@wazo/sdk 0.37.2 → 0.37.6

package/lib/web-rtc-client.js

@@ -780,7 +780,9 @@ class WazoSessionDescriptionHandler extends sessionDescriptionHandler.SessionDes
       .then(() => {
         if (isConference && (options.constraints && !options.constraints.video) && !('hold' in options) && !audioOnly) {
           // Add a video an empty bundle to be able to replaceTrack when joining a conference without video
-          this.peerConnection.addTransceiver('video', { streams: [this._localMediaStream], direction: 'sendrecv' });
+          if (this.peerConnection.addTransceiver) {
+            this.peerConnection.addTransceiver('video', { streams: [this._localMediaStream], direction: 'sendrecv' });
+          }
         }
       })
       .then(() => this.createLocalOfferOrAnswer(options))
@@ -807,12 +809,6 @@ class WazoSessionDescriptionHandler extends sessionDescriptionHandler.SessionDes
         };
       })
       .then((sessionDescription) => this.applyModifiers(sessionDescription, modifiers))
-      .then((sessionDescription) => {
-        // Set new constraints to avoid the constraints check issue in `sdh.getLocalMediaStream` later.
-        this.localMediaStreamConstraints = options.constraints;
-
-        return sessionDescription;
-      })
       .then((sessionDescription) => ({ body: sessionDescription.sdp, contentType: 'application/sdp' }))
       .catch((error) => {
         wazoLogger.error('error when creating media', error);
@@ -1007,6 +1003,9 @@ class WazoSessionDescriptionHandler extends sessionDescriptionHandler.SessionDes
 
         // set the transceiver direction to the answer direction
         this._peerConnection.getTransceivers().forEach((transceiver) => {
+          if (transceiver.stopped) {
+            return;
+          }
           if (transceiver.direction /* guarding, but should always be true */) {
             const { receiver } = transceiver;
             if (isConference && audioOnly && receiver.track && receiver.track.kind === 'video') {
@@ -1051,10 +1050,15 @@ class WazoSessionDescriptionHandler extends sessionDescriptionHandler.SessionDes
       if (kind !== 'audio' && kind !== 'video') {
         throw new Error(`Unknown new track kind ${kind}.`);
       }
-      const sender = pc.getSenders().find((otherSender) => otherSender.track && otherSender.track.kind === kind);
+      const sender = pc.getSenders && pc.getSenders().find((otherSender) =>
+        otherSender.track && otherSender.track.kind === kind);
 
       // Do not reuse sender video tracks in SFU
       if (sender && (!sfu || newTrack.kind === 'audio')) {
+        if (sender.track) {
+          // eslint-disable-next-line no-param-reassign
+          newTrack.enabled = sender.track.enabled;
+        }
         trackUpdates.push(
           new Promise((resolve) => {
             this.logger.debug(`SessionDescriptionHandler.setLocalMediaStream - replacing sender ${kind} track`);
@@ -1088,7 +1092,11 @@ class WazoSessionDescriptionHandler extends sessionDescriptionHandler.SessionDes
             // Review: could make streamless tracks a configurable option?
             // https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addTrack#Usage_notes
             try {
-              pc.addTrack(newTrack, localStream);
+              if (pc.addTrack) {
+                pc.addTrack(newTrack, localStream);
+              } else {
+                pc.addStream(localStream);
+              }
             } catch (error) {
               this.logger.error(`SessionDescriptionHandler.setLocalMediaStream - failed to add sender ${kind} track`);
               throw error;
@@ -3045,6 +3053,65 @@ class ExternalApp {
   }
 }
 
+//      
+
+                                       
+                                  
+                   
+               
+                             
+               
+               
+                
+ 
+
+class Meeting {
+               
+
+                                
+                   
+               
+               
+               
+                 
+                            
+
+  static parse(plain                         )          {
+    return new Meeting({
+      guestSipAuthorization: plain.guest_sip_authorization,
+      hostname: plain.hostname,
+      name: plain.name,
+      ownerUuids: plain.owner_uuids,
+      port: plain.port,
+      uuid: plain.uuid,
+      number: plain.number,
+    });
+  }
+
+  static newFrom(meeting         ) {
+    return newFrom(meeting, Meeting);
+  }
+
+  constructor({ uuid, name, guestSipAuthorization, ownerUuids, port, hostname, number }         = {}) {
+    this.guestSipAuthorization = guestSipAuthorization;
+    this.hostname = hostname;
+    this.name = name;
+    this.ownerUuids = ownerUuids;
+    this.port = port;
+    this.uuid = uuid;
+    this.number = number;
+
+    // Useful to compare instead of instanceof with minified code
+    this.type = 'Meeting';
+  }
+
+  getGuestSipCredentials() {
+    // eslint-disable-next-line no-undef
+    const [username, secret] = atob(this.guestSipAuthorization).split(':');
+    return { username, secret };
+  }
+}
+
 /*       */
 
