@wazo/sdk 0.37.40 → 0.37.41

This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
Files changed (31) hide show
  1. package/dist/wazo-sdk.js +1 -1
  2. package/esm/domain/Phone/WebRTCPhone.js +4 -4
  3. package/esm/index.js +4 -4
  4. package/esm/simple/Auth.js +4 -4
  5. package/esm/simple/Configuration.js +4 -4
  6. package/esm/simple/Phone.js +4 -4
  7. package/esm/simple/index.js +4 -4
  8. package/esm/simple/room/LocalParticipant.js +4 -4
  9. package/esm/simple/room/Participant.js +4 -4
  10. package/esm/simple/room/RemoteParticipant.js +4 -4
  11. package/esm/simple/room/Room.js +4 -4
  12. package/esm/simple/room/SipRoom.js +4 -4
  13. package/esm/simple/room/Stream.js +4 -4
  14. package/esm/simple/room/__tests__/RemoteParticipant.test.js +4 -4
  15. package/esm/simple/utils.js +4 -4
  16. package/lib/domain/Phone/WebRTCPhone.js +4 -4
  17. package/lib/index.js +4 -4
  18. package/lib/simple/Auth.js +4 -4
  19. package/lib/simple/Configuration.js +4 -4
  20. package/lib/simple/Phone.js +4 -4
  21. package/lib/simple/index.js +4 -4
  22. package/lib/simple/room/LocalParticipant.js +4 -4
  23. package/lib/simple/room/Participant.js +4 -4
  24. package/lib/simple/room/RemoteParticipant.js +4 -4
  25. package/lib/simple/room/Room.js +4 -4
  26. package/lib/simple/room/SipRoom.js +4 -4
  27. package/lib/simple/room/Stream.js +4 -4
  28. package/lib/simple/room/__tests__/RemoteParticipant.test.js +4 -4
  29. package/lib/simple/utils.js +4 -4
  30. package/package.json +1 -1
  31. package/src/domain/Phone/WebRTCPhone.js +4 -4
package/esm/domain/Phone/WebRTCPhone.js CHANGED
@@ -1438,7 +1438,7 @@ class WebRTCPhone extends Emitter {
1438
1438
  // When stopping screenshare and we had video before that, we have to re-upgrade
1439
1439
  // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
1440
1440
  // We have to downgrade to audio.
1441
- const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
1441
+ const screenshareStopped = this.currentScreenShare.hadVideo;
1442
1442
  const constraints = { audio: false, video: screenshareStopped };
1443
1443
 
1444
1444
  // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -1882,7 +1882,7 @@ class WebRTCPhone extends Emitter {
1882
1882
 
1883
1883
  logger.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
1884
1884
 
1885
- // Stop screenshre if needed
1885
+ // Stop screenshare if needed
1886
1886
  if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
1887
1887
  this.lastScreenShare = this.currentScreenShare;
1888
1888
  await this.stopScreenSharing(false, callSession);
@@ -2458,7 +2458,7 @@ class WebRTCPhone extends Emitter {
2458
2458
  });
2459
2459
 
2460
2460
  this.client.on(this.client.DISCONNECTED, () => {
2461
- logger.warn('WebRTC client disconnected');
2461
+ logger.info('WebRTC client disconnected');
2462
2462
 
2463
2463
  this.eventEmitter.emit(ON_UNREGISTERED);
2464
2464
 
@@ -2644,7 +2644,7 @@ class WebRTCPhone extends Emitter {
2644
2644
  endTime: fromSession ? fromSession.endTime : null,
2645
2645
  answered: state === SessionState.Established,
2646
2646
  paused: this.client.isCallHeld(sipSession),
2647
- isCaller: 'incoming' in extra ? !extra.incoming : false,
2647
+ isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
2648
2648
  cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
2649
2649
  number,
2650
2650
  ringing: false,
package/esm/index.js CHANGED
@@ -9835,7 +9835,7 @@ class WebRTCPhone extends Emitter {
9835
9835
  // When stopping screenshare and we had video before that, we have to re-upgrade
9836
9836
  // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
9837
9837
  // We have to downgrade to audio.
9838
- const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
9838
+ const screenshareStopped = this.currentScreenShare.hadVideo;
9839
9839
  const constraints = { audio: false, video: screenshareStopped };
9840
9840
 
