@wazo/sdk 0.37.16 → 0.37.20

package/lib/web-rtc-client.js

@@ -675,7 +675,7 @@ class IssueReporter {
           payload._retry = retry + 1;
         }
 
-        this._sendDebugToGrafana(body, retry + 1);
+        this._sendDebugToGrafana(payload, retry + 1);
       }, 5000 + retry * 1000);
     });
   }
@@ -3177,7 +3177,8 @@ class ExternalApp {
                              
                
                
-                
+                 
+                      
  
 
 class Meeting {
@@ -3189,6 +3190,7 @@ class Meeting {
                
                
                  
+                      
                             
 
   static parse(plain                         )          {
@@ -3200,6 +3202,7 @@ class Meeting {
       port: plain.port,
       uuid: plain.uuid,
       number: plain.number,
+      persistent: plain.persistent,
     });
   }
 
@@ -3207,7 +3210,7 @@ class Meeting {
     return newFrom(meeting, Meeting);
   }
 
-  constructor({ uuid, name, guestSipAuthorization, ownerUuids, port, uri, number }         = {}) {
+  constructor({ uuid, name, guestSipAuthorization, ownerUuids, port, uri, number, persistent }         = {}) {
     this.guestSipAuthorization = guestSipAuthorization;
     this.uri = uri;
     this.name = name;
@@ -3215,6 +3218,7 @@ class Meeting {
     this.port = port;
     this.uuid = uuid;
     this.number = number;
+    this.persistent = persistent;
 
     // Useful to compare instead of instanceof with minified code
     this.type = 'Meeting';
@@ -3778,7 +3782,7 @@ class Recording {
               
                
 
-  static parseMany(recordings                     )              {
+  static parseMany(recordings                      = [])              {
     return recordings.map(item => Recording.parse(item));
   }
 
@@ -3904,7 +3908,7 @@ class CallLog {
       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),
+      recordings: Recording.parseMany(plain.recordings || []),
     });
   }
 
@@ -5299,6 +5303,7 @@ const ON_REINVITE = 'reinvite';
 const ON_ERROR = 'onError';
 const ON_SCREEN_SHARING_REINVITE = 'onScreenSharingReinvite';
 const ON_NETWORK_STATS = 'onNetworkStats';
+const ON_DISCONNECTED = 'onDisconnected';
 
 const events = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
 const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
@@ -5375,6 +5380,7 @@ class WebRTCClient extends Emitter {
                    
                                      
                            
+                          
 
   static isAPrivateIp(ip        )          {
     const regex = /^(?:10|127|172\.(?:1[6-9]|2[0-9]|3[01])|192\.168)\..*/;
@@ -5440,6 +5446,7 @@ class WebRTCClient extends Emitter {
     this.ON_ERROR = ON_ERROR;
     this.ON_SCREEN_SHARING_REINVITE = ON_SCREEN_SHARING_REINVITE;
     this.ON_NETWORK_STATS = ON_NETWORK_STATS;
+    this.ON_DISCONNECTED = ON_DISCONNECTED;
   }
 
   configureMedia(media             ) {
@@ -6041,7 +6048,7 @@ class WebRTCClient extends Emitter {
       // We search for the last transceiver without `video-` in the mid (video- means remote transceiver)
       const transceivers = pc.getTransceivers();
       const transceiverIdx = lastIndexOf(transceivers, transceiver =>
-        transceiver.sender.track === null && transceiver.mid.indexOf('video') === -1);
+        transceiver.sender.track === null && transceiver.mid && transceiver.mid.indexOf('video') === -1);
 
       videoSender = transceiverIdx !== -1 ? transceivers[transceiverIdx].sender : null;
     } else {
@@ -6287,10 +6294,13 @@ class WebRTCClient extends Emitter {
       return null;
     }
 
-    // let's update the local value
-    if (this.audio && this.audio.deviceId) {
-      // $FlowFixMe
-      this.audio.deviceId.exact = id;
+    // let's update the local audio value
+    if (this.audio) {
+      this.audio = {
+        deviceId: {
+          exact: id,
+        },
+      };
     }
 
     if (session) {
@@ -6303,6 +6313,9 @@ class WebRTCClient extends Emitter {
         const sender = pc && pc.getSenders().find(s => audioTrack && s && s.track && s.track.kind === audioTrack.kind);
 
         if (sender) {
+          if (sender.track) {
+            audioTrack.enabled = sender.track.enabled;
+          }
           sender.replaceTrack(audioTrack);
         }
 