 var confdMethods = (client              , baseUrl        ) => ({
@@ -3109,6 +3176,18 @@ var confdMethods = (client              , baseUrl        ) => ({
       return null;
     }
   },
+
+  createMyMeeting: (name        )                   =>
+    client.post(`${baseUrl}/users/me/meetings`, { name }).then(Meeting.parse),
+
+  deleteMyMeeting: (meetingUuid        )                   =>
+    client.delete(`${baseUrl}/users/me/meetings/${meetingUuid}`, null),
+
+  getMeeting: (meetingUuid        )                   =>
+    client.get(`${baseUrl}/meetings/${meetingUuid}`, null).then(Meeting.parse),
+
+  guestGetMeeting: (meetingUuid        )                   =>
+    client.get(`${baseUrl}/guests/me/meetings/${meetingUuid}`, null).then(Meeting.parse),
 });
 
 //      
@@ -3693,7 +3772,11 @@ class CallLog {
       answered: plain.answered,
       callDirection: plain.call_direction,
       destination: {
-        extension: plain.requested_extension || plain.destination_extension,
+        // @TEMP: Temporarily assuming empty numbers are meetings
+        // which is admittedly a very dangerous assumption. Did i mention it was temporary?
+        extension: plain.requested_extension
+          || plain.destination_extension
+          || `meeting-${plain.requested_name || plain.destination_name || ''}`,
         name: plain.requested_name || plain.destination_name || '',
       },
       source: {
@@ -3709,27 +3792,11 @@ class CallLog {
   }
 
   static parseNew(plain                 , session         )          {
-    return new CallLog({
-      answer: plain.answer ? moment__default['default'](plain.answer).toDate() : null,
-      answered: plain.answered,
-      callDirection: plain.call_direction,
-      destination: {
-        extension: plain.requested_extension || plain.destination_extension,
-        name: plain.requested_name || plain.destination_name || '',
-      },
-      source: {
-        extension: plain.source_extension,
-        name: plain.source_name,
-      },
-      id: plain.id,
-      duration: (plain.duration || 0) * 1000, // duration is in seconds
-      start: moment__default['default'](plain.start).toDate(),
-      end: plain.end ? moment__default['default'](plain.end).toDate() : null,
-      recordings: Recording.parseMany(plain.recordings),
-      // @TODO: FIXME add verification declined vs missed call
-      newMissedCall: session
-        && session.hasExtension(plain.requested_extension || plain.destination_extension) && !plain.answered,
-    });
+    const callLog          = CallLog.parse(plain);
+    // @TODO: FIXME add verification declined vs missed call
+    callLog.newMissedCall = session
+      && session.hasExtension(plain.requested_extension || plain.destination_extension) && !plain.answered;
+    return callLog;
   }
 
   static newFrom(profile         ) {
@@ -4486,6 +4553,9 @@ var calldMethods = (client              , baseUrl        ) => ({
   getConferenceParticipantsAsUser: async (conferenceId        )                  =>
     client.get(`${baseUrl}/users/me/conferences/${conferenceId}/participants`),
 
+  getMeetingParticipantsAsUser: async (meetingUuid        )                  =>
+    client.get(`${baseUrl}/users/me/meetings/${meetingUuid}/participants`),
+
   listTrunks: () => client.get(`${baseUrl}/trunks`),
 
   mute: (callId        ) =>
@@ -5121,6 +5191,7 @@ const MAX_REGISTER_TRIES = 10;
                            
                          
                            
+                        
                         
   
 
@@ -5147,6 +5218,7 @@ class WebRTCClient extends Emitter {
                          
                                      
                                      