9841
9841
  // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -10279,7 +10279,7 @@ class WebRTCPhone extends Emitter {
10279
10279
 
10280
10280
  logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
10281
10281
 
10282
- // Stop screenshre if needed
10282
+ // Stop screenshare if needed
10283
10283
  if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
10284
10284
  this.lastScreenShare = this.currentScreenShare;
10285
10285
  await this.stopScreenSharing(false, callSession);
@@ -10855,7 +10855,7 @@ class WebRTCPhone extends Emitter {
10855
10855
  });
10856
10856
 
10857
10857
  this.client.on(this.client.DISCONNECTED, () => {
10858
- logger$7.warn('WebRTC client disconnected');
10858
+ logger$7.info('WebRTC client disconnected');
10859
10859
 
10860
10860
  this.eventEmitter.emit(ON_UNREGISTERED);
10861
10861
 
@@ -11041,7 +11041,7 @@ class WebRTCPhone extends Emitter {
11041
11041
  endTime: fromSession ? fromSession.endTime : null,
11042
11042
  answered: state === SessionState.Established,
11043
11043
  paused: this.client.isCallHeld(sipSession),
11044
- isCaller: 'incoming' in extra ? !extra.incoming : false,
11044
+ isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
11045
11045
  cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
11046
11046
  number,
11047
11047
  ringing: false,
package/esm/simple/Auth.js CHANGED
@@ -9112,7 +9112,7 @@ class WebRTCPhone extends Emitter {
9112
9112
  // When stopping screenshare and we had video before that, we have to re-upgrade
9113
9113
  // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
9114
9114
  // We have to downgrade to audio.
9115
- const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
9115
+ const screenshareStopped = this.currentScreenShare.hadVideo;
9116
9116
  const constraints = { audio: false, video: screenshareStopped };
9117
9117
 
9118
9118
  // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -9556,7 +9556,7 @@ class WebRTCPhone extends Emitter {
9556
9556
 
9557
9557
  logger$8.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
9558
9558
 
9559
- // Stop screenshre if needed
9559
+ // Stop screenshare if needed
9560
9560
  if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
9561
9561
  this.lastScreenShare = this.currentScreenShare;
9562
9562
  await this.stopScreenSharing(false, callSession);
@@ -10132,7 +10132,7 @@ class WebRTCPhone extends Emitter {
10132
10132
  });
10133
10133
 
10134
10134
  this.client.on(this.client.DISCONNECTED, () => {
10135
- logger$8.warn('WebRTC client disconnected');
10135
+ logger$8.info('WebRTC client disconnected');
10136
10136
 
10137
10137
  this.eventEmitter.emit(ON_UNREGISTERED);
10138
10138
 
@@ -10318,7 +10318,7 @@ class WebRTCPhone extends Emitter {
10318
10318
  endTime: fromSession ? fromSession.endTime : null,
10319
10319
  answered: state === SessionState.Established,
10320
10320
  paused: this.client.isCallHeld(sipSession),
10321
- isCaller: 'incoming' in extra ? !extra.incoming : false,
10321
+ isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
10322
10322
  cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
10323
10323
  number,
10324
10324
  ringing: false,
package/esm/simple/Configuration.js CHANGED
@@ -9112,7 +9112,7 @@ class WebRTCPhone extends Emitter {
9112
9112
  // When stopping screenshare and we had video before that, we have to re-upgrade
9113
9113
  // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
9114
9114
  // We have to downgrade to audio.
9115
- const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
9115
+ const screenshareStopped = this.currentScreenShare.hadVideo;
9116
9116
  const constraints = { audio: false, video: screenshareStopped };
9117
9117
 
9118
9118
  // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -9556,7 +9556,7 @@ class WebRTCPhone extends Emitter {
9556
9556
 
9557
9557
  logger$8.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
9558
9558
 
