@wazo/sdk 0.38.5 → 0.38.6

package/lib/simple/room/SipRoom.js

@@ -3081,6 +3081,7 @@ const RECORD_STATE_ACTIVE = 'active';
                            
                  
                      
+                     
                    
                         
                    
@@ -3091,6 +3092,7 @@ const RECORD_STATE_ACTIVE = 'active';
 
                       
              
+                    
                     
                     
                      
@@ -3119,6 +3121,7 @@ class Call {
                           
                   
                     
+                   
                   
                  
                  
@@ -3143,6 +3146,7 @@ class Call {
       calleeNumber: plain.peer_caller_id_number,
       dialedExtension: plain.dialed_extension,
       isCaller: plain.is_caller,
+      isVideo: plain.is_video,
       muted: plain.muted,
       onHold: plain.on_hold,
       status: plain.status,
@@ -3166,6 +3170,7 @@ class Call {
     calleeNumber,
     dialedExtension,
     isCaller,
+    isVideo,
     lineId,
     muted,
     onHold,
@@ -3189,6 +3194,7 @@ class Call {
     this.startingTime = startingTime;
     this.talkingToIds = talkingToIds || [];
     this.recording = recording;
+    this.isVideo = !!isVideo;
 
     // Useful to compare instead of instanceof with minified code
     this.type = 'Call';
@@ -3662,7 +3668,7 @@ class CallSession {
       ringing: call.isRinging(),
       answered: call.isUp(),
       answeredBySystem: call.isUp() && call.talkingToIds.length === 0,
-      cameraEnabled: false,
+      cameraEnabled: call.isVideo,
       dialedExtension: call.dialedExtension,
       call,
       conference: false, // @FIXME?
@@ -7797,11 +7803,11 @@ class WebRTCClient extends Emitter {
     });
   }
 
-  changeAudioInputDevice(id        , session          ) {
+  changeAudioInputDevice(id        , session          , force          ) {
     const currentId = this.getAudioDeviceId();
     logger$8.info('setting audio input device', { id, currentId, session: !!session });
 
-    if (id === currentId) {
+    if (!force && id === currentId) {
       return null;
     }
 
@@ -9597,10 +9603,10 @@ class WebRTCPhone extends Emitter                  {
     this.audioRingVolume = volume;
   }
 
-  changeAudioInputDevice(id        ) {
+  changeAudioInputDevice(id        , force          ) {
     logger$7.info('WebRTC phone - changeAudio input device', { deviceId: id });
 
-    return this.client.changeAudioInputDevice(id, this.currentSipSession);
+    return this.client.changeAudioInputDevice(id, this.currentSipSession, force);
   }
 
   changeVideoInputDevice(id         ) {

package/lib/simple/room/Stream.js

@@ -3081,6 +3081,7 @@ const RECORD_STATE_ACTIVE = 'active';
                            
                  
                      
+                     
                    
                         
                    
@@ -3091,6 +3092,7 @@ const RECORD_STATE_ACTIVE = 'active';
 
                       
              
+                    
                     
                     
                      
@@ -3119,6 +3121,7 @@ class Call {
                           
                   
                     
+                   
                   
                  
                  
@@ -3143,6 +3146,7 @@ class Call {
       calleeNumber: plain.peer_caller_id_number,
       dialedExtension: plain.dialed_extension,
       isCaller: plain.is_caller,
+      isVideo: plain.is_video,
       muted: plain.muted,
       onHold: plain.on_hold,
       status: plain.status,
@@ -3166,6 +3170,7 @@ class Call {
     calleeNumber,
     dialedExtension,
     isCaller,
+    isVideo,
     lineId,
     muted,
     onHold,
@@ -3189,6 +3194,7 @@ class Call {
     this.startingTime = startingTime;
     this.talkingToIds = talkingToIds || [];
     this.recording = recording;
+    this.isVideo = !!isVideo;
 
     // Useful to compare instead of instanceof with minified code
     this.type = 'Call';
@@ -3662,7 +3668,7 @@ class CallSession {
       ringing: call.isRinging(),
       answered: call.isUp(),
       answeredBySystem: call.isUp() && call.talkingToIds.length === 0,
-      cameraEnabled: false,
+      cameraEnabled: call.isVideo,
       dialedExtension: call.dialedExtension,
       call,
       conference: false, // @FIXME?
@@ -7797,11 +7803,11 @@ class WebRTCClient extends Emitter {
     });
   }
 
