrade
-    if (constraints && !isUpgrade) {
-      // No reinvite needed
-      this._downgradeToAudio(callSession);
-
-      return;
-    }
-
-    if (isUpgrade) {
-      const shouldReinvite = await this._upgradeToVideo(callSession, constraints, conference);
-
-      if (!shouldReinvite) {
-        return;
-      }
-    }
-
-    return this.client.reinvite(sipSession, constraints, conference, audioOnly, iceRestart);
-  }
-
-  getUserAgent() {
-    return this.client?.config?.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: (...args: Array<any>) => any) {
-    this.client.setOnHeartbeatTimeout(cb);
-  }
-
-  setOnHeartbeatCallback(cb: (...args: Array<any>) => any) {
-    this.client.setOnHeartbeatCallback(cb);
-  }
-
-  reconnect() {
-    logger.info('WebRTC phone - reconnect', {
-      client: !!this.client,
-    });
-
-    if (!this.client) {
-      return;
-    }
-
-    this.client.attemptReconnection();
-  }
-
-  getOptions(): AvailablePhoneOptions {
-    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?: 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
-  async _upgradeToVideo(callSession: CallSession | undefined, constraints: Record<string, any>, isConference: boolean): Promise<boolean> {
-    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 as SessionDescriptionHandlerOptions & { audioOnly: boolean })?.audioOnly;
-    const newStream = await 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: CallSession | undefined, isUpgrade: boolean) {
-    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: MESSAGE_TYPE_SIGNAL,
-        content: {
-          type: 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: WazoSession) {
-    const sipSessionId = this.getSipSessionId(sipSession);
-
-    if (sipSession instanceof Invitation) {
-      // Monkey patch to know when canceled with the CANCEL message
-      const onCancel = sipSession._onCancel.bind(sipSession);
-
-      sipSession._onCancel = (message: IncomingRequestMessage) => {
-        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(ON_CALL_CANCELED, callSession, elsewhere);
-      };
-    }
-
-    sipSession.stateChange.addListener((newState: SessionState) => {
-      switch (newState) {
-        case SessionState.Terminating:
-          logger.info('WebRTC phone - call terminating', {
-            sipId: this.getSipSessionId(sipSession),
-          });
-          this.eventEmitter.emit(ON_CALL_ENDING, this._createCallSession(sipSession));
-          break;
-
-        case 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(ON_CALL_ENDED, callSession, wasCurrentSession);
-        }
-
-        default:
-          break;
-      }
-    });
-
-    if (!sipSession.sessionDescriptionHandler) {
-      return;
-    }
-
-    // Video events
-    const sdh = sipSession.sessionDescriptionHandler as SessionDescriptionHandler;
-    const peerConnection = sdh.peerConnection as PeerConnection;
-
-    if (!peerConnection) {
-      logger.info('null peer connection');
-      return;
-    }
-
-    peerConnection.ontrack = (rawEvent: RTCTrackEvent) => {
-      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(ON_AUDIO_STREAM, stream);
-      }
-
-      return this.eventEmitter.emit(ON_VIDEO_STREAM, stream, event.track.id, event, sipSession);
-    };
-
-    peerConnection.onremovestream = (event: any) => {
-      logger.info('on remove stream called on the peer connection', {
-        id: event.stream.id,
-        tracks: event.stream.getTracks(),
-      });
-      this.eventEmitter.emit(ON_REMOVE_STREAM, event.stream);
-    };
-  }
-
-  async startScreenSharing(constraints: Record<string, any>, callSession?: CallSession): Promise<MediaStream> {
-    logger.info('WebRTC - start screen sharing', {
-      constraints,
-      id: callSession ? callSession.getId() : null,
-    });
-    const screenShareStream = await 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 as WazoSession, callSession, hadVideo, constraints.desktop);
-
-    return screenShareStream;
-  }
-
-  async stopScreenSharing(restoreLocalStream = true, callSession?: CallSession): Promise<OutgoingInviteRequest | void | null> {
-    if (!this.currentScreenShare) {
-      return;
-    }
-
-    logger.info('WebRTC phone - stop screen sharing', {
-      restoreLocalStream,
-    });
-    let reinvited: OutgoingInviteRequest | null | void = null;
-
-    try {
-      if (this.currentScreenShare.stream) {
-        this.currentScreenShare.stream.getTracks()
-          .filter((track: MediaStreamTrack) => track.enabled && track.kind === 'video')
-          .forEach((track: MediaStreamTrack) => 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`.
