@wazo/sdk 0.37.2 → 0.37.6

package/lib/checker/checks/webrtc-transport.js

@@ -778,7 +778,9 @@ 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))
@@ -805,12 +807,6 @@ 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);
@@ -1005,6 +1001,9 @@ 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') {
@@ -1049,10 +1048,15 @@ 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`);
@@ -1086,7 +1090,11 @@ 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;
@@ -3043,6 +3051,65 @@ 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        ) => ({
@@ -3107,6 +3174,18 @@ 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),
 });
 
 //      
@@ -3691,7 +3770,11 @@ 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: {
@@ -3707,27 +3790,11 @@ 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         ) {
@@ -4484,6 +4551,9 @@ 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        ) =>
@@ -5116,6 +5186,7 @@ const MAX_REGISTER_TRIES = 10;
                            
                          
                            
+                        
                         
   
 
@@ -5142,6 +5213,7 @@ class WebRTCClient extends Emitter {
                          
                                      
                                      
+                        
 
   // sugar
                         
@@ -5180,6 +5252,8 @@ 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);
@@ -5307,14 +5381,22 @@ 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');
@@ -5645,12 +5727,18 @@ 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;
@@ -5709,6 +5797,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: true,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -5751,6 +5841,8 @@ class WebRTCClient extends Emitter {
     }
     options.sessionDescriptionHandlerOptions = {
       constraints: options.constraints,
+      hold: false,
+      conference: isConference,
     };
 
     // Send re-INVITE
@@ -5934,7 +6026,8 @@ 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) {
@@ -5960,11 +6053,8 @@ 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;
     });
   }
@@ -6266,9 +6356,11 @@ 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)         {
@@ -6303,6 +6395,25 @@ 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();
@@ -6532,7 +6643,7 @@ 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);
 
@@ -6607,20 +6718,10 @@ 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) {