-  changeAudioInputDevice(id        , session          ) {
+  changeAudioInputDevice(id        , session          , force          ) {
     const currentId = this.getAudioDeviceId();
     logger$8.info('setting audio input device', { id, currentId, session: !!session });
 
-    if (id === currentId) {
+    if (!force && id === currentId) {
       return null;
     }
 
@@ -9597,10 +9603,10 @@ class WebRTCPhone extends Emitter                  {
     this.audioRingVolume = volume;
   }
 
-  changeAudioInputDevice(id        ) {
+  changeAudioInputDevice(id        , force          ) {
     logger$7.info('WebRTC phone - changeAudio input device', { deviceId: id });
 
-    return this.client.changeAudioInputDevice(id, this.currentSipSession);
+    return this.client.changeAudioInputDevice(id, this.currentSipSession, force);
   }
 
   changeVideoInputDevice(id         ) {

package/lib/simple/room/__tests__/RemoteParticipant.test.js

@@ -3081,6 +3081,7 @@ const RECORD_STATE_ACTIVE = 'active';
                            
                  
                      
+                     
                    
                         
                    
@@ -3091,6 +3092,7 @@ const RECORD_STATE_ACTIVE = 'active';
 
                       
              
+                    
                     
                     
                      
@@ -3119,6 +3121,7 @@ class Call {
                           
                   
                     
+                   
                   
                  
                  
@@ -3143,6 +3146,7 @@ class Call {
       calleeNumber: plain.peer_caller_id_number,
       dialedExtension: plain.dialed_extension,
       isCaller: plain.is_caller,
+      isVideo: plain.is_video,
       muted: plain.muted,
       onHold: plain.on_hold,
       status: plain.status,
@@ -3166,6 +3170,7 @@ class Call {
     calleeNumber,
     dialedExtension,
     isCaller,
+    isVideo,
     lineId,
     muted,
     onHold,
@@ -3189,6 +3194,7 @@ class Call {
     this.startingTime = startingTime;
     this.talkingToIds = talkingToIds || [];
     this.recording = recording;
+    this.isVideo = !!isVideo;
 
     // Useful to compare instead of instanceof with minified code
     this.type = 'Call';
@@ -3662,7 +3668,7 @@ class CallSession {
       ringing: call.isRinging(),
       answered: call.isUp(),
       answeredBySystem: call.isUp() && call.talkingToIds.length === 0,
-      cameraEnabled: false,
+      cameraEnabled: call.isVideo,
       dialedExtension: call.dialedExtension,
       call,
       conference: false, // @FIXME?
@@ -7797,11 +7803,11 @@ class WebRTCClient extends Emitter {
     });
   }
 
-  changeAudioInputDevice(id        , session          ) {
+  changeAudioInputDevice(id        , session          , force          ) {
     const currentId = this.getAudioDeviceId();
     logger$8.info('setting audio input device', { id, currentId, session: !!session });
 
-    if (id === currentId) {
+    if (!force && id === currentId) {
       return null;
     }
 
@@ -9597,10 +9603,10 @@ class WebRTCPhone extends Emitter                  {
     this.audioRingVolume = volume;
   }
 
-  changeAudioInputDevice(id        ) {
+  changeAudioInputDevice(id        , force          ) {
     logger$7.info('WebRTC phone - changeAudio input device', { deviceId: id });
 
-    return this.client.changeAudioInputDevice(id, this.currentSipSession);
+    return this.client.changeAudioInputDevice(id, this.currentSipSession, force);
   }
 
   changeVideoInputDevice(id         ) {

package/lib/simple/utils.js

@@ -3083,6 +3083,7 @@ const RECORD_STATE_ACTIVE = 'active';
                            
                  
                      
+                     
                    
                         
                    
@@ -3093,6 +3094,7 @@ const RECORD_STATE_ACTIVE = 'active';
 
                       
              
+                    
                     
                     
                      
@@ -3121,6 +3123,7 @@ class Call {
                           
                   
                     
+                   
                   
                  
                  
@@ -3145,6 +3148,7 @@ class Call {
       calleeNumber: plain.peer_caller_id_number,
       dialedExtension: plain.dialed_extension,
       isCaller: plain.is_caller,
+      isVideo: plain.is_video,
       muted: plain.muted,
       onHold: plain.on_hold,
       status: plain.status,
@@ -3168,6 +3172,7 @@ class Call {
     calleeNumber,
     dialedExtension,
     isCaller,
+    isVideo,
     lineId,
     muted,
     onHold,
@@ -3191,6 +3196,7 @@ class Call {
     this.startingTime = startingTime;
     this.talkingToIds = talkingToIds || [];
     this.recording = recording;
+    this.isVideo = !!isVideo;
 
