@wazo/sdk 0.37.24 → 0.37.28

package/lib/websocket-client.js

@@ -45,11 +45,11 @@ 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 = () => {
-  if (typeof document !== 'undefined') {
+  if (typeof document !== 'undefined' && window.fetch) {
     // Browser
     return window.fetch;
   }
@@ -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": "0.37.28",
   "description": "Wazo's JavaScript Software Development Kit.",
   "main": "dist/wazo-sdk.js",
   "author": "Wazo (http://wazo.io)",

package/src/domain/CallSession.js

@@ -1,5 +1,6 @@
 // @flow
 import { SessionState } from 'sip.js/lib/api/session-state';
+import type { Session } from 'sip.js/lib/api/session';
 
 import Call from './Call';
 import newFrom from '../utils/new-from';
@@ -12,8 +13,10 @@ type CallSessionArguments = {
   callId: string,
   callerNumber: string,
   cameraEnabled: boolean,
+  creationTime?: ?Date,
   dialedExtension?: string | null,
   displayName: string,
+  endTime?: ?Date,
   isCaller: boolean,
   muted: boolean,
   videoMuted: boolean,
@@ -30,6 +33,7 @@ type CallSessionArguments = {
   screensharing: boolean,
   recording: boolean,
   recordingPaused: boolean,
+  sipSession?: Session,
 };
 
 export default class CallSession {
@@ -49,8 +53,12 @@ export default class CallSession {
 
   callerNumber: string;
 
+  creationTime: ?Date;
+
   startTime: number;
 
+  endTime: ?Date;
+
   isCaller: boolean;
 
   answered: boolean;
@@ -89,6 +97,8 @@ export default class CallSession {
 
   recordingPaused: boolean;
 
+  sipSession: Session;
+
   constructor({
     answered,
     answeredBySystem,
@@ -101,6 +111,8 @@ export default class CallSession {
     paused,
     ringing,
     startTime,
+    creationTime,
+    endTime,
     cameraEnabled,
     dialedExtension,
     sipCallId,
@@ -113,12 +125,15 @@ export default class CallSession {
     recording,
     recordingPaused,
     videoRemotelyDowngraded,
+    sipSession,
   }: CallSessionArguments) {
     this.callId = callId;
     this.sipCallId = sipCallId;
     this.displayName = displayName;
     this.number = number;
+    this.creationTime = creationTime;
     this.startTime = startTime;
+    this.endTime = endTime;
     this.isCaller = isCaller;
     this.answered = answered;
     this.answeredBySystem = answeredBySystem;
@@ -137,6 +152,7 @@ export default class CallSession {
     this.recording = recording || false;
     this.recordingPaused = recordingPaused || false;
     this.videoRemotelyDowngraded = videoRemotelyDowngraded;
+    this.sipSession = sipSession;
 
     // Useful to compare instead of instanceof with minified code
     this.type = 'CallSession';

package/src/domain/Contact.js

@@ -654,7 +654,7 @@ export default class Contact {
       this[key] = old[key] || this[key];
     });
 
-    if (old.lineState && !this.lineState) {
+    if (old.lineState) {
       this.lineState = old.lineState;
     }
 

package/src/domain/Phone/WebRTCPhone.js

@@ -30,6 +30,7 @@ export const ON_CAMERA_DISABLED = 'onCameraDisabled';
 export const ON_CAMERA_RESUMED = 'onCameraResumed';
 export const ON_CALL_CANCELED = 'onCallCanceled';
 export const ON_CALL_FAILED = 'onCallFailed';
+export const ON_CALL_REJECTED = 'onCallRejected';
 export const ON_CALL_ENDED = 'onCallEnded';
 export const ON_CALL_ENDING = 'onCallEnding';
 export const ON_MESSAGE = 'onMessage';
@@ -74,6 +75,7 @@ export const events = [
   ON_CAMERA_DISABLED,
   ON_CALL_FAILED,
   ON_CALL_ENDED,
+  ON_CALL_REJECTED,
   ON_MESSAGE,
   ON_REINVITE,
   ON_TRACK,
@@ -203,6 +205,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       constraints,
       audioOnly,
       conference,
+      iceRestart,
     });
 
     if (!sipSession) {
@@ -429,6 +432,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
 
           // Should be called before `_onCallTerminated` or the callCount will not decrement...
           const callSession = this._createCallSession(sipSession);
+          callSession.endTime = new Date();
           const wasCurrentSession = this._onCallTerminated(sipSession);
 
           return this.eventEmitter.emit(ON_CALL_ENDED, callSession, wasCurrentSession);
@@ -614,7 +618,7 @@ 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;
@@ -683,6 +687,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
   _onCallTerminated(sipSession: Session) {
     logger.info('WebRTC phone - on call terminated', { sipId: sipSession.id });
 
+    const sipSessionId = this.getSipSessionId(sipSession);
     const callSession = this._createCallSession(sipSession);
     const isCurrentSession = this.isCurrentCallSipSession(callSession);
     const isCurrentIncomingCall = callSession.is(this.getIncomingCallSession());
@@ -700,7 +705,6 @@ export default class WebRTCPhone extends Emitter implements Phone {
       }, 5);
     }
 
-    const sipSessionId = this.getSipSessionId(sipSession);
     if (sipSessionId) {
       this.removeIncomingSessions(sipSessionId);
     }
@@ -1126,7 +1130,7 @@ 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);
     }
@@ -1150,6 +1154,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       return Promise.resolve(null);
     }
     const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
+    callSession.creationTime = new Date();
 
     this.eventEmitter.emit(ON_PLAY_PROGRESS_SOUND, this.audioOutputDeviceId, this.audioOutputVolume);
 