-        await new Promise(resolve => setTimeout(resolve, 500));
-        reinvited = await this.sendReinvite(targetCallSession, constraints, conference);
-      }
-    } catch (e: any) {
-      console.warn(e);
-    }
-
-    const sipSession = this.currentSipSession;
-    this.eventEmitter.emit(ON_SHARE_SCREEN_ENDED, callSession && sipSession ? this._createCallSession(sipSession, callSession, {
-      screensharing: false,
-    }) : null);
-    this.currentScreenShare = null;
-
-    return reinvited;
-  }
-
-  _onScreenSharing(screenStream: MediaStream, sipSession: WazoSession, callSession: CallSession | null | undefined, hadVideo: boolean, desktop?: boolean | null | undefined) {
-    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(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(ON_SHARE_SCREEN_STARTED, this._createCallSession(sipSession, callSession, {
-      screensharing: true,
-    }), screenStream);
-  }
-
-  _onCallAccepted(sipSession: WazoSession, cameraEnabled: boolean): CallSession {
-    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(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(ON_CALL_ACCEPTED, callSession, cameraEnabled);
-    return callSession;
-  }
-
-  async changeAudioDevice(id: string) {
-    logger.info('WebRTC phone - change audio device', {
-      deviceId: id,
-    });
-    this.audioOutputDeviceId = id;
-    return this.client.changeAudioOutputDevice(id);
-  }
-
-  createAudioElementFor(sessionId: string) {
-    return this.client.createAudioElementFor(sessionId);
-  }
-
-  changeRingDevice(id: string) {
-    logger.info('WebRTC phone - changing ring device', {
-      id,
-    });
-    this.audioRingDeviceId = id;
-  }
-
-  // volume is a value between 0 and 1
-  changeAudioVolume(volume: number) {
-    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: number) {
-    logger.info('WebRTC phone - changing ring volume', {
-      volume,
-    });
-    this.audioRingVolume = volume;
-  }
-
-  changeAudioInputDevice(id: string, force?: boolean) {
-    logger.info('WebRTC phone - changeAudio input device', {
-      deviceId: id,
-    });
-    return this.client.changeAudioInputDevice(id, this.currentSipSession, force);
-  }
-
-  async changeVideoInputDevice(id: string): Promise<void | MediaStream | null | undefined> {
-    logger.info('WebRTC phone - change video input device', {
-      deviceId: id,
-    });
-    return this.client.changeVideoInputDevice(id, this.currentSipSession);
-  }
-
-  changeSessionVideoInputDevice(id: string | null | undefined) {
-    logger.info('WebRTC phone - change session video input device', {
-      deviceId: id,
-    });
-    return this.currentSipSession && this.client.changeSessionVideoInputDevice(id, this.currentSipSession);
-  }
-
-  _onCallTerminated(sipSession: WazoSession): boolean {
-    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(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(ON_PLAY_RING_SOUND, this.audioRingDeviceId, this.audioRingVolume, nextCallSession);
-          } else {
-            this.eventEmitter.emit(ON_PLAY_INBOUND_CALL_SIGNAL_SOUND, this.audioOutputDeviceId, this.audioOutputVolume, nextCallSession);
-          }
-        }
-
-        this.eventEmitter.emit(ON_CALL_INCOMING, nextCallSession, nextCallSession?.cameraEnabled);
-      }, 100);
-    }
-
-    if (callSession.getId() in this.ignoredSessions) {
-      return false;
-    }
-
-    // @ts-ignore: does not exist
-    if (!sipSession.isCanceled) {
-      setTimeout(() => {
-        this.eventEmitter.emit(ON_PLAY_HANGUP_SOUND, this.audioOutputDeviceId, this.audioOutputVolume, callSession);
-      }, 10);
-    }
-
-    return isCurrentSession;
-  }
-
-  setActiveSipSession(callSession: 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 as SessionDescriptionHandler;
-    const peerConnection = sdh.peerConnection as PeerConnection;
