@wazo/sdk 0.37.21 → 0.37.25

package/esm/simple/index.js

@@ -863,6 +863,9 @@ class Contact {
   }
 
   static parseMany(response                  )                 {
+    if (!response || !response.results) {
+      return [];
+    }
     return response.results.map(r => Contact.parse(r, response.column_types));
   }
 
@@ -2627,8 +2630,9 @@ class ExternalApp {
                              
                
                
-                 
+                
                       
+                        
  
 
 class Meeting {
@@ -2639,9 +2643,10 @@ class Meeting {
                
                
                
-                 
+                    
                       
                             
+                     
 
   static parse(plain                         )          {
     return new Meeting({
@@ -2651,12 +2656,16 @@ class Meeting {
       ownerUuids: plain.owner_uuids,
       port: plain.port,
       uuid: plain.uuid,
-      number: plain.number,
+      extension: plain.exten,
       persistent: plain.persistent,
+      creationTime: moment(plain.creation_time).toDate(),
     });
   }
 
   static parseMany(items                           )                 {
+    if (!items) {
+      return [];
+    }
     return items.map(plain => Meeting.parse(plain));
   }
 
@@ -2671,8 +2680,9 @@ class Meeting {
     ownerUuids,
     port,
     uri,
-    number,
+    extension,
     persistent,
+    creationTime,
   }         = {}) {
     this.guestSipAuthorization = guestSipAuthorization;
     this.uri = uri;
@@ -2680,8 +2690,9 @@ class Meeting {
     this.ownerUuids = ownerUuids;
     this.port = port;
     this.uuid = uuid;
-    this.number = number;
+    this.extension = extension;
     this.persistent = persistent;
+    this.creationTime = creationTime;
 
     // Useful to compare instead of instanceof with minified code
     this.type = 'Meeting';
@@ -2859,6 +2870,10 @@ class ChatMessage {
                 
 
   static parseMany(plain                         )                     {
+    if (!plain || !plain.items) {
+      return [];
+    }
+
     return plain.items.map(item => ChatMessage.parse(item));
   }
 
@@ -2968,6 +2983,9 @@ class Voicemail {
   }
 
   static parseMany(plain          )                   {
+    if (!plain) {
+      return [];
+    }
     const plainUnread = plain.folders.filter(folder => folder.type === 'new')[0].messages;
     const plainRead = plain.folders.filter(folder => folder.type === 'old')[0].messages;
 
@@ -3077,6 +3095,9 @@ class Call {
                      
 
   static parseMany(plain                     )              {
+    if (!plain) {
+      return [];
+    }
     return plain.map((plainCall              ) => Call.parse(plainCall));
   }
 
@@ -3620,6 +3641,9 @@ class Recording {
                
 
   static parseMany(recordings                      = [])              {
+    if (!recordings) {
+      return [];
+    }
     return recordings.map(item => Recording.parse(item));
   }
 
@@ -3721,6 +3745,10 @@ class CallLog {
   }
 
   static parseMany(plain          )                 {
+    if (!plain || !plain.items) {
+      return [];
+    }
+
     return plain.items.map(item => CallLog.parse(item));
   }
 
@@ -4290,6 +4318,10 @@ class CTIPhone extends Emitter                  {
     return false;
   }
 
+  hasAVideoTrack()          {
+    return false;
+  }
+
   getLocalStreamForCall()               {
     return null;
   }
@@ -4683,6 +4715,9 @@ class ChatRoom {
                          
 
   static parseMany(plain                  )                  {
+    if (!plain || !plain.items) {
+      return [];
+    }
     return plain.items.map(item => ChatRoom.parse(item));
   }
 
@@ -5928,10 +5963,6 @@ const getSrflxOrRelay = (candidates          )           =>
   candidates
     .filter(candidate => candidate && (candidate.type === 'srflx' || candidate.type === 'relay'));
 
-const areCandidateValid = (candidates          )          =>
-  getSrflxOrRelay(candidates)
-    .length > 0;
-
 const getVideoDirection = (sdp        )          => {
   const parsedSdp = sdpParser.parse(sdp);
   const videoMedia = parsedSdp.media.find(media => media.type === 'video');
@@ -5984,8 +6015,6 @@ const fixSdp = (sdp        , candidates          , forcePort          = true)
 /* global RTCSessionDescriptionInit, navigator */
 
 const wazoLogger = IssueReporter$1 ? IssueReporter$1.loggerFor('webrtc-sdh') : console;
-const MAX_WAIT_FOR_ICE_TRIES = 20;
-const WAIT_FOR_ICE_TIMEOUT = 500;
 
 // Customized mediaStreamFactory allowing to send screensharing stream directory when upgrading
 const wazoMediaStreamFactory = (constraints        )                       => {
@@ -6175,38 +6204,6 @@ class WazoSessionDescriptionHandler extends SessionDescriptionHandler {
     this._peerConnection = undefined;
   }
 