@@ -1345,8 +1350,12 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
 
     const sessionId = this.incomingSessions[0];
+    const sipSession = this.client.getSipSession(sessionId);
+    if (!sipSession) {
+      return null;
+    }
 
-    return this._createCallSession(this.client.getSipSession(sessionId));
+    return this._createCallSession(sipSession);
   }
 
   sendMessage(sipSession: Session = null, body: string, contentType: string = 'text/plain') {
@@ -1424,6 +1433,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
           callSession);
       }
 
+      callSession.creationTime = new Date();
       this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
     });
 
@@ -1463,6 +1473,13 @@ export default class WebRTCPhone extends Emitter implements Phone {
       this.eventEmitter.emit(ON_CALL_ERROR, e);
     });
 
+    this.client.on(this.client.REJECTED, session => {
+      logger.info('WebRTC call rejected', session.id);
+      const callSession = this._createCallSession(session);
+      callSession.endTime = new Date();
+      this.eventEmitter.emit(ON_CALL_REJECTED, callSession);
+    });
+
     this.client.on(this.client.UNREGISTERED, () => {
       logger.info('WebRTC unregistered');
       this.eventEmitter.emit(ON_UNREGISTERED);
@@ -1677,6 +1694,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
     const number = identity ? identity.uri._normal.user : null;
     const { state } = sipSession || {};
     const sessionId = this.getSipSessionId(sipSession);
+    fromSession = fromSession || this.callSessions[sessionId];
 
     const callSession = new CallSession({
       callId: fromSession && fromSession.callId,
@@ -1684,6 +1702,8 @@ export default class WebRTCPhone extends Emitter implements Phone {
       sipStatus: state,
       displayName: identity ? identity.displayName || number : number,
       startTime: fromSession ? fromSession.startTime : new Date(),
+      creationTime: fromSession ? fromSession.creationTime : null,
+      endTime: fromSession ? fromSession.endTime : null,
       answered: state === SessionState.Established,
       paused: this.client.isCallHeld(sipSession),
       isCaller: 'incoming' in extra ? !extra.incoming : false,
@@ -1694,6 +1714,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       videoMuted: fromSession ? fromSession.isVideoMuted() : false,
       recording: fromSession ? fromSession.isRecording() : false,
       recordingPaused: false, // @TODO
+      sipSession,
       ...extra,
     });
 

package/src/domain/SFUNotAvailableError.js

@@ -0,0 +1,7 @@
+/* @flow */
+export default class SFUNotAvailableError extends Error {
+  constructor(message: string = 'Sorry your user is not configured to support video conference') {
+    super(message);
+    this.name = 'SFUNotAvailableError';
+  }
+}

package/src/index.js

@@ -9,6 +9,7 @@ import IssueReporter from './service/IssueReporter';
 // Domain
 import BadResponse from './domain/BadResponse';
 import ServerError from './domain/ServerError';
+import SFUNotAvailableError from './domain/SFUNotAvailableError';
 import Call from './domain/Call';
 import CallLog from './domain/CallLog';
 import Recording from './domain/Recording';
@@ -95,6 +96,7 @@ export default {
   WazoWebSocketClient: WebSocketClient,
   BadResponse,
   ServerError,
+  SFUNotAvailableError,
   Call,
   CallSession,
   CTIPhone,

package/src/lib/WazoSessionDescriptionHandler.js

@@ -62,11 +62,16 @@ class WazoSessionDescriptionHandler extends SessionDescriptionHandler {
 
   // $FlowFixMe
   setRemoteSessionDescription(sessionDescription: RTCSessionDescriptionInit): Promise<void> {
-    const result = super.setRemoteSessionDescription(sessionDescription);
+    try {
+      const result = super.setRemoteSessionDescription(sessionDescription);
 
-    this.eventEmitter.emit('setDescription', sessionDescription);
+      this.eventEmitter.emit('setDescription', sessionDescription);
 
-    return result;
+      return result;
+    } catch (error) {
+      this.logger.error('SessionDescriptionHandler.setRemoteSessionDescription error', error);
+      throw error;
+    }
   }
 
   // Overridden to avoid to fix ice-candidates missing in the SDP in react-native and chrome canary.

package/src/simple/Phone.js

@@ -14,6 +14,7 @@ import IssueReporter from '../service/IssueReporter';
 import Emitter from '../utils/Emitter';
 
 import Wazo from './index';
+import SFUNotAvailableError from '../domain/SFUNotAvailableError';
 
 const logger = IssueReporter.loggerFor('simple-phone');
 const sipLogger = IssueReporter.loggerFor('sip.js');
@@ -397,7 +398,7 @@ class Phone extends Emitter {
 
   checkSfu() {
     if (!this.hasSfu()) {
-      throw new Error('Sorry your user is not configured to support video conference');
+      throw new SFUNotAvailableError();
     }
   }
 

package/src/simple/index.js

@@ -15,6 +15,7 @@ import { CanceledCallError } from '../web-rtc-client';
 
 import BadResponse from '../domain/BadResponse';
 import ServerError from '../domain/ServerError';
+import SFUNotAvailableError from '../domain/SFUNotAvailableError';
 import Call from '../domain/Call';
 import CallLog from '../domain/CallLog';
 import Recording from '../domain/Recording';
@@ -101,6 +102,7 @@ const Wazo = {
   InvalidSubscription,
   InvalidAuthorization,
   CanceledCallError,
+  SFUNotAvailableError,
 };
 
 if (window) {

package/src/utils/api-requester.js

@@ -25,7 +25,7 @@ const logger = IssueReporter ? IssueReporter.loggerFor('api') : console;
 
 // Use a function here to be able to mock it in tests
 export const realFetch = () => {
-  if (typeof document !== 'undefined') {
+  if (typeof document !== 'undefined' && window.fetch) {
     // Browser
     return window.fetch;
   }

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;