-    const remoteStream: MediaStream = peerConnection.getRemoteStreams().find((stream: MediaStream) => !!stream.getVideoTracks().length) as MediaStream;
-    return remoteStream && remoteStream.getVideoTracks().some(track => !track.muted);
-  }
-
-  callCount(): number {
-    return Object.keys(this.callSessions).length;
-  }
-
-  isCurrentCallSipSession(callSession: CallSession): boolean {
-    if (!this.currentSipSession) {
-      return false;
-    }
-
-    return this.currentSipSession && this.getSipSessionId(this.currentSipSession) === callSession.getId();
-  }
-
-  hasVideo(callSession: CallSession): boolean {
-    return this.client.hasVideo(callSession.getId());
-  }
-
-  hasAVideoTrack(callSession: CallSession): boolean {
-    return this.client.hasAVideoTrack(callSession.getId());
-  }
-
-  // Deprecated
-  getLocalStreamForCall(callSession: CallSession): MediaStream | null | undefined {
-    logger.warn('WebRTCPhone.getLocalStreamForCall is deprecated, use WebRTCPhone.getLocalStream instead');
-    return this.getLocalStream(callSession);
-  }
-
-  // Deprecated
-  getRemoteStreamForCall(callSession: CallSession): MediaStream | null | undefined {
-    logger.warn('WebRTCPhone.getRemoteStreamForCall is deprecated, use WebRTCPhone.getRemoteStream instead');
-    return this.getRemoteVideoStream(callSession);
-  }
-
-  getRemoteStreamsForCall(callSession: CallSession): MediaStream | null | undefined {
-    logger.warn('WebRTCPhone.getRemoteStreamsForCall is deprecated, use WebRTCPhone.getRemoteStreams instead');
-    return this.getRemoteStream(callSession);
-  }
-
-  accept(callSession: CallSession, cameraEnabled?: boolean): Promise<string | null> {
-    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(ON_CALL_ANSWERED, callSession);
-    const sipSession = this.client.getSipSession(callSession.getId()) as Invitation;
-
-    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(ON_CALL_ERROR, new Error(error), callSession);
-    }
-
-    return Promise.resolve(null);
-  }
-
-  isAccepted(callSession: CallSession): boolean {
-    return callSession && callSession.getId() in this.acceptedSessions;
-  }
-
-  async reject(callSession: CallSession): Promise<void> {
-    logger.info('reject WebRTC called', {
-      id: callSession.getId(),
-    });
-    this.eventEmitter.emit(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);
-    }
-  }
-
-  async ignore(callSession: CallSession): Promise<void> {
-    logger.info('WebRTC ignore', {
-      id: callSession.getId(),
-    });
-    // kill the ring
-    this.eventEmitter.emit(ON_TERMINATE_SOUND, callSession, 'ignoring call');
-    this.ignoredSessions[callSession.getId()] = true;
-    callSession.ignore();
-  }
-
-  atxfer(callSession: CallSession): Record<string, any> | null | undefined {
-    const sipSession = this.findSipSession(callSession) as Inviter;
-
-    if (sipSession) {
-      logger.info('WebRTC atxfer', {
-        sipId: this.getSipSessionId(sipSession),
-      });
-      return this.client.atxfer(sipSession);
-    }
-  }
-
-  hold(callSession: CallSession, withEvent = true): Promise<OutgoingInviteRequest | void> | null | undefined {
-    logger.info('WebRTC hold', {
-      id: callSession.getId(),
-    });
-    const sipSession = this.findSipSession(callSession);
-    return sipSession ? this.holdSipSession(sipSession, callSession, withEvent) : null;
-  }
-
-  // @Deprecated
-  unhold(callSession: CallSession, withEvent = true): Promise<MediaStream | void | null> {
-    console.warn('Please note that `phone.unhold()` is being deprecated; `phone.resume()` is the preferred method');
-    return this.resume(callSession, withEvent);
-  }
-
-  resume(callSession?: CallSession, withEvent = true): Promise<MediaStream | void | null> {
-    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;
-  }
-