@@ -6327,7 +6340,7 @@ class WebRTCClient extends Emitter {
       return null;
     }
 
-    // let's update the local value
+    // let's update the local video value
     if (this.video) {
       this.video = {
         deviceId: {
@@ -6362,12 +6375,21 @@ class WebRTCClient extends Emitter {
     return navigator.mediaDevices.getUserMedia(constraints).then(async stream => {
       const videoTrack = stream.getVideoTracks()[0];
       let sender = pc && pc.getSenders().find(s => videoTrack && s && s.track && s.track.kind === videoTrack.kind);
+      let wasTrackEnabled = false;
       if (!sender) {
         sender = pc && pc.getSenders().find(s => !s.track);
       }
 
       if (sender) {
-        sender.replaceTrack(videoTrack);
+        // No video track means video not enabled
+        wasTrackEnabled = sender.track ? sender.track.enabled : false;
+        videoTrack.enabled = wasTrackEnabled;
+
+        if (!wasTrackEnabled) {
+          videoTrack.stop();
+        }
+
+        sender.replaceTrack(wasTrackEnabled ? videoTrack : null);
       }
 
       // let's update the local stream
@@ -7051,6 +7073,7 @@ class WebRTCClient extends Emitter {
 
     this.updateRemoteStream(this.getSipSessionId(session));
 
+    const pc = session.sessionDescriptionHandler.peerConnection;
     const onTrack = (event     ) => {
       const isAudioOnly = this._isAudioOnly(session);
       const { kind, label, readyState, id, muted } = event.track;
@@ -7070,6 +7093,12 @@ class WebRTCClient extends Emitter {
 
     session.sessionDescriptionHandler.remoteMediaStream.onaddtrack = onTrack;
 
+    pc.oniceconnectionstatechange = () => {
+      if (pc.iceConnectionState === 'disconnected') {
+        this.eventEmitter.emit(ON_DISCONNECTED);
+      }
+    };
+
     if (withEvent) {
       this.eventEmitter.emit(ACCEPTED, session, sessionDialog);
     }
@@ -7281,25 +7310,33 @@ class WebRTCClient extends Emitter {
     const lastVideoSent = lastNetworkStats ? lastNetworkStats.videoBytesSent : 0;
     const lastAudioReceived = lastNetworkStats ? lastNetworkStats.audioBytesReceived : 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 videoBytesSent = 0;
     let videoBytesReceived = 0;
+    let transportSent = 0;
+    let transportReceived = 0;
+    let packetsLost = 0;
 
     stats.forEach(report => {
       if (report.type === 'outbound-rtp' && report.kind === 'audio') {
+        audioBytesSent += report.bytesSent + report.headerBytesSent;
+      }
+      if (report.type === 'remote-outbound-rtp' && report.kind === 'audio') {
         audioBytesSent += report.bytesSent;
       }
       if (report.type === 'outbound-rtp' && report.kind === 'video') {
-        videoBytesSent += report.bytesSent;
+        videoBytesSent += report.bytesSent + report.headerBytesSent;
       }
       if (report.type === 'inbound-rtp' && report.kind === 'audio') {
-        networkStats.packetsLost = report.packetsLost;
+        packetsLost += report.packetsLost;
         networkStats.packetsReceived = report.packetsReceived;
-        audioBytesReceived += report.bytesReceived;
+        audioBytesReceived += report.bytesReceived + report.headerBytesReceived;
       }
       if (report.type === 'inbound-rtp' && report.kind === 'video') {
-        videoBytesReceived += report.bytesReceived;
+        videoBytesReceived += report.bytesReceived + report.headerBytesReceived;
       }
       if (report.type === 'outbound-rtp' && report.kind === 'video') {
         if ('framesPerSecond' in report) {
@@ -7311,19 +7348,32 @@ class WebRTCClient extends Emitter {
         }
       }
       if (report.type === 'remote-inbound-rtp' && report.kind === 'audio') {
+        packetsLost += report.packetsLost;
         networkStats.roundTripTime = report.roundTripTime;
         networkStats.jitter = report.jitter;
       }
+      if (report.type === 'remote-inbound-rtp' && report.kind === 'video') {
+        packetsLost += report.packetsLost;
+      }
+      if (report.type === 'transport') {
+        transportSent += report.bytesSent;
+        transportReceived += report.bytesReceived;
+      }
     });
 
