@wazo/sdk 0.37.19 → 0.37.23

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/service/IssueReporter.js

@@ -64,6 +64,7 @@ class IssueReporter {
   bufferTimeout: ?TimeoutID;
   _boundProcessBuffer: Function;
   _boundParseLoggerBody: Function;
+  _callback: ?Function;
 
   constructor() {
     addLevelsTo(this);
@@ -75,12 +76,17 @@ class IssueReporter {
     this.bufferTimeout = null;
     this._boundProcessBuffer = this._processBuffer.bind(this);
     this._boundParseLoggerBody = this._parseLoggerBody.bind(this);
+    this._callback = null;
   }
 
   init() {
     this._catchConsole();
   }
 
+  setCallback(cb: Function) {
+    this._callback = cb;
+  }
+
   configureRemoteClient(configuration: Object = { tag: 'wazo-sdk', host: null, port: null, level: null, extra: {} }) {
     this.remoteClientConfiguration = configuration;
   }
@@ -116,6 +122,7 @@ class IssueReporter {
 
     // Handle category label
     let category = null;
+    let skipSendToRemote = false;
     let extra = {};
     if (args[0].indexOf(CATEGORY_PREFIX) === 0) {
       category = args[0].split('=')[1];
@@ -131,6 +138,7 @@ class IssueReporter {
           errorMessage: lastArg.message,
           errorStack: lastArg.stack,
           errorType: lastArg.constructor.name,
+          skipSendToRemote: lastArg.skipSendToRemote,
         };
       } else {
         extra = lastArg;
@@ -139,6 +147,11 @@ class IssueReporter {
       args.splice(1, 1);
     }
 
+    if (extra.skipSendToRemote) {
+      skipSendToRemote = true;
+      delete extra.skipSendToRemote;
+    }
+
     const date = moment().format('YYYY-MM-DD HH:mm:ss.SSS');
     const message = args.join(', ');
     let consoleMessage = message;
@@ -159,7 +172,13 @@ class IssueReporter {
     const oldMethod = this.oldConsoleMethods[consoleLevel] || this.oldConsoleMethods.log;
     oldMethod.apply(oldMethod, [date, consoleMessage]);
 
-    this._sendToRemoteLogger(level, { date, message, category, ...extra });
+    if (this._callback) {
+      this._callback(level, consoleMessage);
+    }
+
+    if (!skipSendToRemote) {
+      this._sendToRemoteLogger(level, { date, message, category, ...extra });
+    }
   }
 
   logRequest(url: string, options: Object, response: Object, start: Date) {
@@ -304,7 +323,11 @@ class IssueReporter {
       },
       // $FlowFixMe
       body,
-    }).catch(() => {
+    }).catch((e: Error) => {
+      // $FlowFixMe
+      e.skipSendToRemote = true;
+      this.log('error', this._makeCategory('grafana'), 'Sending log to grafana, error', e);
+
       setTimeout(() => {
         if (isArray) {
           payload = payload.map(message => {

package/src/simple/Phone.js

@@ -347,6 +347,11 @@ class Phone extends Emitter {
     return this.getRemoteVideoStream(callSession);
   }
 
+  //  Useful in a react-native environment when remoteMediaStream is not updated
+  getRemoteVideoStreamFromPc(callSession: CallSession) {
+    return this.phone ? this.phone.getRemoteVideoStreamFromPc(callSession) : false;
+  }
+
   hasVideo(callSession: CallSession): boolean {
     return this.phone ? this.phone.hasVideo(callSession) : false;
   }

package/src/types/WebSocketMessage.js

@@ -7,6 +7,7 @@ import type {
 } from '../domain/SwitchboardCall';
 import type { ChatMessageResponse } from '../domain/ChatMessage';
 import ChatRoom from '../domain/ChatRoom';
+import Meeting from '../domain/Meeting';
 
 type WebSocketBaseMessage = {
   op: string,
@@ -187,6 +188,8 @@ type AgentStatusUpdate = WebSocketBaseMessage & { data: AgentStatusUpdateEvent,
 type AgentPaused = WebSocketBaseMessage & { data: any, name: 'agent_paused' };
 type AgentUnpaused = WebSocketBaseMessage & { data: any, name: 'agent_unpaused' };
 
