@wazo/sdk 0.37.24 → 0.37.25

package/lib/websocket-client.js

@@ -45,7 +45,7 @@ const isMobile = () => typeof navigator !== 'undefined' && navigator.product ===
 
 /*       */
 
-const logger = IssueReporter$1 ? IssueReporter$1.loggerFor('api') : console;
+IssueReporter$1 ? IssueReporter$1.loggerFor('api') : console;
 
 // Use a function here to be able to mock it in tests
 const realFetch = () => {
@@ -621,7 +621,7 @@ const BLACKLIST_EVENTS = [
 ];
 
 const HEARTBEAT_ENGINE_VERSION = '20.09';
-const logger$1 = IssueReporter$1.loggerFor('wazo-ws');
+const logger = IssueReporter$1.loggerFor('wazo-ws');
 const messageLogger = IssueReporter$1.loggerFor('wazo-ws-message');
 
 class WebSocketClient extends Emitter {
@@ -670,19 +670,19 @@ class WebSocketClient extends Emitter {
   }
 
   connect() {
-    logger$1.info('connect method started', { host: this.host, token: this.token });
+    logger.info('connect method started', { host: this.host, token: this.token });
     this.socket = new ReconnectingWebSocket__default['default'](this._getUrl.bind(this), [], this.options);
     if (this.options.binaryType) {
       this.socket.binaryType = this.options.binaryType;
     }
 
     this.socket.onopen = () => {
-      logger$1.info('on Wazo WS open', { method: 'connect', host: this.host });
+      logger.info('on Wazo WS open', { method: 'connect', host: this.host });
       this.eventEmitter.emit(SOCKET_EVENTS.ON_OPEN);
     };
 
     this.socket.onerror = event => {
-      logger$1.error('on Wazo WS error', event.target);
+      logger.error('on Wazo WS error', event.target);
       this.eventEmitter.emit(SOCKET_EVENTS.ON_ERROR, event);
     };
 
@@ -712,7 +712,7 @@ class WebSocketClient extends Emitter {
 
     this.socket.onclose = event => {
       // Can't be converted to JSON (circular structure)
-      logger$1.info('on Wazo WS close', {
+      logger.info('on Wazo WS close', {
         reason: event.reason,
         code: event.code,
         readyState: event.target.readyState,
@@ -730,12 +730,12 @@ class WebSocketClient extends Emitter {
     };
 
     this.socket.onerror = event => {
-      logger$1.info('Wazo WS error', { message: event.message, code: event.code, readyState: event.target.readyState });
+      logger.info('Wazo WS error', { message: event.message, code: event.code, readyState: event.target.readyState });
     };
   }
 
   close(force          = false)       {
-    logger$1.info('Wazo WS close', { socket: !!this.socket, host: this.host, token: this.token });
+    logger.info('Wazo WS close', { socket: !!this.socket, host: this.host, token: this.token });
 
     if (!this.socket) {
       return;
@@ -753,7 +753,7 @@ class WebSocketClient extends Emitter {
 
   updateToken(token        ) {
     this.token = token;
-    logger$1.info('Wazo WS updating token', { url: this._getUrl(), token, socket: !!this.socket });
+    logger.info('Wazo WS updating token', { url: this._getUrl(), token, socket: !!this.socket });
 
     if (this.socket) {
       // If still connected, send the token to the WS
@@ -771,7 +771,7 @@ class WebSocketClient extends Emitter {
   }
 
   startHeartbeat() {
-    logger$1.info('Wazo WS start heartbeat', { host: this.host, token: this.token });
+    logger.info('Wazo WS start heartbeat', { host: this.host, token: this.token });
 
     if (!this.socket) {
       this.heartbeat.stop();
@@ -785,7 +785,7 @@ class WebSocketClient extends Emitter {
   }
 
   stopHeartbeat() {
-    logger$1.info('Wazo WS stop heartbeat', { host: this.host, token: this.token });
+    logger.info('Wazo WS stop heartbeat', { host: this.host, token: this.token });
 
     this.heartbeat.stop();
   }
@@ -815,7 +815,7 @@ class WebSocketClient extends Emitter {
   }
 
   reconnect(reason        ) {
-    logger$1.info('Wazo WS reconnect', { reason, socket: !!this.socket, host: this.host, token: this.token });
+    logger.info('Wazo WS reconnect', { reason, socket: !!this.socket, host: this.host, token: this.token });
     if (!this.socket) {
       return;
     }
@@ -871,7 +871,7 @@ class WebSocketClient extends Emitter {
       this.close();
     }
     const url = `wss://${this.host || ''}/api/websocketd/?token=${this.token || ''}&version=${this.version}`;
-    logger$1.log('Wazo WS url computed to reconnect', { url });
+    logger.log('Wazo WS url computed to reconnect', { url });
 