-  async _waitForValidGatheredIce()         {
-    let tries = 0;
-
-    while (!areCandidateValid(this.gatheredCandidates) && tries < MAX_WAIT_FOR_ICE_TRIES) {
-      wazoLogger.trace('SessionDescriptionHandler._waitForValidGatheredIce, waiting for ice', {
-        tries,
-        max: MAX_WAIT_FOR_ICE_TRIES,
-      });
-
-      tries++;
-
-      // eslint-disable-next-line no-await-in-loop
-      await new Promise(resolve => setTimeout(resolve, WAIT_FOR_ICE_TIMEOUT));
-    }
-
-    if (tries >= MAX_WAIT_FOR_ICE_TRIES) {
-      const errorMsg = 'No valid candidates found, can\'t answer the call';
-      const error = new Error(errorMsg);
-      wazoLogger.error(errorMsg, { tries, max: MAX_WAIT_FOR_ICE_TRIES });
-
-      // Emit an error, because sip.js catches the exception and switch to status Terminated.
-      this.eventEmitter.emit('error', error);
-
-      wazoLogger.error('No valid candidates found', { tries, candidates: JSON.stringify(this.gatheredCandidates) });
-    }
-
-    // eslint-disable-next-line
-    wazoLogger.trace('Found valid candidates', { tries, candidates: JSON.stringify(this.gatheredCandidates) });
-
-    return this.getLocalSessionDescription();
-  }
-
   // Overridden to send `inactive` in conference
   updateDirection(options                                   , isConference          = false,
     audioOnly          = false)                {
@@ -6974,7 +6971,7 @@ class WebRTCClient extends Emitter {
   }
 
   call(number        , enableVideo          , audioOnly          = false, conference          = false)          {
-    logger$8.info('sdk webrtc creating call', { number, enableVideo, audioOnly });
+    logger$8.info('sdk webrtc creating call', { number, enableVideo, audioOnly, conference });
     const inviterOptions         = {
       sessionDescriptionHandlerOptionsReInvite: {
         conference,
@@ -8050,6 +8047,7 @@ class WebRTCClient extends Emitter {
     const screenSharing = constraints && 'screen' in constraints ? constraints.screen : false;
     const isDesktop = constraints && 'desktop' in constraints ? constraints.desktop : false;
     const withAudio = constraints && 'audio' in constraints ? constraints.audio : true;
+    const mandatoryVideo = constraints && typeof constraints.video === 'object' ? constraints.video.mandatory : {};
 
     logger$8.info('Retrieving media a configuration', { enableVideo, screenSharing, isDesktop, withAudio, constraints });
 
@@ -8057,7 +8055,8 @@ class WebRTCClient extends Emitter {
       constraints: {
         // Exact constraint are not supported with `getDisplayMedia` and we must have a video=false in desktop screenshare
         audio: screenSharing ? !isDesktop : (withAudio ? this._getAudioConstraints() : false),
-        video: screenSharing ? (isDesktop ? ({ mandatory: { chromeMediaSource: 'desktop' } }) : { cursor: 'always' })
+        video: screenSharing ? (isDesktop
+          ? ({ mandatory: { chromeMediaSource: 'desktop', ...(mandatoryVideo || {}) } }) : { cursor: 'always' })
           : this._getVideoConstraints(enableVideo),
         screen: screenSharing,
         desktop: isDesktop,
@@ -8392,11 +8391,14 @@ class WebRTCClient extends Emitter {
 
     session.sessionDescriptionHandler.remoteMediaStream.onaddtrack = onTrack;
 
-    pc.oniceconnectionstatechange = () => {
-      if (pc.iceConnectionState === 'disconnected') {
-        this.eventEmitter.emit(ON_DISCONNECTED$1);
-      }
-    };
+    if (pc) {
+      pc.oniceconnectionstatechange = () => {
+        logger$8.info('on ice connection state changed', { state: pc.iceConnectionState });
+        if (pc.iceConnectionState === 'disconnected') {
+          this.eventEmitter.emit(ON_DISCONNECTED$1);
+        }
+      };
+    }
 
