@wazo/sdk 0.37.33 → 0.37.36

package/src/domain/Phone/WebRTCPhone.js

@@ -833,7 +833,11 @@ export default class WebRTCPhone extends Emitter implements Phone {
   accept(callSession: CallSession, cameraEnabled?: boolean): Promise<string | null> {
     logger.info('WebRTC phone - accept call', { id: callSession ? callSession.getId() : 'n/a', cameraEnabled });
     if (this.currentSipSession) {
-      this.holdSipSession(this.currentSipSession, this.currentCallSession, true);
+      this.holdSipSession(
+        this.currentSipSession,
+        this.currentCallSession,
+        true,
+      );
     }
 
     if (!callSession) {
@@ -841,6 +845,12 @@ export default class WebRTCPhone extends Emitter implements Phone {
       return Promise.resolve(null);
     }
 
+    const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
+    if (sipSessionId && !(sipSessionId in this.callSessions)) {
+      logger.warn('Call session already ended or not found, can\'t accept it.', { sipSessionId });
+      return Promise.resolve(null);
+    }
+
     if (callSession.getId() in this.acceptedSessions) {
       logger.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
       return Promise.resolve(callSession.sipCallId);
@@ -920,22 +930,22 @@ export default class WebRTCPhone extends Emitter implements Phone {
     }
   }
 
-  hold(callSession: CallSession, withEvent: boolean = true, isConference: boolean = false): ?Promise<any> {
+  hold(callSession: CallSession, withEvent: boolean = true): ?Promise<any> {
     logger.info('WebRTC hold', { id: callSession.getId() });
 
     const sipSession = this.findSipSession(callSession);
 
-    return sipSession ? this.holdSipSession(sipSession, callSession, withEvent, isConference) : null;
+    return sipSession ? this.holdSipSession(sipSession, callSession, withEvent) : null;
   }
 
   // @Deprecated
-  unhold(callSession: CallSession, withEvent: boolean = true, isConference: boolean = false): ?Promise<any> {
+  unhold(callSession: CallSession, withEvent: boolean = true): ?Promise<any> {
     console.warn('Please note that `phone.unhold()` is being deprecated; `phone.resume()` is the preferred method');
 
-    return this.resume(callSession, isConference, withEvent);
+    return this.resume(callSession, withEvent);
   }
 
-  resume(callSession?: CallSession, isConference: boolean = false, withEvent: boolean = true): Promise<any> {
+  resume(callSession?: CallSession, withEvent: boolean = true): Promise<any> {
     logger.info('WebRTC resume called', { id: callSession ? callSession.getId() : null });
 
     const sipSession = this.findSipSession(callSession);
@@ -948,10 +958,14 @@ export default class WebRTCPhone extends Emitter implements Phone {
     // Hold current session if different from the current one (we don't want 2 sessions active at the same time).
     if (this.currentSipSession && this.currentSipSession.id !== sipSession.id) {
       logger.info('WebRTC hold call after resume', { id: this.currentSipSession.id });
-      this.holdSipSession(this.currentSipSession, this.currentCallSession);
+      this.holdSipSession(
+        this.currentSipSession,
+        this.currentCallSession,
+        withEvent,
+      );
     }
 
-    const promise = this.unholdSipSession(sipSession, callSession, withEvent, isConference);
+    const promise = this.unholdSipSession(sipSession, callSession, withEvent);
     this.currentSipSession = sipSession;
     if (callSession) {
       this.currentCallSession = callSession;
@@ -964,7 +978,6 @@ export default class WebRTCPhone extends Emitter implements Phone {
     sipSession: Session,
     callSession: ?CallSession,
     withEvent: boolean = true,
-    isConference: boolean = false,
   ): Promise<any> {
     if (!sipSession) {
       return new Promise((resolve, reject) => reject(new Error('No session to hold')));
@@ -985,6 +998,8 @@ export default class WebRTCPhone extends Emitter implements Phone {
       await this.client.downgradeToAudio(sipSession);
     }
 
+    const isConference = !!callSession && callSession.isConference();
+
     const promise = this.client.hold(sipSession, isConference, hasVideo);
     if (withEvent) {
       this.eventEmitter.emit(ON_CALL_HELD, this._createCallSession(sipSession, callSession));
@@ -997,11 +1012,13 @@ export default class WebRTCPhone extends Emitter implements Phone {
     sipSession: Session,
     callSession: ?CallSession,
     withEvent: boolean = true,
-    isConference: boolean = false,
   ): Promise<any> {
     if (!sipSession) {
       return new Promise((resolve, reject) => reject(new Error('No session to unhold')));
     }
+
+    const isConference = !!callSession && callSession.isConference();
+
     const sessionId = this.getSipSessionId(sipSession);
     logger.info('WebRTC unhold sip session', { sessionId, isConference });
 
@@ -1143,7 +1160,11 @@ export default class WebRTCPhone extends Emitter implements Phone {
       await this.client.register();
     }
     if (this.currentSipSession) {
-      this.holdSipSession(this.currentSipSession, this.currentCallSession, true).catch(e => {
+      this.holdSipSession(
+        this.currentSipSession,
+        this.currentCallSession,
+        true,
+      ).catch(e => {
         logger.warn('Unable to hold current session when making another call', e);
       });
     }
@@ -1158,6 +1179,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       return Promise.resolve(null);
     }
     const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
+    callSession.setIsConference(conference);
     callSession.creationTime = new Date();
     this._updateCallSession(callSession);
 