-  async holdSipSession(sipSession: WazoSession, callSession?: CallSession, withEvent = true): Promise<OutgoingInviteRequest | void> {
-    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;
-      await this.stopScreenSharing(false, callSession);
-    }
-
-    // Downgrade to audio if needed
-    if (hasVideo) {
-      await this.client.downgradeToAudio(sipSession);
-    }
-
-    const isConference = !!callSession && callSession.isConference();
-    const promise = this.client.hold(sipSession, isConference, hasVideo);
-
-    if (withEvent) {
-      this.eventEmitter.emit(ON_CALL_HELD, this._createCallSession(sipSession, callSession));
-    }
-
-    return promise;
-  }
-
-  async unholdSipSession(sipSession: WazoSession, callSession: CallSession | null | undefined, withEvent = true): Promise<MediaStream | void | null> {
-    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,
-      };
-      await this.client.upgradeToVideo(sipSession, constraints, isConference);
-    }
-
-    const onScreenSharing = (stream: MediaStream | null | undefined) => {
-      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(ON_CALL_UNHELD, updatedCallSession);
-      this.eventEmitter.emit(ON_CALL_RESUMED, updatedCallSession);
-    }
-
-    return promise.then(() => {
-      const stream = callSession ? this.getLocalVideoStream(callSession) : null;
-
-      if (wasScreensharing) {
-        onScreenSharing(stream);
-      }
-
-      return stream;
-    });
-  }
-
-  mute(callSession?: CallSession, withEvent = true): void {
-    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(ON_CALL_MUTED, newCallSession);
-    }
-  }
-
-  unmute(callSession?: CallSession, withEvent = true): void {
-    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(ON_CALL_UNMUTED, newCallSession);
-    }
-  }
-
-  isAudioMuted(callSession: CallSession): boolean {
-    const sipSession = this.findSipSession(callSession);
-
-    if (!sipSession) {
-      return true;
-    }
-
-    return this.client.isAudioMuted(sipSession);
-  }
-
-  turnCameraOn(callSession?: CallSession): void {
-    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(ON_CAMERA_RESUMED, this._createCameraResumedCallSession(sipSession, callSession));
-  }
-
-  turnCameraOff(callSession?: CallSession): void {
-    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(ON_CAMERA_DISABLED, this._createCameraDisabledCallSession(sipSession, callSession));
-  }
-
-  sendKey(callSession: CallSession | undefined, tone: string): void {
-    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
-  async makeCall(number: string, line: any, cameraEnabled?: boolean, audioOnly = false, conference = false, options = {}): Promise<CallSession | null | undefined> {
-    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()) {
-      await 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: WazoSession;
-
-    try {
-      sipSession = this.client.call(number, this.allowVideo ? cameraEnabled : false, audioOnly, conference, options) as WazoSession;
-
-      this._bindEvents(sipSession);
-    } catch (error: any) {
-      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(ON_PLAY_PROGRESS_SOUND, this.audioOutputDeviceId, this.audioOutputVolume);
-    this.currentSipSession = sipSession;
-    this.currentCallSession = callSession;
-    this.eventEmitter.emit(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(ON_CALL_ERROR, error, callSession);
-      });
-    }
-
-    return Promise.resolve(callSession);
-  }
-
-  transfer(callSession: CallSession, target: string): void {
-    const sipSession = this.findSipSession(callSession);
-
-    if (!sipSession) {
-      return;
-    }
-
-    logger.info('WebRTC transfer', {
-      sipId: this.getSipSessionId(sipSession),
-      target,
-    });
-    this.client.transfer(sipSession, target);
-  }
-
-  async indirectTransfer(source: CallSession, destination: CallSession): Promise<boolean> {