+                        
 
   // sugar
                         
@@ -5185,6 +5257,8 @@ class WebRTCClient extends Emitter {
     super();
     this.uaConfigOverrides = uaConfigOverrides;
     this.config = config;
+    this.skipRegister = config.skipRegister;
+
     this._buildConfig(config, session).then((newConfig              ) => {
       this.config = newConfig;
       this.userAgent = this.createUserAgent(uaConfigOverrides);
@@ -5312,14 +5386,22 @@ class WebRTCClient extends Emitter {
   }
 
   register(tries         = 0)               {
-    logger.info('sdk webrtc registering...', {
+    const logInfo = {
       userAgent: !!this.userAgent,
       registered: this.isRegistered(),
       connectionPromise: !!this.connectionPromise,
       registerer: !!this.registerer,
       waiting: this.registerer && this.registerer.waiting,
       tries,
-    });
+      skipRegister: this.skipRegister,
+    };
+
+    if (this.skipRegister) {
+      logger.info('sdk webrtc skip register...', logInfo);
+      return Promise.resolve();
+    }
+
+    logger.info('sdk webrtc registering...', logInfo);
 
     if (!this.userAgent) {
       logger.info('sdk webrtc recreating User Agent');
@@ -5650,12 +5732,18 @@ class WebRTCClient extends Emitter {
     }
 
     if (pc.getSenders) {
+      if (!pc.getSenders().length) {
+        return false;
+      }
       pc.getSenders().forEach(sender => {
         if (sender && sender.track && sender.track.kind === 'audio') {
           muted = muted && !sender.track.enabled;
         }
       });
     } else {
+      if (!pc.getLocalStreams().length) {
+        return false;
+      }
       pc.getLocalStreams().forEach(stream => {
         stream.getAudioTracks().forEach(track => {
           muted = muted && !track.enabled;
@@ -5714,6 +5802,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -5756,6 +5846,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -5939,7 +6031,8 @@ class WebRTCClient extends Emitter {
     }
     const sdh = session.sessionDescriptionHandler;
     const pc = sdh.peerConnection;
-    const localStream = this.getLocalStream(this.getSipSessionId(session));
+    const sessionId = this.getSipSessionId(session);
+    const localStream = this.getLocalStream(sessionId);
 
     // Release old video stream
     if (localStream) {
@@ -5965,11 +6058,8 @@ class WebRTCClient extends Emitter {
 
       // let's update the local stream
       this.eventEmitter.emit('onVideoInputChange', stream);
-      try {
-        await sdh.setLocalMediaStream(stream);
-      } catch (_) {
-        // Nothing to do
-      }
+      this.setLocalMediaStream(sessionId, stream);
+
       return stream;
     });
   }
@@ -6271,9 +6361,11 @@ class WebRTCClient extends Emitter {
       remoteStream.removeTrack(track);
     });
 
-    pc.getReceivers().forEach(receiver => {
-      remoteStream.addTrack(receiver.track);
-    });
+    if (pc.getReceivers) {
+      pc.getReceivers().forEach(receiver => {
+        remoteStream.addTrack(receiver.track);
+      });
+    }
   }
 
   getMediaConfiguration(enableVideo         , conference          = false, constraints          = null)         {
@@ -6308,6 +6400,25 @@ class WebRTCClient extends Emitter {
     return sessionId in this.conferences;
   }
 
+  createAudioElementFor(sessionId        ) {
+    const audio      = document.createElement('audio');
+    audio.setAttribute('id', `audio-${sessionId}`);
+
+    if (audio.setSinkId && this.audioOutputDeviceId) {
+      audio.setSinkId(this.audioOutputDeviceId);
+    }
+
+    if (document.body) {
+      document.body.appendChild(audio);
+    }
+    this.audioElements[sessionId] = audio;
+
+    audio.volume = this.audioOutputVolume;
+    audio.autoplay = true;
+
+    return audio;
+  }
+
   _onTransportError() {
     this.eventEmitter.emit(TRANSPORT_ERROR);
     this.attemptReconnection();
@@ -6537,7 +6648,7 @@ class WebRTCClient extends Emitter {
     let hadRemoteVideo = false;
 
     // Monkey patch `onInviteRequest` to be able to know if there was a remote video stream before `onInvite` is called
-    // Becase when `onInvite` is called we already got the video track
+    // Because when `onInvite` is called we already got the video track
     session.onInviteRequest = (request) => {
       hadRemoteVideo = this.hasARemoteVideoTrack(sipSessionId);
 