@@ -1465,9 +1487,17 @@ export default class WebRTCPhone extends Emitter implements Phone {
     });
 
     this.client.on(this.client.ACCEPTED, (sipSession: Session) => {
-      logger.info('WebRTC call accepted', { sipId: sipSession.id });
+      const sessionId = this.getSipSessionId(sipSession);
+      const hasSession = sessionId in this.callSessions;
+
+      logger.info('WebRTC call accepted', { sipId: sessionId, hasSession });
 
-      this._onCallAccepted(sipSession, this.client.hasVideo(this.getSipSessionId(sipSession)));
+      if (!hasSession) {
+        logger.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', { sessionId });
+        return;
+      }
+
+      this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
 
       if (this.audioOutputDeviceId) {
         this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
@@ -1603,9 +1633,14 @@ export default class WebRTCPhone extends Emitter implements Phone {
     return this.client.getSipSessionId(sipSession);
   }
 
-  _updateCallSession(callSession: CallSession) {
+  getSipSessionIdFromCallSession(callSession: CallSession) {
     const sipSession = this.findSipSession(callSession);
-    const sipSessionId = sipSession ? this.getSipSessionId(sipSession) : null;
+
+    return sipSession ? this.getSipSessionId(sipSession) : null;
+  }
+
+  _updateCallSession(callSession: CallSession) {
+    const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
 
     if (sipSessionId) {
       this.callSessions[sipSessionId] = callSession;
@@ -1725,6 +1760,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
       recording: fromSession ? fromSession.isRecording() : false,
       recordingPaused: false, // @TODO
       sipSession,
+      conference: !!fromSession && fromSession.isConference(),
       ...extra,
     });
 

package/src/domain/Profile.js

@@ -57,6 +57,7 @@ type ProfileResponse = {
     },
   },
   mobile_phone_number: ?string,
+  ring_seconds?: number,
   subscription_type: ?number,
   services: {
     dnd: {
@@ -93,6 +94,7 @@ type ProfileArguments = {
   mobileNumber: string,
   forwards: Array<ForwardOption>,
   doNotDisturb?: boolean,
+  ringSeconds?: number,
   state?: string,
   status: string,
   subscriptionType: ?number,
@@ -124,6 +126,7 @@ export default class Profile {
   doNotDisturb: ?boolean;
   onlineCallRecordEnabled: ?boolean;
   state: ?string;
+  ringSeconds: ?number;
   voicemail: ?{ id: number, name: string };
   status: string;
   subscriptionType: ?number;
@@ -140,6 +143,7 @@ export default class Profile {
       lines: plain.lines.map(line => Line.parse(line)),
       username: plain.username,
       mobileNumber: plain.mobile_phone_number || '',
+      ringSeconds: plain.ring_seconds,
       forwards: [
         ForwardOption.parse(plain.forwards.unconditional, FORWARD_KEYS.UNCONDITIONAL),
         ForwardOption.parse(plain.forwards.noanswer, FORWARD_KEYS.NO_ANSWER),
@@ -176,6 +180,7 @@ export default class Profile {
     switchboards,
     agent,
     status,
+    ringSeconds,
     sipLines,
     callPickupTargetUsers,
     onlineCallRecordEnabled,
@@ -197,6 +202,7 @@ export default class Profile {
     this.status = status;
     this.callPickupTargetUsers = callPickupTargetUsers;
     this.onlineCallRecordEnabled = onlineCallRecordEnabled;
+    this.ringSeconds = ringSeconds;
 
     this.sipLines = sipLines || [];
   }

package/src/domain/__tests__/CallLog.test.js

@@ -15,8 +15,10 @@ describe('CallLog domain', () => {
           call_direction: 'outbound',
           source_extension: '8020',
           source_name: 'Jonathan Lessard',
+          source_user_uuid: '123',
           destination_extension: '4182250459',
           destination_name: null,
+          destination_user_uuid: '456',
           duration: 4,
           start: '2017-08-06T13:50:55.819057+00:00',
           end: '2017-08-08T13:51:23.822529+00:00',
@@ -47,10 +49,12 @@ describe('CallLog domain', () => {
         destination: {
           extension: '4182250459',
           name: '',
+          uuid: '456',
         },
         source: {
           extension: '8020',
           name: 'Jonathan Lessard',
+          uuid: '123',
         },
         id: 215,
         duration: 4000,

package/src/simple/Phone.js

@@ -204,16 +204,16 @@ class Phone extends Emitter {
     }
   }
 
-  hold(callSession: CallSession, isConference: boolean = false) {
-    return this.phone && this.phone.hold(callSession, true, isConference);
+  hold(callSession: CallSession) {
+    return this.phone && this.phone.hold(callSession, true);
   }
 
-  unhold(callSession: CallSession, isConference: boolean = false) {
-    return this.phone && this.phone.unhold(callSession, true, isConference);
+  unhold(callSession: CallSession) {
+    return this.phone && this.phone.unhold(callSession, true);
   }
 
-  resume(callSession: CallSession, isConference: boolean = false) {
-    return this.phone && this.phone.resume(callSession, isConference);
+  resume(callSession: CallSession) {
+    return this.phone && this.phone.resume(callSession);
   }
 
   reject(callSession: CallSession) {

package/src/web-rtc-client.js

@@ -75,7 +75,7 @@ export const events = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
 export const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
 export class CanceledCallError extends Error {}
 
-const MAX_REGISTER_TRIES = 10;
+const MAX_REGISTER_TRIES = 5;
 
 type MediaConfig = {
   audio: Object & boolean,