+    networkStats.packetsLost = packetsLost;
     networkStats.audioBytesSent = audioBytesSent;
     networkStats.audioBytesReceived = audioBytesReceived;
     networkStats.videoBytesSent = videoBytesSent;
     networkStats.videoBytesReceived = videoBytesReceived;
+    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.bandwidth = (networkStats.audioBytesSent - lastAudioSent)
-      + (networkStats.videoBytesSent - lastVideoSent) + (networkStats.audioBytesReceived - lastAudioReceived)
-      + (networkStats.videoBytesReceived - lastVideoReceived);
+    networkStats.bandwidth = bandwidth;
 
     this.eventEmitter.emit(ON_NETWORK_STATS, session, networkStats, this.sessionNetworkStats[sessionId]);
     this.sessionNetworkStats[sessionId].push(networkStats);

package/lib/websocket-client.js

@@ -383,7 +383,7 @@ class IssueReporter {
           payload._retry = retry + 1;
         }
 
-        this._sendDebugToGrafana(body, retry + 1);
+        this._sendDebugToGrafana(payload, retry + 1);
       }, 5000 + retry * 1000);
     });
   }

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@wazo/sdk",
-  "version": "0.37.16",
+  "version": "0.37.20",
   "description": "Wazo's JavaScript Software Development Kit.",
   "main": "dist/wazo-sdk.js",
   "author": "Wazo (http://wazo.io)",
@@ -47,7 +47,7 @@
   "dependencies": {
     "events": "^3.3.0",
     "fstream": "^1.0.12",
-    "getstats": "manuquentin/getStats#fix_undefined_results",
+    "getstats": "manuquentin/getStats#fix_stats_mobile",
     "globby": "^11.0.3",
     "google-libphonenumber": "^3.2.18",
     "js-base64": "^3.6.0",

package/src/domain/CallLog.js

@@ -108,7 +108,7 @@ export default class CallLog {
       duration: (plain.duration || 0) * 1000, // duration is in seconds
       start: moment(plain.start).toDate(),
       end: plain.end ? moment(plain.end).toDate() : null,
-      recordings: Recording.parseMany(plain.recordings),
+      recordings: Recording.parseMany(plain.recordings || []),
     });
   }
 

package/src/domain/Meeting.js

@@ -8,7 +8,8 @@ export type MeetingCreationResponse = {
   owner_uuids: Array<string>,
   port: string,
   uuid: string,
-  number: string
+  number: string,
+  persistent: boolean,
 }
 
 export default class Meeting {
@@ -20,6 +21,7 @@ export default class Meeting {
   name: string;
   port: string;
   number: string;
+  persistent: boolean;
   ownerUuids: Array<string>;
 
   static parse(plain: MeetingCreationResponse): Meeting {
@@ -31,6 +33,7 @@ export default class Meeting {
       port: plain.port,
       uuid: plain.uuid,
       number: plain.number,
+      persistent: plain.persistent,
     });
   }
 
@@ -38,7 +41,7 @@ export default class Meeting {
     return newFrom(meeting, Meeting);
   }
 
-  constructor({ uuid, name, guestSipAuthorization, ownerUuids, port, uri, number }: Object = {}) {
+  constructor({ uuid, name, guestSipAuthorization, ownerUuids, port, uri, number, persistent }: Object = {}) {
     this.guestSipAuthorization = guestSipAuthorization;
     this.uri = uri;
     this.name = name;
@@ -46,6 +49,7 @@ export default class Meeting {
     this.port = port;
     this.uuid = uuid;
     this.number = number;
+    this.persistent = persistent;
 
     // Useful to compare instead of instanceof with minified code
     this.type = 'Meeting';

package/src/domain/Phone/WebRTCPhone.js

@@ -52,6 +52,7 @@ export const ON_MESSAGE_TRACK_UPDATED = 'onTrackUpdated';
 export const ON_NETWORK_STATS = 'onNetworkStats';
 export const ON_CHAT = 'phone/ON_CHAT';
 export const ON_SIGNAL = 'phone/ON_SIGNAL';
+export const ON_DISCONNECTED = 'onDisconnected';
 
 export const MESSAGE_TYPE_CHAT = 'message/TYPE_CHAT';
 export const MESSAGE_TYPE_SIGNAL = 'message/TYPE_SIGNAL';
@@ -91,6 +92,7 @@ export const events = [
   ON_CHAT,
   ON_SIGNAL,
   ON_NETWORK_STATS,
+  ON_DISCONNECTED,
 ];
 
 const logger = IssueReporter.loggerFor('webrtc-phone');
@@ -1445,6 +1447,11 @@ export default class WebRTCPhone extends Emitter implements Phone {
       this.eventEmitter.emit(ON_UNREGISTERED);
     });
 