     if (withEvent) {
       this.eventEmitter.emit(ACCEPTED, session, sessionDialog);
@@ -8606,13 +8608,17 @@ class WebRTCClient extends Emitter {
     const nbStats = this.sessionNetworkStats[sessionId].length;
     const lastNetworkStats = this.sessionNetworkStats[sessionId][nbStats - 1];
     const lastAudioSent = lastNetworkStats ? lastNetworkStats.audioBytesSent : 0;
+    const lastAudioContentSent = lastNetworkStats ? lastNetworkStats.audioContentSent : 0;
     const lastVideoSent = lastNetworkStats ? lastNetworkStats.videoBytesSent : 0;
     const lastAudioReceived = lastNetworkStats ? lastNetworkStats.audioBytesReceived : 0;
+    const lastAudioContentReceived = lastNetworkStats ? lastNetworkStats.audioContentReceived : 0;
     const lastVideoReceived = lastNetworkStats ? lastNetworkStats.videoBytesReceived : 0;
     const lastTransportSent = lastNetworkStats ? lastNetworkStats.transportSent : 0;
     const lastTransportReceived = lastNetworkStats ? lastNetworkStats.transportReceived : 0;
-    let audioBytesSent = 0;
-    let audioBytesReceived = 0;
+    let audioBytesSent = 0; // content + header
+    let audioBytesReceived = 0; // content + header
+    let audioContentSent = 0; // useful to detect blank call
+    let audioContentReceived = 0;
     let videoBytesSent = 0;
     let videoBytesReceived = 0;
     let transportSent = 0;
@@ -8621,25 +8627,27 @@ class WebRTCClient extends Emitter {
 
     stats.forEach(report => {
       if (report.type === 'outbound-rtp' && report.kind === 'audio') {
-        audioBytesSent += report.bytesSent + report.headerBytesSent;
+        audioBytesSent += Number(report.bytesSent) + Number(report.headerBytesSent);
+        audioContentSent += Number(report.bytesSent);
       }
       if (report.type === 'remote-outbound-rtp' && report.kind === 'audio') {
-        audioBytesSent += report.bytesSent;
+        audioBytesSent += Number(report.bytesSent);
       }
       if (report.type === 'outbound-rtp' && report.kind === 'video') {
-        videoBytesSent += report.bytesSent + report.headerBytesSent;
+        videoBytesSent += Number(report.bytesSent) + Number(report.headerBytesSent);
       }
       if (report.type === 'inbound-rtp' && report.kind === 'audio') {
-        packetsLost += report.packetsLost;
-        networkStats.packetsReceived = report.packetsReceived;
-        audioBytesReceived += report.bytesReceived + report.headerBytesReceived;
+        packetsLost += Number(report.packetsLost);
+        networkStats.packetsReceived = Number(report.packetsReceived);
+        audioBytesReceived += Number(report.bytesReceived) + Number(report.headerBytesReceived);
+        audioContentReceived += Number(report.bytesReceived);
       }
       if (report.type === 'inbound-rtp' && report.kind === 'video') {
-        videoBytesReceived += report.bytesReceived + report.headerBytesReceived;
+        videoBytesReceived += Number(report.bytesReceived) + Number(report.headerBytesReceived);
       }
       if (report.type === 'outbound-rtp' && report.kind === 'video') {
         if ('framesPerSecond' in report) {
-          networkStats.framesPerSecond = report.framesPerSecond;
+          networkStats.framesPerSecond = Number(report.framesPerSecond);
         }
         if ('framerateMean' in report) {
           // framerateMean is only available in FF
@@ -8647,24 +8655,31 @@ class WebRTCClient extends Emitter {
         }
       }
       if (report.type === 'remote-inbound-rtp' && report.kind === 'audio') {
-        packetsLost += report.packetsLost;
-        networkStats.roundTripTime = report.roundTripTime;
-        networkStats.jitter = report.jitter;
+        packetsLost += Number(report.packetsLost);
+        networkStats.roundTripTime = Number(report.roundTripTime);
+        networkStats.jitter = Number(report.jitter);
       }
       if (report.type === 'remote-inbound-rtp' && report.kind === 'video') {
-        packetsLost += report.packetsLost;
+        packetsLost += Number(report.packetsLost);
       }
       if (report.type === 'transport') {
-        transportSent += report.bytesSent;
-        transportReceived += report.bytesReceived;
+        transportSent += Number(report.bytesSent);
+        transportReceived += Number(report.bytesReceived);
       }
     });
 
     networkStats.packetsLost = packetsLost;
+
     networkStats.totalAudioBytesSent = audioBytesSent;
     networkStats.audioBytesSent = audioBytesSent - lastAudioSent;
     networkStats.totalAudioBytesReceived = audioBytesReceived - lastAudioReceived;
     networkStats.audioBytesReceived = audioBytesReceived;
+
+    networkStats.totalAudioContentSent = audioContentSent;
+    networkStats.audioContentSent = audioContentSent - lastAudioContentSent;
+    networkStats.totalAudioBytesReceived = audioContentReceived - lastAudioContentReceived;
+    networkStats.audioContentReceived = audioContentReceived;
+
     networkStats.totalVideoBytesSent = videoBytesSent;
     networkStats.videoBytesSent = videoBytesSent - lastVideoSent;
     networkStats.totalVideoBytesReceived = videoBytesReceived;
@@ -8876,6 +8891,7 @@ class WebRTCPhone extends Emitter                  {
       constraints,
       audioOnly,
       conference,
+      iceRestart,
     });
 
     if (!sipSession) {
@@ -8979,6 +8995,11 @@ class WebRTCPhone extends Emitter                  {
 
     this.client.downgradeToAudio(sipSession);
 