-    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 as Session), options);
-    });
-  }
-
-  initiateCTIIndirectTransfer() { return Promise.resolve(null); }
-
-  cancelCTIIndirectTransfer() {}
-
-  confirmCTIIndirectTransfer() {}
-
-  async hangup(callSession?: CallSession): Promise<boolean> {
-    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,
-    });
-    await 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;
-  }
-
-  async getStats(callSession?: CallSession): Promise<RTCStatsReport | null | undefined> {
-    const sipSession = this.findSipSession(callSession);
-    return sipSession ? this.client.getStats(sipSession) : null;
-  }
-
-  startNetworkMonitoring(callSession?: CallSession, interval = 1000) {
-    const sipSession = this.findSipSession(callSession);
-    return sipSession ? this.client.startNetworkMonitoring(sipSession, interval) : null;
-  }
-
-  stopNetworkMonitoring(callSession?: CallSession) {
-    const sipSession = this.findSipSession(callSession);
-    return sipSession ? this.client.stopNetworkMonitoring(sipSession) : null;
-  }
-
-  forceCancel(sipSession: Inviter): void {
-    // @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: CallSession): void {
-    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(ON_TERMINATE_SOUND, callSession, 'locally ended');
-  }
-
-  onConnectionMade(): void {}
-
-  isConference(callSession: CallSession | null | undefined): boolean {
-    if (!callSession) {
-      return false;
-    }
-
-    return this.client && this.client.isConference(callSession.sipCallId);
-  }
-
-  async close(force = false): Promise<void> {
-    logger.info('WebRTC close');
-
-    try {
-      await Promise.race([
-        this.unregister(),
-        new Promise((resolve, reject) => setTimeout(() => reject(new Error('Unregister, timed out')), 3000)),
-      ]);
-    } catch (e: any) {
-      logger.error('WebRTC close, unregister error', e);
-    }
-
-    await this.client.close(force);
-    this.unbind();
-    this.incomingSessions = [];
-    this.currentSipSession = undefined;
-    this.currentCallSession = undefined;
-    this.shouldSendReinvite = false;
-    this.rejectedSessions = {};
-  }
-
-  isRegistered(): boolean {
-    return this.client && this.client.isRegistered();
-  }
-
-  enableRinging(): Promise<void> | void {
-    logger.info('WebRTC enable ringing');
-    this.ringingEnabled = true;
-  }
-
-  disableRinging(): Promise<void> | void {
-    logger.info('WebRTC disable ringing');
-    this.ringingEnabled = false;
-  }
-
-  getCurrentCallSession(): CallSession | null | undefined {
-    if (!this.currentSipSession) {
-      return null;
-    }
-
-    return this._createCallSession(this.currentSipSession, this.currentCallSession);
-  }
-
-  hasIncomingCallSession(): boolean {
-    return this.incomingSessions.length > 0;
-  }
-
-  getIncomingCallSession(): CallSession | null | undefined {
-    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: WazoSession | null = null, body: string, contentType = 'text/plain'): void {
-    return this.client.sendMessage(sipSession, body, contentType);
-  }
-
-  getLocalStream(callSession: CallSession): MediaStream | null | undefined {
-    return callSession ? this.client.getLocalStream(callSession.sipCallId) : null;
-  }
-
-  getLocalVideoStream(callSession: CallSession): MediaStream | null | undefined {
-    return callSession ? this.client.getLocalVideoStream(callSession.sipCallId) : null;
-  }
-
-  // Deprecated
-  getLocalMediaStream(callSession: CallSession): MediaStream | null | undefined {
-    logger.warn('WebRTCPhone.getLocalMediaStream is deprecated, use WebRTCPhone.getLocalStream instead');
-    return this.getLocalStream(callSession);
-  }
-
-  hasLocalVideo(callSession: CallSession): boolean {