+    this.client.on(this.client.ON_DISCONNECTED, () => {
+      logger.info('WebRTC disconnected');
+      this.eventEmitter.emit(ON_DISCONNECTED);
+    });
+
     this.client.on(this.client.REGISTERED, () => {
       logger.info('WebRTC registered', {
         shouldSendReinvite: this.shouldSendReinvite,

package/src/domain/Recording.js

@@ -24,7 +24,7 @@ export default class Recording {
   start: Date;
   uuid: string;
 
-  static parseMany(recordings: RecordingResponse[]): Recording[] {
+  static parseMany(recordings: RecordingResponse[] = []): Recording[] {
     return recordings.map(item => Recording.parse(item));
   }
 

package/src/service/IssueReporter.js

@@ -316,7 +316,7 @@ class IssueReporter {
           payload._retry = retry + 1;
         }
 
-        this._sendDebugToGrafana(body, retry + 1);
+        this._sendDebugToGrafana(payload, retry + 1);
       }, 5000 + retry * 1000);
     });
   }

package/src/web-rtc-client.js

@@ -69,6 +69,7 @@ const ON_REINVITE = 'reinvite';
 const ON_ERROR = 'onError';
 const ON_SCREEN_SHARING_REINVITE = 'onScreenSharingReinvite';
 const ON_NETWORK_STATS = 'onNetworkStats';
+const ON_DISCONNECTED = 'onDisconnected';
 
 export const events = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
 export const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
@@ -145,6 +146,7 @@ export default class WebRTCClient extends Emitter {
   ON_ERROR: string;
   ON_SCREEN_SHARING_REINVITE: string;
   ON_NETWORK_STATS: string;
+  ON_DISCONNECTED: string;
 
   static isAPrivateIp(ip: string): boolean {
     const regex = /^(?:10|127|172\.(?:1[6-9]|2[0-9]|3[01])|192\.168)\..*/;
@@ -210,6 +212,7 @@ export default class WebRTCClient extends Emitter {
     this.ON_ERROR = ON_ERROR;
     this.ON_SCREEN_SHARING_REINVITE = ON_SCREEN_SHARING_REINVITE;
     this.ON_NETWORK_STATS = ON_NETWORK_STATS;
+    this.ON_DISCONNECTED = ON_DISCONNECTED;
   }
 
   configureMedia(media: MediaConfig) {
@@ -811,7 +814,7 @@ export default class WebRTCClient extends Emitter {
       // We search for the last transceiver without `video-` in the mid (video- means remote transceiver)
       const transceivers = pc.getTransceivers();
       const transceiverIdx = lastIndexOf(transceivers, transceiver =>
-        transceiver.sender.track === null && transceiver.mid.indexOf('video') === -1);
+        transceiver.sender.track === null && transceiver.mid && transceiver.mid.indexOf('video') === -1);
 
       videoSender = transceiverIdx !== -1 ? transceivers[transceiverIdx].sender : null;
     } else {
@@ -1057,10 +1060,13 @@ export default class WebRTCClient extends Emitter {
       return null;
     }
 
-    // let's update the local value
-    if (this.audio && this.audio.deviceId) {
-      // $FlowFixMe
-      this.audio.deviceId.exact = id;
+    // let's update the local audio value
+    if (this.audio) {
+      this.audio = {
+        deviceId: {
+          exact: id,
+        },
+      };
     }
 
     if (session) {
@@ -1073,6 +1079,9 @@ export default class WebRTCClient extends Emitter {
         const sender = pc && pc.getSenders().find(s => audioTrack && s && s.track && s.track.kind === audioTrack.kind);
 
         if (sender) {
+          if (sender.track) {
+            audioTrack.enabled = sender.track.enabled;
+          }
           sender.replaceTrack(audioTrack);
         }
 
@@ -1097,7 +1106,7 @@ export default class WebRTCClient extends Emitter {
       return null;
     }
 