     // Useful to compare instead of instanceof with minified code
     this.type = 'Call';
@@ -3664,7 +3670,7 @@ class CallSession {
       ringing: call.isRinging(),
       answered: call.isUp(),
       answeredBySystem: call.isUp() && call.talkingToIds.length === 0,
-      cameraEnabled: false,
+      cameraEnabled: call.isVideo,
       dialedExtension: call.dialedExtension,
       call,
       conference: false, // @FIXME?
@@ -7799,11 +7805,11 @@ class WebRTCClient extends Emitter {
     });
   }
 
-  changeAudioInputDevice(id        , session          ) {
+  changeAudioInputDevice(id        , session          , force          ) {
     const currentId = this.getAudioDeviceId();
     logger$8.info('setting audio input device', { id, currentId, session: !!session });
 
-    if (id === currentId) {
+    if (!force && id === currentId) {
       return null;
     }
 
@@ -9599,10 +9605,10 @@ class WebRTCPhone extends Emitter                  {
     this.audioRingVolume = volume;
   }
 
-  changeAudioInputDevice(id        ) {
+  changeAudioInputDevice(id        , force          ) {
     logger$7.info('WebRTC phone - changeAudio input device', { deviceId: id });
 
-    return this.client.changeAudioInputDevice(id, this.currentSipSession);
+    return this.client.changeAudioInputDevice(id, this.currentSipSession, force);
   }
 
   changeVideoInputDevice(id         ) {

package/lib/web-rtc-client.js

@@ -3618,6 +3618,7 @@ const RECORD_STATE_ACTIVE = 'active';
                            
                  
                      
+                     
                    
                         
                    
@@ -3628,6 +3629,7 @@ const RECORD_STATE_ACTIVE = 'active';
 
                       
              
+                    
                     
                     
                      
@@ -3656,6 +3658,7 @@ class Call {
                           
                   
                     
+                   
                   
                  
                  
@@ -3680,6 +3683,7 @@ class Call {
       calleeNumber: plain.peer_caller_id_number,
       dialedExtension: plain.dialed_extension,
       isCaller: plain.is_caller,
+      isVideo: plain.is_video,
       muted: plain.muted,
       onHold: plain.on_hold,
       status: plain.status,
@@ -3703,6 +3707,7 @@ class Call {
     calleeNumber,
     dialedExtension,
     isCaller,
+    isVideo,
     lineId,
     muted,
     onHold,
@@ -3726,6 +3731,7 @@ class Call {
     this.startingTime = startingTime;
     this.talkingToIds = talkingToIds || [];
     this.recording = recording;
+    this.isVideo = !!isVideo;
 
     // Useful to compare instead of instanceof with minified code
     this.type = 'Call';
@@ -6428,11 +6434,11 @@ class WebRTCClient extends Emitter {
     });
   }
 
-  changeAudioInputDevice(id        , session          ) {
+  changeAudioInputDevice(id        , session          , force          ) {
     const currentId = this.getAudioDeviceId();
     logger.info('setting audio input device', { id, currentId, session: !!session });
 
-    if (id === currentId) {
+    if (!force && id === currentId) {
       return null;
     }
 

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@wazo/sdk",
-  "version": "0.38.5",
+  "version": "0.38.6",
   "description": "Wazo's JavaScript Software Development Kit.",
   "main": "dist/wazo-sdk.js",
   "author": "Wazo (http://wazo.io)",