9559
- // Stop screenshre if needed
9559
+ // Stop screenshare if needed
9560
9560
  if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
9561
9561
  this.lastScreenShare = this.currentScreenShare;
9562
9562
  await this.stopScreenSharing(false, callSession);
@@ -10132,7 +10132,7 @@ class WebRTCPhone extends Emitter {
10132
10132
  });
10133
10133
 
10134
10134
  this.client.on(this.client.DISCONNECTED, () => {
10135
- logger$8.warn('WebRTC client disconnected');
10135
+ logger$8.info('WebRTC client disconnected');
10136
10136
 
10137
10137
  this.eventEmitter.emit(ON_UNREGISTERED);
10138
10138
 
@@ -10318,7 +10318,7 @@ class WebRTCPhone extends Emitter {
10318
10318
  endTime: fromSession ? fromSession.endTime : null,
10319
10319
  answered: state === SessionState.Established,
10320
10320
  paused: this.client.isCallHeld(sipSession),
10321
- isCaller: 'incoming' in extra ? !extra.incoming : false,
10321
+ isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
10322
10322
  cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
10323
10323
  number,
10324
10324
  ringing: false,
package/esm/simple/Phone.js CHANGED
@@ -9112,7 +9112,7 @@ class WebRTCPhone extends Emitter {
9112
9112
  // When stopping screenshare and we had video before that, we have to re-upgrade
9113
9113
  // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
9114
9114
  // We have to downgrade to audio.
9115
- const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
9115
+ const screenshareStopped = this.currentScreenShare.hadVideo;
9116
9116
  const constraints = { audio: false, video: screenshareStopped };
9117
9117
 
9118
9118
  // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -9556,7 +9556,7 @@ class WebRTCPhone extends Emitter {
9556
9556
 
9557
9557
  logger$8.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
9558
9558
 
9559
- // Stop screenshre if needed
9559
+ // Stop screenshare if needed
9560
9560
  if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
9561
9561
  this.lastScreenShare = this.currentScreenShare;
9562
9562
  await this.stopScreenSharing(false, callSession);
@@ -10132,7 +10132,7 @@ class WebRTCPhone extends Emitter {
10132
10132
  });
10133
10133
 
10134
10134
  this.client.on(this.client.DISCONNECTED, () => {
10135
- logger$8.warn('WebRTC client disconnected');
10135
+ logger$8.info('WebRTC client disconnected');
10136
10136
 
10137
10137
  this.eventEmitter.emit(ON_UNREGISTERED);
10138
10138
 
@@ -10318,7 +10318,7 @@ class WebRTCPhone extends Emitter {
10318
10318
  endTime: fromSession ? fromSession.endTime : null,
10319
10319
  answered: state === SessionState.Established,
10320
10320
  paused: this.client.isCallHeld(sipSession),
10321
- isCaller: 'incoming' in extra ? !extra.incoming : false,
10321
+ isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
10322
10322
  cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
10323
10323
  number,
10324
10324
  ringing: false,
package/esm/simple/index.js CHANGED
@@ -9394,7 +9394,7 @@ class WebRTCPhone extends Emitter {
9394
9394
  // When stopping screenshare and we had video before that, we have to re-upgrade
9395
9395
  // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
9396
9396
  // We have to downgrade to audio.
9397
- const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
9397
+ const screenshareStopped = this.currentScreenShare.hadVideo;
9398
9398
  const constraints = { audio: false, video: screenshareStopped };
9399
9399
 
9400
9400
  // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -9838,7 +9838,7 @@ class WebRTCPhone extends Emitter {
9838
9838
 
9839
9839
  logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
9840
9840
 
9841
- // Stop screenshre if needed
9841
+ // Stop screenshare if needed
9842
9842
  if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
9843
9843
  this.lastScreenShare = this.currentScreenShare;
9844
9844
  await this.stopScreenSharing(false, callSession);
@@ -10414,7 +10414,7 @@ class WebRTCPhone extends Emitter {
10414
10414
  });
10415
10415
 
10416
10416
  this.client.on(this.client.DISCONNECTED, () => {
10417
- logger$7.warn('WebRTC client disconnected');
10417
+ logger$7.info('WebRTC client disconnected');
10418
10418
 