     return url;
   }
@@ -879,12 +879,12 @@ class WebSocketClient extends Emitter {
   _onHeartbeat(message        ) {
     if (message.payload === 'pong') {
       this.heartbeat.onHeartbeat();
-      logger$1.log('on heartbeat received from Wazo WS');
+      logger.log('on heartbeat received from Wazo WS');
     }
   }
 
   async _onHeartbeatTimeout() {
-    logger$1.log('heartbeat timed out', { host: this.host, token: this.token });
+    logger.log('heartbeat timed out', { host: this.host, token: this.token });
 
     this.close();
     this.eventEmitter.emit(SOCKET_EVENTS.ON_CLOSE, new Error('Websocket ping failure.'));

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@wazo/sdk",
-  "version": "0.37.24",
+  "version": "0.37.25",
   "description": "Wazo's JavaScript Software Development Kit.",
   "main": "dist/wazo-sdk.js",
   "author": "Wazo (http://wazo.io)",

package/src/domain/Phone/WebRTCPhone.js

@@ -203,6 +203,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       constraints,
       audioOnly,
       conference,
+      iceRestart,
     });
 
     if (!sipSession) {
@@ -614,7 +615,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
   }
 
   _onCallAccepted(sipSession: Session, cameraEnabled: boolean): CallSession {
-    logger.info('WebRTC phone -  on call accepted', { sipId: sipSession.id, cameraEnabled });
+    logger.info('WebRTC phone - on call accepted', { sipId: sipSession.id, cameraEnabled });
 
     const callSession = this._createAcceptedCallSession(sipSession, cameraEnabled);
     this.currentSipSession = sipSession;
@@ -1126,7 +1127,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
   // @TODO: line is not used here
   async makeCall(number: string, line: any, cameraEnabled?: boolean,
     audioOnly: boolean = false, conference: boolean = false): Promise<?CallSession> {
-    logger.info('make WebRTC call', { number, lineId: line ? line.id : null, cameraEnabled });
+    logger.info('make WebRTC call', { number, lineId: line ? line.id : null, cameraEnabled, audioOnly, conference });
     if (!number) {
       return Promise.resolve(null);
     }

package/src/web-rtc-client.js

@@ -453,7 +453,7 @@ export default class WebRTCClient extends Emitter {
   }
 
   call(number: string, enableVideo?: boolean, audioOnly: boolean = false, conference: boolean = false): Session {
-    logger.info('sdk webrtc creating call', { number, enableVideo, audioOnly });
+    logger.info('sdk webrtc creating call', { number, enableVideo, audioOnly, conference });
     const inviterOptions: Object = {
       sessionDescriptionHandlerOptionsReInvite: {
         conference,
@@ -1529,6 +1529,7 @@ export default 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.info('Retrieving media a configuration', { enableVideo, screenSharing, isDesktop, withAudio, constraints });
 
@@ -1536,7 +1537,8 @@ export default 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,
@@ -1871,12 +1873,14 @@ 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);
-      }
-    };
+    if (pc) {
+      pc.oniceconnectionstatechange = () => {
+        logger.info('on ice connection state changed', { state: pc.iceConnectionState });
+        if (pc.iceConnectionState === 'disconnected') {
+          this.eventEmitter.emit(ON_DISCONNECTED);
+        }
+      };
+    }
 
     if (withEvent) {
       this.eventEmitter.emit(ACCEPTED, session, sessionDialog);
@@ -2086,13 +2090,17 @@ export default 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;
@@ -2102,6 +2110,7 @@ export default class WebRTCClient extends Emitter {
     stats.forEach(report => {
       if (report.type === 'outbound-rtp' && report.kind === 'audio') {
         audioBytesSent += Number(report.bytesSent) + Number(report.headerBytesSent);
+        audioContentSent += Number(report.bytesSent);
       }
       if (report.type === 'remote-outbound-rtp' && report.kind === 'audio') {
         audioBytesSent += Number(report.bytesSent);
@@ -2113,6 +2122,7 @@ export default class WebRTCClient extends Emitter {
         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 += Number(report.bytesReceived) + Number(report.headerBytesReceived);
@@ -2141,10 +2151,17 @@ export default class WebRTCClient extends Emitter {
     });
 
     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;