@wazo/sdk 0.37.22 → 0.37.23

package/lib/simple/room/LocalParticipant.js

@@ -1586,6 +1586,9 @@ class Contact {
   }
 
   static parseMany(response                  )                 {
+    if (!response || !response.results) {
+      return [];
+    }
     return response.results.map(r => Contact.parse(r, response.column_types));
   }
 
@@ -2693,6 +2696,9 @@ class Meeting {
   }
 
   static parseMany(items                           )                 {
+    if (!items) {
+      return [];
+    }
     return items.map(plain => Meeting.parse(plain));
   }
 
@@ -2895,6 +2901,10 @@ class ChatMessage {
                 
 
   static parseMany(plain                         )                     {
+    if (!plain || !plain.items) {
+      return [];
+    }
+
     return plain.items.map(item => ChatMessage.parse(item));
   }
 
@@ -3004,6 +3014,9 @@ 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;
 
@@ -3113,6 +3126,9 @@ class Call {
                      
 
   static parseMany(plain                     )              {
+    if (!plain) {
+      return [];
+    }
     return plain.map((plainCall              ) => Call.parse(plainCall));
   }
 
@@ -3630,6 +3646,9 @@ class Recording {
                
 
   static parseMany(recordings                      = [])              {
+    if (!recordings) {
+      return [];
+    }
     return recordings.map(item => Recording.parse(item));
   }
 
@@ -3731,6 +3750,10 @@ class CallLog {
   }
 
   static parseMany(plain          )                 {
+    if (!plain || !plain.items) {
+      return [];
+    }
+
     return plain.items.map(item => CallLog.parse(item));
   }
 
@@ -4300,6 +4323,10 @@ class CTIPhone extends Emitter                  {
     return false;
   }
 
+  hasAVideoTrack()          {
+    return false;
+  }
+
   getLocalStreamForCall()               {
     return null;
   }
@@ -4693,6 +4720,9 @@ class ChatRoom {
                          
 
   static parseMany(plain                  )                  {
+    if (!plain || !plain.items) {
+      return [];
+    }
     return plain.items.map(item => ChatRoom.parse(item));
   }
 
@@ -5938,10 +5968,6 @@ 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__default['default'].parse(sdp);
   const videoMedia = parsedSdp.media.find(media => media.type === 'video');
@@ -5994,8 +6020,6 @@ 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        )                       => {
@@ -6185,38 +6209,6 @@ class WazoSessionDescriptionHandler extends sessionDescriptionHandler.SessionDes
     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)                {
@@ -8632,25 +8624,25 @@ 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);
       }
       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);
       }
       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
@@ -8658,16 +8650,16 @@ 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);
       }
     });
 
@@ -9253,18 +9245,20 @@ 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,

package/lib/simple/room/Participant.js

@@ -1586,6 +1586,9 @@ class Contact {
   }
 
   static parseMany(response                  )                 {
+    if (!response || !response.results) {
+      return [];
+    }
     return response.results.map(r => Contact.parse(r, response.column_types));
   }
 
@@ -2693,6 +2696,9 @@ class Meeting {
   }
 
   static parseMany(items                           )                 {
+    if (!items) {
+      return [];
+    }
     return items.map(plain => Meeting.parse(plain));
   }
 
@@ -2895,6 +2901,10 @@ class ChatMessage {
                 
 
   static parseMany(plain                         )                     {
+    if (!plain || !plain.items) {
+      return [];
+    }
+
     return plain.items.map(item => ChatMessage.parse(item));
   }
 
@@ -3004,6 +3014,9 @@ 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;
 
@@ -3113,6 +3126,9 @@ class Call {
                      
 
   static parseMany(plain                     )              {
+    if (!plain) {
+      return [];
+    }
     return plain.map((plainCall              ) => Call.parse(plainCall));
   }
 
@@ -3630,6 +3646,9 @@ class Recording {
                
 
   static parseMany(recordings                      = [])              {
+    if (!recordings) {
+      return [];
+    }
     return recordings.map(item => Recording.parse(item));
   }
 
@@ -3731,6 +3750,10 @@ class CallLog {
   }
 
   static parseMany(plain          )                 {
+    if (!plain || !plain.items) {
+      return [];
+    }
+
     return plain.items.map(item => CallLog.parse(item));
   }
 
@@ -4300,6 +4323,10 @@ class CTIPhone extends Emitter                  {
     return false;
   }
 
+  hasAVideoTrack()          {
+    return false;
+  }
+
   getLocalStreamForCall()               {
     return null;
   }
@@ -4693,6 +4720,9 @@ class ChatRoom {
                          
 
   static parseMany(plain                  )                  {
+    if (!plain || !plain.items) {
+      return [];
+    }
     return plain.items.map(item => ChatRoom.parse(item));
   }
 
@@ -5938,10 +5968,6 @@ 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__default['default'].parse(sdp);
   const videoMedia = parsedSdp.media.find(media => media.type === 'video');
@@ -5994,8 +6020,6 @@ 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        )                       => {
@@ -6185,38 +6209,6 @@ class WazoSessionDescriptionHandler extends sessionDescriptionHandler.SessionDes
     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)                {
@@ -8632,25 +8624,25 @@ 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);
       }
       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);
       }
       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
@@ -8658,16 +8650,16 @@ 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);
       }
     });
 
@@ -9253,18 +9245,20 @@ 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,