-    // let's update the local value
+    // let's update the local video value
     if (this.video) {
       this.video = {
         deviceId: {
@@ -1132,12 +1141,21 @@ export default class WebRTCClient extends Emitter {
     return navigator.mediaDevices.getUserMedia(constraints).then(async stream => {
       const videoTrack = stream.getVideoTracks()[0];
       let sender = pc && pc.getSenders().find(s => videoTrack && s && s.track && s.track.kind === videoTrack.kind);
+      let wasTrackEnabled = false;
       if (!sender) {
         sender = pc && pc.getSenders().find(s => !s.track);
       }
 
       if (sender) {
-        sender.replaceTrack(videoTrack);
+        // No video track means video not enabled
+        wasTrackEnabled = sender.track ? sender.track.enabled : false;
+        videoTrack.enabled = wasTrackEnabled;
+
+        if (!wasTrackEnabled) {
+          videoTrack.stop();
+        }
+
+        sender.replaceTrack(wasTrackEnabled ? videoTrack : null);
       }
 
       // let's update the local stream
@@ -1821,6 +1839,7 @@ export default class WebRTCClient extends Emitter {
 
     this.updateRemoteStream(this.getSipSessionId(session));
 
+    const pc = session.sessionDescriptionHandler.peerConnection;
     const onTrack = (event: any) => {
       const isAudioOnly = this._isAudioOnly(session);
       const { kind, label, readyState, id, muted } = event.track;
@@ -1840,6 +1859,12 @@ export default class WebRTCClient extends Emitter {
 
     session.sessionDescriptionHandler.remoteMediaStream.onaddtrack = onTrack;
 
+    pc.oniceconnectionstatechange = () => {
+      if (pc.iceConnectionState === 'disconnected') {
+        this.eventEmitter.emit(ON_DISCONNECTED);
+      }
+    };
+
     if (withEvent) {
       this.eventEmitter.emit(ACCEPTED, session, sessionDialog);
     }
@@ -2051,25 +2076,33 @@ export default class WebRTCClient extends Emitter {
     const lastVideoSent = lastNetworkStats ? lastNetworkStats.videoBytesSent : 0;
     const lastAudioReceived = lastNetworkStats ? lastNetworkStats.audioBytesReceived : 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 videoBytesSent = 0;
     let videoBytesReceived = 0;
+    let transportSent = 0;
+    let transportReceived = 0;
+    let packetsLost = 0;
 
     stats.forEach(report => {
       if (report.type === 'outbound-rtp' && report.kind === 'audio') {
+        audioBytesSent += report.bytesSent + report.headerBytesSent;
+      }
+      if (report.type === 'remote-outbound-rtp' && report.kind === 'audio') {
         audioBytesSent += report.bytesSent;
       }
       if (report.type === 'outbound-rtp' && report.kind === 'video') {
-        videoBytesSent += report.bytesSent;
+        videoBytesSent += report.bytesSent + report.headerBytesSent;
       }
       if (report.type === 'inbound-rtp' && report.kind === 'audio') {
-        networkStats.packetsLost = report.packetsLost;
+        packetsLost += report.packetsLost;
         networkStats.packetsReceived = report.packetsReceived;
-        audioBytesReceived += report.bytesReceived;
+        audioBytesReceived += report.bytesReceived + report.headerBytesReceived;
       }
       if (report.type === 'inbound-rtp' && report.kind === 'video') {
-        videoBytesReceived += report.bytesReceived;
+        videoBytesReceived += report.bytesReceived + report.headerBytesReceived;
       }
       if (report.type === 'outbound-rtp' && report.kind === 'video') {
         if ('framesPerSecond' in report) {
@@ -2081,19 +2114,32 @@ 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;
       }
+      if (report.type === 'remote-inbound-rtp' && report.kind === 'video') {
+        packetsLost += report.packetsLost;
+      }
+      if (report.type === 'transport') {
+        transportSent += report.bytesSent;
+        transportReceived += report.bytesReceived;
+      }
     });
 
+    networkStats.packetsLost = packetsLost;
     networkStats.audioBytesSent = audioBytesSent;
     networkStats.audioBytesReceived = audioBytesReceived;
     networkStats.videoBytesSent = videoBytesSent;
     networkStats.videoBytesReceived = videoBytesReceived;
+    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.bandwidth = (networkStats.audioBytesSent - lastAudioSent)
-      + (networkStats.videoBytesSent - lastVideoSent) + (networkStats.audioBytesReceived - lastAudioReceived)
-      + (networkStats.videoBytesReceived - lastVideoReceived);
+    networkStats.bandwidth = bandwidth;
 
     this.eventEmitter.emit(ON_NETWORK_STATS, session, networkStats, this.sessionNetworkStats[sessionId]);
     this.sessionNetworkStats[sessionId].push(networkStats);