10419
10419
  this.eventEmitter.emit(ON_UNREGISTERED);
10420
10420
 
@@ -10600,7 +10600,7 @@ class WebRTCPhone extends Emitter {
10600
10600
  endTime: fromSession ? fromSession.endTime : null,
10601
10601
  answered: state === SessionState.Established,
10602
10602
  paused: this.client.isCallHeld(sipSession),
10603
- isCaller: 'incoming' in extra ? !extra.incoming : false,
10603
+ isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
10604
10604
  cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
10605
10605
  number,
10606
10606
  ringing: false,
package/esm/simple/room/LocalParticipant.js CHANGED
@@ -9394,7 +9394,7 @@ class WebRTCPhone extends Emitter {
9394
9394
  // When stopping screenshare and we had video before that, we have to re-upgrade
9395
9395
  // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
9396
9396
  // We have to downgrade to audio.
9397
- const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
9397
+ const screenshareStopped = this.currentScreenShare.hadVideo;
9398
9398
  const constraints = { audio: false, video: screenshareStopped };
9399
9399
 
9400
9400
  // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -9838,7 +9838,7 @@ class WebRTCPhone extends Emitter {
9838
9838
 
9839
9839
  logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
9840
9840
 
9841
- // Stop screenshre if needed
9841
+ // Stop screenshare if needed
9842
9842
  if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
9843
9843
  this.lastScreenShare = this.currentScreenShare;
9844
9844
  await this.stopScreenSharing(false, callSession);
@@ -10414,7 +10414,7 @@ class WebRTCPhone extends Emitter {
10414
10414
  });
10415
10415
 
10416
10416
  this.client.on(this.client.DISCONNECTED, () => {
10417
- logger$7.warn('WebRTC client disconnected');
10417
+ logger$7.info('WebRTC client disconnected');
10418
10418
 
10419
10419
  this.eventEmitter.emit(ON_UNREGISTERED);
10420
10420
 
@@ -10600,7 +10600,7 @@ class WebRTCPhone extends Emitter {
10600
10600
  endTime: fromSession ? fromSession.endTime : null,
10601
10601
  answered: state === SessionState.Established,
10602
10602
  paused: this.client.isCallHeld(sipSession),
10603
- isCaller: 'incoming' in extra ? !extra.incoming : false,
10603
+ isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
10604
10604
  cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
10605
10605
  number,
10606
10606
  ringing: false,
package/esm/simple/room/Participant.js CHANGED
@@ -9394,7 +9394,7 @@ class WebRTCPhone extends Emitter {
9394
9394
  // When stopping screenshare and we had video before that, we have to re-upgrade
9395
9395
  // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
9396
9396
  // We have to downgrade to audio.
9397
- const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
9397
+ const screenshareStopped = this.currentScreenShare.hadVideo;
9398
9398
  const constraints = { audio: false, video: screenshareStopped };
9399
9399
 
9400
9400
  // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -9838,7 +9838,7 @@ class WebRTCPhone extends Emitter {
9838
9838
 
9839
9839
  logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
9840
9840
 
9841
- // Stop screenshre if needed
9841
+ // Stop screenshare if needed
9842
9842
  if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
9843
9843
  this.lastScreenShare = this.currentScreenShare;
9844
9844
  await this.stopScreenSharing(false, callSession);
@@ -10414,7 +10414,7 @@ class WebRTCPhone extends Emitter {
10414
10414
  });
10415
10415
 
10416
10416
  this.client.on(this.client.DISCONNECTED, () => {
10417
- logger$7.warn('WebRTC client disconnected');
10417
+ logger$7.info('WebRTC client disconnected');
10418
10418
 
10419
10419
  this.eventEmitter.emit(ON_UNREGISTERED);
10420
10420
 
@@ -10600,7 +10600,7 @@ class WebRTCPhone extends Emitter {
10600
10600
  endTime: fromSession ? fromSession.endTime : null,
10601
10601
  answered: state === SessionState.Established,
10602
10602
  paused: this.client.isCallHeld(sipSession),
10603
- isCaller: 'incoming' in extra ? !extra.incoming : false,
10603
+ isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
10604
10604
  cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
10605
10605
  number,
10606
10606
  ringing: false,
package/esm/simple/room/RemoteParticipant.js CHANGED
@@ -9394,7 +9394,7 @@ class WebRTCPhone extends Emitter {
9394
9394
  // When stopping screenshare and we had video before that, we have to re-upgrade
9395
9395
  // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
9396
9396
  // We have to downgrade to audio.
9397
- const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
9397
+ const screenshareStopped = this.currentScreenShare.hadVideo;
9398
9398
  const constraints = { audio: false, video: screenshareStopped };
9399
9399
 
