@wazo/sdk 0.37.22 → 0.37.23

package/src/domain/CallLog.js

@@ -84,6 +84,10 @@ export default class CallLog {
   }
 
   static parseMany(plain: Response): Array<CallLog> {
+    if (!plain || !plain.items) {
+      return [];
+    }
+
     return plain.items.map(item => CallLog.parse(item));
   }
 

package/src/domain/ChatMessage.js

@@ -42,6 +42,10 @@ export default class ChatMessage {
   read: boolean;
 
   static parseMany(plain: ChatMessageListResponse): Array<ChatMessage> {
+    if (!plain || !plain.items) {
+      return [];
+    }
+
     return plain.items.map(item => ChatMessage.parse(item));
   }
 

package/src/domain/ChatRoom.js

@@ -22,6 +22,9 @@ export default class ChatRoom {
   users: Array<ChatUser>;
 
   static parseMany(plain: RootListResponse): Array<ChatRoom> {
+    if (!plain || !plain.items) {
+      return [];
+    }
     return plain.items.map(item => ChatRoom.parse(item));
   }
 

package/src/domain/Contact.js

@@ -281,6 +281,9 @@ export default class Contact {
   }
 
   static parseMany(response: ContactsResponse): Array<Contact> {
+    if (!response || !response.results) {
+      return [];
+    }
     return response.results.map(r => Contact.parse(r, response.column_types));
   }
 

package/src/domain/Meeting.js

@@ -38,6 +38,9 @@ export default class Meeting {
   }
 
   static parseMany(items: MeetingCreationResponse[]): Array<Meeting> {
+    if (!items) {
+      return [];
+    }
     return items.map(plain => Meeting.parse(plain));
   }
 

package/src/domain/Phone/CTIPhone.js

@@ -280,6 +280,10 @@ export default class CTIPhone extends Emitter implements Phone {
     return false;
   }
 
+  hasAVideoTrack(): boolean {
+    return false;
+  }
+
   getLocalStreamForCall(): ?MediaStream {
     return null;
   }

package/src/domain/Phone/Phone.js

@@ -80,6 +80,8 @@ export interface Phone {
 
   hasVideo(callSession: CallSession): boolean;
 
+  hasAVideoTrack(callSession: CallSession): boolean;
+
   isWebRTC(): boolean;
 
   getUserAgent(): string;

package/src/domain/Phone/WebRTCPhone.js

@@ -571,18 +571,20 @@ export default class WebRTCPhone extends Emitter implements Phone {
       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),
-      );
-    };
+    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,

package/src/domain/Recording.js

@@ -25,6 +25,9 @@ export default class Recording {
   uuid: string;
 
   static parseMany(recordings: RecordingResponse[] = []): Recording[] {
+    if (!recordings) {
+      return [];
+    }
     return recordings.map(item => Recording.parse(item));
   }
 

package/src/domain/Voicemail.js

@@ -61,6 +61,9 @@ export default class Voicemail {
   }
 
   static parseMany(plain: Response): Array<Voicemail> {
+    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;
 

package/src/lib/WazoSessionDescriptionHandler.js

@@ -12,11 +12,9 @@ import { SessionDescriptionHandler }
 import { SessionDescriptionHandlerOptions }
   from 'sip.js/lib/platform/web/session-description-handler/session-description-handler-options';
 import IssueReporter from '../service/IssueReporter';
-import { areCandidateValid, fixSdp, parseCandidate } from '../utils/sdp';
+import { fixSdp, parseCandidate } from '../utils/sdp';
 
 const wazoLogger = IssueReporter ? IssueReporter.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
 export const wazoMediaStreamFactory = (constraints: Object): Promise<MediaStream> => {
@@ -206,38 +204,6 @@ class WazoSessionDescriptionHandler extends SessionDescriptionHandler {
     this._peerConnection = undefined;
   }
 
-  async _waitForValidGatheredIce(): Object {
-    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?: SessionDescriptionHandlerOptions, isConference: boolean = false,
     audioOnly: boolean = false): Promise<void> {

package/src/web-rtc-client.js

@@ -2101,25 +2101,25 @@ export default 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
@@ -2127,16 +2127,16 @@ export default 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);
       }
     });