+type MeetingUserProgress = WebSocketBaseMessage & { data: Meeting, name: 'meeting_user_progress' };
+
 export type WebSocketMessage =
   | CallCreated
   | CallUpdated
@@ -219,4 +222,5 @@ export type WebSocketMessage =
   | FaxOutboundSucceeded
   | AgentStatusUpdate
   | AgentPaused
-  | AgentUnpaused;
+  | AgentUnpaused
+  | MeetingUserProgress;

package/src/web-rtc-client.js

@@ -478,7 +478,9 @@ export default class WebRTCClient extends Emitter {
     const inviteOptions: InviterInviteOptions = {
       requestDelegate: {
         onAccept: (response: IncomingResponse) => {
-          session.sessionDescriptionHandler.peerConnection.sfu = conference;
+          if (session.sessionDescriptionHandler.peerConnection) {
+            session.sessionDescriptionHandler.peerConnection.sfu = conference;
+          }
           this._onAccepted(session, response.session, true);
         },
         onReject: (response: IncomingResponse) => {
@@ -1350,6 +1352,16 @@ export default class WebRTCClient extends Emitter {
     return this.hasRemoteVideo(sessionId) ? this.getRemoteStream(sessionId) : null;
   }
 
+  //  Useful in a react-native environment when remoteMediaStream is not updated
+  getRemoteVideoStreamFromPc(sessionId: string): ?MediaStream {
+    const pc = this.getPeerConnection(sessionId);
+    if (!pc) {
+      return null;
+    }
+
+    return pc.getRemoteStreams().find(stream => !!stream.getVideoTracks().length);
+  }
+
   hasVideo(sessionId: string): boolean {
     return this.hasLocalVideo(sessionId) || this.hasRemoteVideo(sessionId);
   }
@@ -1860,6 +1872,7 @@ export default class WebRTCClient extends Emitter {
     session.sessionDescriptionHandler.remoteMediaStream.onaddtrack = onTrack;
 
     pc.oniceconnectionstatechange = () => {
+      logger.info('on ice connection state changed', { state: pc.iceConnectionState });
       if (pc.iceConnectionState === 'disconnected') {
         this.eventEmitter.emit(ON_DISCONNECTED);
       }
@@ -2088,25 +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
@@ -2114,32 +2127,36 @@ 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);
       }
     });
 
     networkStats.packetsLost = packetsLost;
-    networkStats.audioBytesSent = audioBytesSent;
+    networkStats.totalAudioBytesSent = audioBytesSent;
+    networkStats.audioBytesSent = audioBytesSent - lastAudioSent;
+    networkStats.totalAudioBytesReceived = audioBytesReceived - lastAudioReceived;
     networkStats.audioBytesReceived = audioBytesReceived;
-    networkStats.videoBytesSent = videoBytesSent;
-    networkStats.videoBytesReceived = videoBytesReceived;
+    networkStats.totalVideoBytesSent = videoBytesSent;
+    networkStats.videoBytesSent = videoBytesSent - lastVideoSent;
+    networkStats.totalVideoBytesReceived = videoBytesReceived;
+    networkStats.videoBytesReceived = videoBytesReceived - lastVideoReceived;
+    networkStats.totalTransportSent = transportSent - lastTransportSent;
     networkStats.transportSent = transportSent;
-    networkStats.transportReceived = transportReceived;
-
-    const bandwidth = (networkStats.audioBytesSent - lastAudioSent) + (networkStats.videoBytesSent - lastVideoSent)
-      + (networkStats.audioBytesReceived - lastAudioReceived) + (networkStats.videoBytesReceived - lastVideoReceived)
-      + (networkStats.transportReceived - lastTransportReceived) + (networkStats.transportSent - lastTransportSent);
+    networkStats.totalTransportReceived = transportReceived;
+    networkStats.transportReceived = transportReceived - lastTransportReceived;
 
-    networkStats.bandwidth = bandwidth;
+    networkStats.bandwidth = networkStats.audioBytesSent + networkStats.audioBytesReceived
+      + networkStats.videoBytesSent + networkStats.videoBytesReceived + networkStats.transportReceived
+      + networkStats.transportSent;
 
     this.eventEmitter.emit(ON_NETWORK_STATS, session, networkStats, this.sessionNetworkStats[sessionId]);
     this.sessionNetworkStats[sessionId].push(networkStats);

package/src/websocket-client.js

@@ -93,6 +93,7 @@ export const AGENT_PAUSED = 'agent_paused';
 export const AGENT_UNPAUSED = 'agent_unpaused';
 export const CONFERENCE_ADHOC_PARTICIPANT_LEFT = 'conference_adhoc_participant_left';
 export const CONFERENCE_ADHOC_DELETED = 'conference_adhoc_deleted';
+export const MEETING_USER_PROGRESS = 'meeting_user_progress';
 
 const BLACKLIST_EVENTS = [
   CHAT_MESSAGE_SENT,
@@ -446,6 +447,7 @@ WebSocketClient.eventLists = [
   AGENT_UNPAUSED,
   CONFERENCE_ADHOC_PARTICIPANT_LEFT,
   CONFERENCE_ADHOC_DELETED,
+  MEETING_USER_PROGRESS,
 ];
 
 export default WebSocketClient;