9400
9400
  // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -9838,7 +9838,7 @@ class WebRTCPhone extends Emitter {
9838
9838
 
9839
9839
  logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
9840
9840
 
9841
- // Stop screenshre if needed
9841
+ // Stop screenshare if needed
9842
9842
  if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
9843
9843
  this.lastScreenShare = this.currentScreenShare;
9844
9844
  await this.stopScreenSharing(false, callSession);
@@ -10414,7 +10414,7 @@ class WebRTCPhone extends Emitter {
10414
10414
  });
10415
10415
 
10416
10416
  this.client.on(this.client.DISCONNECTED, () => {
10417
- logger$7.warn('WebRTC client disconnected');
10417
+ logger$7.info('WebRTC client disconnected');
10418
10418
 
10419
10419
  this.eventEmitter.emit(ON_UNREGISTERED);
10420
10420
 
@@ -10600,7 +10600,7 @@ class WebRTCPhone extends Emitter {
10600
10600
  endTime: fromSession ? fromSession.endTime : null,
10601
10601
  answered: state === SessionState.Established,
10602
10602
  paused: this.client.isCallHeld(sipSession),
10603
- isCaller: 'incoming' in extra ? !extra.incoming : false,
10603
+ isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
10604
10604
  cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
10605
10605
  number,
10606
10606
  ringing: false,
package/esm/simple/room/Room.js CHANGED
@@ -9394,7 +9394,7 @@ class WebRTCPhone extends Emitter {
9394
9394
  // When stopping screenshare and we had video before that, we have to re-upgrade
9395
9395
  // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
9396
9396
  // We have to downgrade to audio.
9397
- const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
9397
+ const screenshareStopped = this.currentScreenShare.hadVideo;
9398
9398
  const constraints = { audio: false, video: screenshareStopped };
9399
9399
 
9400
9400
  // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -9838,7 +9838,7 @@ class WebRTCPhone extends Emitter {
9838
9838
 
9839
9839
  logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
9840
9840
 
9841
- // Stop screenshre if needed
9841
+ // Stop screenshare if needed
9842
9842
  if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
9843
9843
  this.lastScreenShare = this.currentScreenShare;
9844
9844
  await this.stopScreenSharing(false, callSession);
@@ -10414,7 +10414,7 @@ class WebRTCPhone extends Emitter {
10414
10414
  });
10415
10415
 
10416
10416
  this.client.on(this.client.DISCONNECTED, () => {
10417
- logger$7.warn('WebRTC client disconnected');
10417
+ logger$7.info('WebRTC client disconnected');
10418
10418
 
10419
10419
  this.eventEmitter.emit(ON_UNREGISTERED);
10420
10420
 
@@ -10600,7 +10600,7 @@ class WebRTCPhone extends Emitter {
10600
10600
  endTime: fromSession ? fromSession.endTime : null,
10601
10601
  answered: state === SessionState.Established,
10602
10602
  paused: this.client.isCallHeld(sipSession),
10603
- isCaller: 'incoming' in extra ? !extra.incoming : false,
10603
+ isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
10604
10604
  cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
10605
10605
  number,
10606
10606
  ringing: false,
package/esm/simple/room/SipRoom.js CHANGED
@@ -9394,7 +9394,7 @@ class WebRTCPhone extends Emitter {
9394
9394
  // When stopping screenshare and we had video before that, we have to re-upgrade
9395
9395
  // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
9396
9396
  // We have to downgrade to audio.
9397
- const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
9397
+ const screenshareStopped = this.currentScreenShare.hadVideo;
9398
9398
  const constraints = { audio: false, video: screenshareStopped };
9399
9399
 