package/src/__tests__/web-rtc-client.test.js

@@ -43,3 +43,53 @@ describe('WebRTC client', () => {
     expect(client.isAudioMuted(oldKindUnmuted)).toBeFalsy();
   });
 });
+
+describe('changeAudioInputDevice', () => {
+  const defaultId = 'default';
+  const deviceId = 'device1';
+  const constraints = { audio: { deviceId: { exact: defaultId } }, video: null };
+
+  const session = {
+    sessionDescriptionHandler: {
+      peerConnection: {
+        getSenders: () => [{ track: { kind: 'audio', enabled: true } }],
+      },
+    },
+  };
+
+  const stream = {};
+  const getAudioTracksMock = jest.fn(() => []);
+  Object.defineProperty(stream, 'getAudioTracks', {
+    value: getAudioTracksMock,
+  });
+
+  const getUserMediaMock = jest.fn(async () => {
+    return new Promise(resolve => {
+      resolve(stream);
+    });
+  });
+
+  Object.defineProperty(global.navigator, 'mediaDevices', {
+    value: {
+      getUserMedia: getUserMediaMock,
+    },
+  });
+
+  it('should change the audio input track if the provided id is different', async () => {
+    client.setMediaConstraints(constraints);
+    expect(client.getAudioDeviceId()).toBe(defaultId);
+    expect(client.changeAudioInputDevice(deviceId, session)).toBeTruthy();
+  });
+
+  it('should NOT change the audio input track if the provided id is the same', async () => {
+    client.setMediaConstraints(constraints);
+    expect(client.getAudioDeviceId()).toBe(defaultId);
+    expect(client.changeAudioInputDevice(defaultId, session)).toBeFalsy();
+  });
+
+  it('should change the audio input track if the provided id is the same and force param is TRUE', async () => {
+    client.setMediaConstraints(constraints);
+    expect(client.getAudioDeviceId()).toBe(defaultId);
+    expect(client.changeAudioInputDevice(defaultId, session, true)).toBeTruthy();
+  });
+});

package/src/domain/Call.js

@@ -15,6 +15,7 @@ type CallResponse = {
   dialed_extension: string,
   status: string,
   is_caller: boolean,
+  is_video?: boolean,
   line_id: ?number,
   creation_time: string,
   on_hold: boolean,
@@ -27,6 +28,7 @@ type CallArguments = {
   id: string,
   sipCallId: string,
   isCaller: boolean,
+  isVideo?: boolean,
   callerName: string,
   callerNumber: string,
   calleeName: string,
@@ -53,6 +55,7 @@ export default class Call {
   dialedExtension: string;
   lineId: ?number;
   isCaller: boolean;
+  isVideo: boolean;
   onHold: boolean;
   muted: boolean;
   status: string;
@@ -77,6 +80,7 @@ export default class Call {
       calleeNumber: plain.peer_caller_id_number,
       dialedExtension: plain.dialed_extension,
       isCaller: plain.is_caller,
+      isVideo: plain.is_video,
       muted: plain.muted,
       onHold: plain.on_hold,
       status: plain.status,
@@ -100,6 +104,7 @@ export default class Call {
     calleeNumber,
     dialedExtension,
     isCaller,
+    isVideo,
     lineId,
     muted,
     onHold,
@@ -123,6 +128,7 @@ export default class Call {
     this.startingTime = startingTime;
     this.talkingToIds = talkingToIds || [];
     this.recording = recording;
+    this.isVideo = !!isVideo;
 
     // Useful to compare instead of instanceof with minified code
     this.type = 'Call';

package/src/domain/CallSession.js

@@ -390,7 +390,7 @@ export default class CallSession {
       ringing: call.isRinging(),
       answered: call.isUp(),
       answeredBySystem: call.isUp() && call.talkingToIds.length === 0,
-      cameraEnabled: false,
+      cameraEnabled: call.isVideo,
       dialedExtension: call.dialedExtension,
       call,
       conference: false, // @FIXME?

package/src/domain/Phone/WebRTCPhone.js

@@ -668,10 +668,10 @@ export default class WebRTCPhone extends Emitter implements Phone {
     this.audioRingVolume = volume;
   }
 
-  changeAudioInputDevice(id: string) {
+  changeAudioInputDevice(id: string, force: ?boolean) {
     logger.info('WebRTC phone - changeAudio input device', { deviceId: id });
 
-    return this.client.changeAudioInputDevice(id, this.currentSipSession);
+    return this.client.changeAudioInputDevice(id, this.currentSipSession, force);
   }
 
   changeVideoInputDevice(id: ?string) {

package/src/web-rtc-client.js

@@ -1070,11 +1070,11 @@ export default class WebRTCClient extends Emitter {
     });
   }
 
-  changeAudioInputDevice(id: string, session: ?Inviter) {
+  changeAudioInputDevice(id: string, session: ?Inviter, force: ?boolean) {
     const currentId = this.getAudioDeviceId();
     logger.info('setting audio input device', { id, currentId, session: !!session });
 
-    if (id === currentId) {
+    if (!force && id === currentId) {
       return null;
     }