@@ -6612,20 +6723,10 @@ class WebRTCClient extends Emitter {
     const isConference = this.isConference(sessionId);
 
     if (this._hasAudio() && this._isWeb() && !(sessionId in this.audioElements)) {
-      const audio      = document.createElement('audio');
-      audio.setAttribute('id', `audio-${sessionId}`);
-
-      if (audio.setSinkId && this.audioOutputDeviceId) {
-        audio.setSinkId(this.audioOutputDeviceId);
-      }
-
-      if (document.body) {
-        document.body.appendChild(audio);
-      }
-      this.audioElements[sessionId] = audio;
+      this.createAudioElementFor(sessionId);
     }
 
-    const audioElement = this.audioElements[this.getSipSessionId(session)];
+    const audioElement = this.audioElements[sessionId];
     const stream = newStream || this.getRemoteStream(sessionId);
 
     if (!this._isWeb() || !stream) {

package/lib/websocket-client.js

@@ -475,6 +475,8 @@ const CONFERENCE_USER_PARTICIPANT_JOINED = 'conference_user_participant_joined';
 const CONFERENCE_USER_PARTICIPANT_LEFT = 'conference_user_participant_left';
 const CONFERENCE_USER_PARTICIPANT_TALK_STARTED = 'conference_user_participant_talk_started';
 const CONFERENCE_USER_PARTICIPANT_TALK_STOPPED = 'conference_user_participant_talk_stopped';
+const MEETING_USER_PARTICIPANT_JOINED = 'meeting_user_participant_joined';
+const MEETING_USER_PARTICIPANT_LEFT = 'meeting_user_participant_left';
 const SWITCHBOARD_QUEUED_CALLS_UPDATED = 'switchboard_queued_calls_updated';
 const SWITCHBOARD_QUEUED_CALL_ANSWERED = 'switchboard_queued_call_answered';
 const SWITCHBOARD_HELD_CALLS_UPDATED = 'switchboard_held_calls_updated';
@@ -821,6 +823,8 @@ WebSocketClient.eventLists = [
   CHATD_USER_ROOM_CREATED,
   CONFERENCE_USER_PARTICIPANT_JOINED,
   CONFERENCE_USER_PARTICIPANT_LEFT,
+  MEETING_USER_PARTICIPANT_JOINED,
+  MEETING_USER_PARTICIPANT_LEFT,
   CONFERENCE_USER_PARTICIPANT_TALK_STARTED,
   CONFERENCE_USER_PARTICIPANT_TALK_STOPPED,
   SWITCHBOARD_QUEUED_CALLS_UPDATED,
@@ -908,6 +912,8 @@ exports.FAX_OUTBOUND_USER_FAILED = FAX_OUTBOUND_USER_FAILED;
 exports.FAX_OUTBOUND_USER_SUCCEEDED = FAX_OUTBOUND_USER_SUCCEEDED;
 exports.HEARTBEAT_ENGINE_VERSION = HEARTBEAT_ENGINE_VERSION;
 exports.LINE_STATUS_UPDATED = LINE_STATUS_UPDATED;
+exports.MEETING_USER_PARTICIPANT_JOINED = MEETING_USER_PARTICIPANT_JOINED;
+exports.MEETING_USER_PARTICIPANT_LEFT = MEETING_USER_PARTICIPANT_LEFT;
 exports.SOCKET_EVENTS = SOCKET_EVENTS;
 exports.SWITCHBOARD_HELD_CALLS_UPDATED = SWITCHBOARD_HELD_CALLS_UPDATED;
 exports.SWITCHBOARD_HELD_CALL_ANSWERED = SWITCHBOARD_HELD_CALL_ANSWERED;

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@wazo/sdk",
-  "version": "0.37.2",
+  "version": "0.37.6",
   "description": "Wazo's JavaScript Software Development Kit.",
   "main": "dist/wazo-sdk.js",
   "author": "Wazo (http://wazo.io)",

package/src/api/calld.js

@@ -133,6 +133,9 @@ export default (client: ApiRequester, baseUrl: string) => ({
   getConferenceParticipantsAsUser: async (conferenceId: string): Promise<Object> =>
     client.get(`${baseUrl}/users/me/conferences/${conferenceId}/participants`),
 