9400
9400
  // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -9838,7 +9838,7 @@ class WebRTCPhone extends Emitter {
9838
9838
 
9839
9839
  logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
9840
9840
 
9841
- // Stop screenshre if needed
9841
+ // Stop screenshare if needed
9842
9842
  if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
9843
9843
  this.lastScreenShare = this.currentScreenShare;
9844
9844
  await this.stopScreenSharing(false, callSession);
@@ -10414,7 +10414,7 @@ class WebRTCPhone extends Emitter {
10414
10414
  });
10415
10415
 
10416
10416
  this.client.on(this.client.DISCONNECTED, () => {
10417
- logger$7.warn('WebRTC client disconnected');
10417
+ logger$7.info('WebRTC client disconnected');
10418
10418
 
10419
10419
  this.eventEmitter.emit(ON_UNREGISTERED);
10420
10420
 
@@ -10600,7 +10600,7 @@ class WebRTCPhone extends Emitter {
10600
10600
  endTime: fromSession ? fromSession.endTime : null,
10601
10601
  answered: state === SessionState.Established,
10602
10602
  paused: this.client.isCallHeld(sipSession),
10603
- isCaller: 'incoming' in extra ? !extra.incoming : false,
10603
+ isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
10604
10604
  cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
10605
10605
  number,
10606
10606
  ringing: false,
package/esm/simple/room/Stream.js CHANGED
@@ -9394,7 +9394,7 @@ class WebRTCPhone extends Emitter {
9394
9394
  // When stopping screenshare and we had video before that, we have to re-upgrade
9395
9395
  // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
9396
9396
  // We have to downgrade to audio.
9397
- const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
9397
+ const screenshareStopped = this.currentScreenShare.hadVideo;
9398
9398
  const constraints = { audio: false, video: screenshareStopped };
9399
9399
 
9400
9400
  // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -9838,7 +9838,7 @@ class WebRTCPhone extends Emitter {
9838
9838
 
9839
9839
  logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
9840
9840
 
9841
- // Stop screenshre if needed
9841
+ // Stop screenshare if needed
9842
9842
  if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
9843
9843
  this.lastScreenShare = this.currentScreenShare;
9844
9844
  await this.stopScreenSharing(false, callSession);
@@ -10414,7 +10414,7 @@ class WebRTCPhone extends Emitter {
10414
10414
  });
10415
10415
 
10416
10416
  this.client.on(this.client.DISCONNECTED, () => {
10417
- logger$7.warn('WebRTC client disconnected');
10417
+ logger$7.info('WebRTC client disconnected');
10418
10418
 
10419
10419
  this.eventEmitter.emit(ON_UNREGISTERED);
10420
10420
 
@@ -10600,7 +10600,7 @@ class WebRTCPhone extends Emitter {
10600
10600
  endTime: fromSession ? fromSession.endTime : null,
10601
10601
  answered: state === SessionState.Established,
10602
10602
  paused: this.client.isCallHeld(sipSession),
10603
- isCaller: 'incoming' in extra ? !extra.incoming : false,
10603
+ isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
10604
10604
  cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
10605
10605
  number,
10606
10606
  ringing: false,
package/esm/simple/room/__tests__/RemoteParticipant.test.js CHANGED
@@ -9394,7 +9394,7 @@ class WebRTCPhone extends Emitter {
9394
9394
  // When stopping screenshare and we had video before that, we have to re-upgrade
9395
9395
  // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
9396
9396
  // We have to downgrade to audio.
9397
- const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
9397
+ const screenshareStopped = this.currentScreenShare.hadVideo;
9398
9398
  const constraints = { audio: false, video: screenshareStopped };
9399
9399
 
9400
9400
  // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -9838,7 +9838,7 @@ class WebRTCPhone extends Emitter {
9838
9838
 
9839
9839
  logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
9840
9840
 
9841
- // Stop screenshre if needed
9841
+ // Stop screenshare if needed
9842
9842
  if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
9843
9843
  this.lastScreenShare = this.currentScreenShare;
9844
9844
  await this.stopScreenSharing(false, callSession);
@@ -10414,7 +10414,7 @@ class WebRTCPhone extends Emitter {
10414
10414
  });
10415
10415
 