-    return callSession ? this.client.hasLocalVideo(callSession.sipCallId) : false;
-  }
-
-  hasALocalVideoTrack(callSession: CallSession): boolean {
-    return callSession ? this.client.hasALocalVideoTrack(callSession.sipCallId) : false;
-  }
-
-  getRemoteStream(callSession: CallSession): MediaStream | null | undefined {
-    return callSession ? this.client.getRemoteStream(callSession.sipCallId) : null;
-  }
-
-  getRemoteVideoStream(callSession: CallSession): MediaStream | null | undefined {
-    return callSession ? this.client.getRemoteVideoStream(callSession.sipCallId) : null;
-  }
-
-  getRemoteVideoStreamFromPc(callSession: CallSession): MediaStream | null | undefined {
-    return callSession ? this.client.getRemoteVideoStreamFromPc(callSession.sipCallId) : null;
-  }
-
-  hasRemoteVideo(callSession: CallSession): boolean {
-    return callSession ? this.client.hasRemoteVideo(callSession.sipCallId) : false;
-  }
-
-  setMediaConstraints(media: MediaStreamConstraints): void {
-    this.client.setMediaConstraints(media);
-  }
-
-  isVideoRemotelyHeld(callSession: CallSession): boolean {
-    return callSession ? this.client.isVideoRemotelyHeld(callSession.sipCallId) : false;
-  }
-
-  bindClientEvents() {
-    this.client.unbind();
-    this.client.on(this.client.INVITE, (sipSession: WazoSession, wantsToDoVideo: boolean) => {
-      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(ON_CALL_INCOMING, callSession, wantsToDoVideo);
-
-      if (!this.currentSipSession) {
-        if (this.ringingEnabled) {
-          this.eventEmitter.emit(ON_PLAY_RING_SOUND, this.audioRingDeviceId, this.audioRingVolume, callSession);
-        }
-      } else {
-        this.eventEmitter.emit(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, async (sipSession: WazoSession) => {
-      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) {
-        await 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(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(ON_CALL_REJECTED, callSession);
-    });
-    this.client.on(this.client.UNREGISTERED, () => {
-      logger.info('WebRTC unregistered');
-      this.eventEmitter.emit(ON_UNREGISTERED);
-    });
-    this.client.on(this.client.ON_DISCONNECTED, () => {
-      logger.info('WebRTC disconnected');
-      this.eventEmitter.emit(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(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(ON_EARLY_MEDIA, callSession, this.audioOutputDeviceId, this.audioOutputVolume);
-    });
-    this.client.on(this.client.REGISTERED, () => {
-      logger.info('WebRTC registered', {
-        shouldSendReinvite: this.shouldSendReinvite,
-        state: this.currentSipSession ? this.currentSipSession.state : null,
-        currentSipSession: !!this.currentSipSession,
-      });
-      this.stopHeartbeat();
-      this.eventEmitter.emit(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 === SessionState.Established) {
-        this.shouldSendReinvite = false;
-        const pc = (this.currentSipSession?.sessionDescriptionHandler as SessionDescriptionHandler)?.peerConnection as 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: any) {
-          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(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(ON_TRACK, session, event);
-    });
-    this.client.on('onVideoInputChange', stream => {
-      this.eventEmitter.emit(ON_VIDEO_INPUT_CHANGE, stream);
-    });
-    this.client.on(this.client.MESSAGE, (message: Message) => {
-      this._onMessage(message);
-
-      this.eventEmitter.emit(ON_MESSAGE, message);
-    });
-    this.client.on(this.client.ON_NETWORK_STATS, (session, stats, previousStats) => {
-      const callSession = this.getCallSession(this.getSipSessionId(session));
-      this.eventEmitter.emit(ON_NETWORK_STATS, callSession, stats, previousStats);
-    });
-    // Used when upgrading directly in screenshare mode