+  getMeetingParticipantsAsUser: async (meetingUuid: string): Promise<Object> =>
+    client.get(`${baseUrl}/users/me/meetings/${meetingUuid}/participants`),
+
   listTrunks: () => client.get(`${baseUrl}/trunks`),
 
   mute: (callId: string) =>

package/src/api/confd.js

@@ -4,6 +4,7 @@ import type { UUID, ListConfdUsersResponse, ListApplicationsResponse } from '../
 import Profile from '../domain/Profile';
 import SipLine from '../domain/SipLine';
 import ExternalApp from '../domain/ExternalApp';
+import Meeting from '../domain/Meeting';
 
 export default (client: ApiRequester, baseUrl: string) => ({
   listUsers: (): Promise<ListConfdUsersResponse> => client.get(`${baseUrl}/users`, null),
@@ -67,4 +68,16 @@ export default (client: ApiRequester, baseUrl: string) => ({
       return null;
     }
   },
+
+  createMyMeeting: (name: string): Promise<Meeting> =>
+    client.post(`${baseUrl}/users/me/meetings`, { name }).then(Meeting.parse),
+
+  deleteMyMeeting: (meetingUuid: string): Promise<Meeting> =>
+    client.delete(`${baseUrl}/users/me/meetings/${meetingUuid}`, null),
+
+  getMeeting: (meetingUuid: string): Promise<Meeting> =>
+    client.get(`${baseUrl}/meetings/${meetingUuid}`, null).then(Meeting.parse),
+
+  guestGetMeeting: (meetingUuid: string): Promise<Meeting> =>
+    client.get(`${baseUrl}/guests/me/meetings/${meetingUuid}`, null).then(Meeting.parse),
 });

package/src/domain/CallLog.js

@@ -91,7 +91,11 @@ export default class CallLog {
       answered: plain.answered,
       callDirection: plain.call_direction,
       destination: {
-        extension: plain.requested_extension || plain.destination_extension,
+        // @TEMP: Temporarily assuming empty numbers are meetings
+        // which is admittedly a very dangerous assumption. Did i mention it was temporary?
+        extension: plain.requested_extension
+          || plain.destination_extension
+          || `meeting-${plain.requested_name || plain.destination_name || ''}`,
         name: plain.requested_name || plain.destination_name || '',
       },
       source: {
@@ -107,27 +111,11 @@ export default class CallLog {
   }
 
   static parseNew(plain: CallLogResponse, session: Session): CallLog {
-    return new CallLog({
-      answer: plain.answer ? moment(plain.answer).toDate() : null,
-      answered: plain.answered,
-      callDirection: plain.call_direction,
-      destination: {
-        extension: plain.requested_extension || plain.destination_extension,
-        name: plain.requested_name || plain.destination_name || '',
-      },
-      source: {
-        extension: plain.source_extension,
-        name: plain.source_name,
-      },
-      id: plain.id,
-      duration: (plain.duration || 0) * 1000, // duration is in seconds
-      start: moment(plain.start).toDate(),
-      end: plain.end ? moment(plain.end).toDate() : null,
-      recordings: Recording.parseMany(plain.recordings),
-      // @TODO: FIXME add verification declined vs missed call
-      newMissedCall: session
-        && session.hasExtension(plain.requested_extension || plain.destination_extension) && !plain.answered,
-    });
+    const callLog: CallLog = CallLog.parse(plain);
+    // @TODO: FIXME add verification declined vs missed call
+    callLog.newMissedCall = session
+      && session.hasExtension(plain.requested_extension || plain.destination_extension) && !plain.answered;
+    return callLog;
   }
 
   static newFrom(profile: CallLog) {

package/src/domain/Meeting.js

@@ -0,0 +1,59 @@
+// @flow
+import newFrom from '../utils/new-from';
+
+export type MeetingCreationResponse = {
+  guest_sip_authorization: string,
+  hostname: string,
+  name: string,
+  owner_uuids: Array<string>,
+  port: string,
+  uuid: string,
+  number: string
+}
+
+export default class Meeting {
+  type: string;
+
+  guestSipAuthorization: string;
+  hostname: string;
+  uuid: string;
+  name: string;
+  port: string;
+  number: string;
+  ownerUuids: Array<string>;
+
+  static parse(plain: MeetingCreationResponse): Meeting {
+    return new Meeting({
+      guestSipAuthorization: plain.guest_sip_authorization,
+      hostname: plain.hostname,
+      name: plain.name,
+      ownerUuids: plain.owner_uuids,
+      port: plain.port,
+      uuid: plain.uuid,
+      number: plain.number,
+    });
+  }
+
+  static newFrom(meeting: Meeting) {
+    return newFrom(meeting, Meeting);
+  }
+
+  constructor({ uuid, name, guestSipAuthorization, ownerUuids, port, hostname, number }: Object = {}) {
+    this.guestSipAuthorization = guestSipAuthorization;
+    this.hostname = hostname;
+    this.name = name;
+    this.ownerUuids = ownerUuids;
+    this.port = port;
+    this.uuid = uuid;
+    this.number = number;
+
+    // Useful to compare instead of instanceof with minified code
+    this.type = 'Meeting';
+  }
+
+  getGuestSipCredentials() {
+    // eslint-disable-next-line no-undef
+    const [username, secret] = atob(this.guestSipAuthorization).split(':');
+    return { username, secret };
+  }
+}