+    if (callSession) {
+      callSession.cameraEnabled = false;
+      this._updateCallSession(callSession);
+    }
+
     if (withMessage) {
       this._sendReinviteMessage(callSession, false);
     }
@@ -9006,6 +9027,11 @@ class WebRTCPhone extends Emitter                  {
 
     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;
@@ -9058,7 +9084,10 @@ class WebRTCPhone extends Emitter                  {
 
         onCancel(message);
         const elsewhere = message.data.indexOf('cause=26') !== -1 && message.data.indexOf('completed elsewhere') !== -1;
-        this.eventEmitter.emit(ON_CALL_CANCELED, this._createCallSession(sipSession), elsewhere);
+        const callSession = this._createCallSession(sipSession);
+        delete this.callSessions[callSession.getId()];
+
+        this.eventEmitter.emit(ON_CALL_CANCELED, callSession, elsewhere);
       };
     } else if (sipSession instanceof Invitation) {
       console.warn('sipSession._onCancel not found, please update the wazo SDK accordingly');
@@ -9239,18 +9268,20 @@ class WebRTCPhone extends Emitter                  {
       screenTrack,
     });
 
-    screenTrack.onended = () => {
-      logger$7.info('WebRTC phone - on screenshare ended', {
-        hadVideo,
-        id: this.getSipSessionId(sipSession),
-        screenTrack,
-      });
+    if (screenTrack) {
+      screenTrack.onended = () => {
+        logger$7.info('WebRTC phone - on screenshare ended', {
+          hadVideo,
+          id: this.getSipSessionId(sipSession),
+          screenTrack,
+        });
 
-      this.eventEmitter.emit(
-        ON_SHARE_SCREEN_ENDING,
-        this._createCallSession(sipSession, callSession),
-      );
-    };
+        this.eventEmitter.emit(
+          ON_SHARE_SCREEN_ENDING,
+          this._createCallSession(sipSession, callSession),
+        );
+      };
+    }
 
     this.currentScreenShare = {
       stream: screenStream,
@@ -9270,7 +9301,7 @@ class WebRTCPhone extends Emitter                  {
   }
 
   _onCallAccepted(sipSession         , cameraEnabled         )              {
-    logger$7.info('WebRTC phone -  on call accepted', { sipId: sipSession.id, cameraEnabled });
+    logger$7.info('WebRTC phone - on call accepted', { sipId: sipSession.id, cameraEnabled });
 
     const callSession = this._createAcceptedCallSession(sipSession, cameraEnabled);
     this.currentSipSession = sipSession;
@@ -9782,7 +9813,7 @@ class WebRTCPhone extends Emitter                  {
   // @TODO: line is not used here
   async makeCall(number        , line     , cameraEnabled          ,
     audioOnly          = false, conference          = false)                        {
-    logger$7.info('make WebRTC call', { number, lineId: line ? line.id : null, cameraEnabled });
+    logger$7.info('make WebRTC call', { number, lineId: line ? line.id : null, cameraEnabled, audioOnly, conference });
     if (!number) {
       return Promise.resolve(null);
     }
@@ -10143,9 +10174,13 @@ class WebRTCPhone extends Emitter                  {
       if (this.shouldSendReinvite && this.currentSipSession
         && this.currentSipSession.state === SessionState.Established) {
         this.shouldSendReinvite = false;
+        const pc = this.currentSipSession.sessionDescriptionHandler.peerConnection;
+        const isConference = pc ? pc.sfu : false;
+        const hasVideo = this.currentCallSession && this.currentCallSession.cameraEnabled;
+
         try {
           // Send reinvite with iceRestart
-          this.sendReinvite(this.currentCallSession, null, false, false, true);
+          this.sendReinvite(this.currentCallSession, null, isConference, !hasVideo, true);
         } catch (e) {
           logger$7.error('WebRTC reinvite after register, error', { message: e.message, stack: e.stack });
         }
@@ -10229,6 +10264,15 @@ class WebRTCPhone extends Emitter                  {
     return this.client.getSipSessionId(sipSession);
   }
 
+  _updateCallSession(callSession             ) {
+    const sipSession = this.findSipSession(callSession);
+    const sipSessionId = sipSession ? this.getSipSessionId(sipSession) : null;
+
+    if (sipSessionId) {
+      this.callSessions[sipSessionId] = callSession;
+    }
+  }
+
   _onMessage(message         ) {
     if (!message || message.method !== 'MESSAGE') {
       return;