-    this.client.on(this.client.ON_SCREEN_SHARING_REINVITE, (sipSession: WazoSession, response: any, desktop: boolean) => {
-      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?.getTracks() || null,
-      });
-
-      this._onScreenSharing(localStream as MediaStream, sipSession, callSession, false, desktop);
-    });
-  }
-
-  // Find a corresponding sipSession from a CallSession
-  findSipSession(callSession?: CallSession): WazoSession | null | undefined {
-    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: string): CallSession | null | undefined {
-    return this.callSessions[sipSessionId];
-  }
-
-  getSipSessionId(sipSession: WazoSession | null | undefined): string {
-    if (!sipSession) {
-      return '';
-    }
-    return this.client.getSipSessionId(sipSession);
-  }
-
-  getSipSessionIdFromCallSession(callSession: CallSession): string | null {
-    const sipSession = this.findSipSession(callSession);
-    return sipSession ? this.getSipSessionId(sipSession) : null;
-  }
-
-  _updateCallSession(callSession: CallSession): void {
-    const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
-
-    if (sipSessionId) {
-      this.callSessions[sipSessionId] = callSession;
-    }
-  }
-
-  _onMessage(message: Message & { method?: string, body?: string }): void {
-    if (!message || message.method !== 'MESSAGE') {
-      return;
-    }
-
-    let body;
-
-    try {
-      body = JSON.parse(message.body || '');
-    } catch (e: any) {
-      return;
-    }
-
-    const {
-      type,
-      content,
-    } = body;
-
-    switch (type) {
-      case MESSAGE_TYPE_CHAT:
-        this.eventEmitter.emit(ON_CHAT, content);
-        break;
-
-      case MESSAGE_TYPE_SIGNAL:
-      {
-        this.eventEmitter.emit(ON_SIGNAL, content);
-        break;
-      }
-
-      default:
-    }
-  }
-
-  _createIncomingCallSession(sipSession: WazoSession, cameraEnabled: boolean, fromSession?: CallSession | null | undefined, autoAnswer = false): CallSession {
-    return this._createCallSession(sipSession, fromSession, {
-      incoming: true,
-      ringing: true,
-      cameraEnabled,
-      autoAnswer,
-    });
-  }
-
-  _createOutgoingCallSession(sipSession: WazoSession, cameraEnabled: boolean, fromSession?: CallSession): CallSession {
-    return this._createCallSession(sipSession, fromSession, {
-      incoming: false,
-      ringing: true,
-      cameraEnabled,
-    });
-  }
-
-  _createAcceptedCallSession(sipSession: WazoSession, cameraEnabled?: boolean, fromSession?: CallSession): CallSession {
-    return this._createCallSession(sipSession, fromSession, {
-      cameraEnabled: cameraEnabled !== undefined ? cameraEnabled : false,
-    });
-  }
-
-  _createMutedCallSession(sipSession: WazoSession, fromSession?: CallSession): CallSession {
-    return this._createCallSession(sipSession, fromSession, {
-      muted: true,
-    });
-  }
-
-  _createUnmutedCallSession(sipSession: WazoSession, fromSession?: CallSession): CallSession {
-    return this._createCallSession(sipSession, fromSession, {
-      muted: false,
-    });
-  }
-
-  _createCameraResumedCallSession(sipSession: WazoSession, fromSession?: CallSession): CallSession {
-    return this._createCallSession(sipSession, fromSession, {
-      videoMuted: false,
-    });
-  }
-
-  _createCameraDisabledCallSession(sipSession: WazoSession, fromSession?: CallSession): CallSession {
-    return this._createCallSession(sipSession, fromSession, {
-      videoMuted: true,
-    });
-  }
-
-  _createCallSession(sipSession: WazoSession, fromSession?: CallSession | null | undefined, extra: Record<string, any> = {}): CallSession {
-    // 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({
-      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 === 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;
-  }
-
-}