package/src/domain/Phone/WebRTCPhone.js

@@ -200,7 +200,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     });
 
     if (!sipSession) {
-      return null;
+      return false;
     }
 
     // $FlowFixMe
@@ -210,19 +210,17 @@ export default class WebRTCPhone extends Emitter implements Phone {
     // Downgrade
     if (constraints && !isUpgrade) {
       // No reinvite needed
-      return this._downgradeToAudio(callSession);
+      this._downgradeToAudio(callSession);
+      return false;
     }
 
     if (isUpgrade) {
       const shouldReinvite = await this._upgradeToVideo(callSession, constraints, conference);
       if (!shouldReinvite) {
-        return;
+        return false;
       }
     }
 
-    // Force reinvite in SDH
-    sipSession.sessionDescriptionHandler.localMediaStreamConstraints = null;
-
     return this.client.reinvite(sipSession, constraints, conference, audioOnly);
   }
 
@@ -295,7 +293,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
   _downgradeToAudio(callSession: ?CallSession) {
     const sipSession = this.findSipSession(callSession);
     if (!sipSession) {
-      return;
+      return false;
     }
 
     // Release local video stream when downgrading to audio
@@ -499,6 +497,10 @@ export default class WebRTCPhone extends Emitter implements Phone {
       localStream.removeTrack(videoTrack);
     }
 
+    if (localStream) {
+      localStream.addTrack(screenTrack);
+    }
+
     if (sender) {
       sender.replaceTrack(screenTrack);
     }
@@ -514,6 +516,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
 
     logger.info('WebRTC phone - stop screen sharing');
+    let reinvited = false;
 
     try {
       if (this.currentScreenShare.stream) {
@@ -528,7 +531,8 @@ export default class WebRTCPhone extends Emitter implements Phone {
         // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
         // We have to downgrade to audio.
         const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
-        await this.sendReinvite(targetCallSession, { audio: false, video: screenshareStopped }, conference);
+        const constraints = { audio: false, video: screenshareStopped };
+        reinvited = await this.sendReinvite(targetCallSession, constraints, conference);
       }
     } catch (e) {
       console.warn(e);
@@ -542,6 +546,8 @@ export default class WebRTCPhone extends Emitter implements Phone {
     );
 
     this.currentScreenShare = null;
+
+    return reinvited;
   }
 
   async _getStreamFromConstraints(constraints: Object, conference: boolean = false): Promise<?MediaStream> {
@@ -602,6 +608,10 @@ export default class WebRTCPhone extends Emitter implements Phone {
     this.client.changeAudioOutputDevice(id);
   }
 
+  createAudioElementFor(sessionId: string) {
+    return this.client.createAudioElementFor(sessionId);
+  }
+
   changeRingDevice(id: string) {
     this.audioRingDeviceId = id;
   }
@@ -1249,6 +1259,10 @@ export default class WebRTCPhone extends Emitter implements Phone {
     return callSession ? this.client.hasLocalVideo(callSession.sipCallId) : null;
   }
 
+  hasALocalVideoTrack(callSession: CallSession) {
+    return callSession ? this.client.hasALocalVideoTrack(callSession.sipCallId) : null;
+  }
+
   getRemoteStream(callSession: CallSession): ?MediaStream {
     return callSession ? this.client.getRemoteStream(callSession.sipCallId) : null;
   }

package/src/index.js

@@ -30,6 +30,7 @@ import IndirectTransfer from './domain/IndirectTransfer';
 import SwitchboardCall from './domain/SwitchboardCall';
 import WebRTCPhone from './domain/Phone/WebRTCPhone';
 import CTIPhone from './domain/Phone/CTIPhone';
+import Meeting from './domain/Meeting';
 import type {
   NewContact as NewContactType,
   ContactResponse as ContactResponseType,
@@ -122,4 +123,5 @@ export default {
   SOCKET_EVENTS,
   Wazo,
   WebRTCPhone,
+  Meeting,
 };