10416
10416
  this.client.on(this.client.DISCONNECTED, () => {
10417
- logger$7.warn('WebRTC client disconnected');
10417
+ logger$7.info('WebRTC client disconnected');
10418
10418
 
10419
10419
  this.eventEmitter.emit(ON_UNREGISTERED);
10420
10420
 
@@ -10600,7 +10600,7 @@ class WebRTCPhone extends Emitter {
10600
10600
  endTime: fromSession ? fromSession.endTime : null,
10601
10601
  answered: state === SessionState.Established,
10602
10602
  paused: this.client.isCallHeld(sipSession),
10603
- isCaller: 'incoming' in extra ? !extra.incoming : false,
10603
+ isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
10604
10604
  cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
10605
10605
  number,
10606
10606
  ringing: false,
package/esm/simple/utils.js CHANGED
@@ -9394,7 +9394,7 @@ class WebRTCPhone extends Emitter {
9394
9394
  // When stopping screenshare and we had video before that, we have to re-upgrade
9395
9395
  // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
9396
9396
  // We have to downgrade to audio.
9397
- const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
9397
+ const screenshareStopped = this.currentScreenShare.hadVideo;
9398
9398
  const constraints = { audio: false, video: screenshareStopped };
9399
9399
 
9400
9400
  // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -9838,7 +9838,7 @@ class WebRTCPhone extends Emitter {
9838
9838
 
9839
9839
  logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
9840
9840
 
9841
- // Stop screenshre if needed
9841
+ // Stop screenshare if needed
9842
9842
  if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
9843
9843
  this.lastScreenShare = this.currentScreenShare;
9844
9844
  await this.stopScreenSharing(false, callSession);
@@ -10414,7 +10414,7 @@ class WebRTCPhone extends Emitter {
10414
10414
  });
10415
10415
 
10416
10416
  this.client.on(this.client.DISCONNECTED, () => {
10417
- logger$7.warn('WebRTC client disconnected');
10417
+ logger$7.info('WebRTC client disconnected');
10418
10418
 
10419
10419
  this.eventEmitter.emit(ON_UNREGISTERED);
10420
10420
 
@@ -10600,7 +10600,7 @@ class WebRTCPhone extends Emitter {
10600
10600
  endTime: fromSession ? fromSession.endTime : null,
10601
10601
  answered: state === SessionState.Established,
10602
10602
  paused: this.client.isCallHeld(sipSession),
10603
- isCaller: 'incoming' in extra ? !extra.incoming : false,
10603
+ isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
10604
10604
  cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
10605
10605
  number,
10606
10606
  ringing: false,
package/lib/domain/Phone/WebRTCPhone.js CHANGED
@@ -1447,7 +1447,7 @@ class WebRTCPhone extends Emitter {
1447
1447
  // When stopping screenshare and we had video before that, we have to re-upgrade
1448
1448
  // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
1449
1449
  // We have to downgrade to audio.
1450
- const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
1450
+ const screenshareStopped = this.currentScreenShare.hadVideo;
1451
1451
  const constraints = { audio: false, video: screenshareStopped };
1452
1452
 
1453
1453
  // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -1891,7 +1891,7 @@ class WebRTCPhone extends Emitter {
1891
1891
 
1892
1892
  logger.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
1893
1893
 
1894
- // Stop screenshre if needed
1894
+ // Stop screenshare if needed
1895
1895
  if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
1896
1896
  this.lastScreenShare = this.currentScreenShare;
1897
1897
  await this.stopScreenSharing(false, callSession);
@@ -2467,7 +2467,7 @@ class WebRTCPhone extends Emitter {
2467
2467
  });
2468
2468
 
2469
2469
  this.client.on(this.client.DISCONNECTED, () => {
2470
- logger.warn('WebRTC client disconnected');
2470
+ logger.info('WebRTC client disconnected');
2471
2471
 
2472
2472
  this.eventEmitter.emit(ON_UNREGISTERED);
2473
2473
 
@@ -2653,7 +2653,7 @@ class WebRTCPhone extends Emitter {
2653
2653
  endTime: fromSession ? fromSession.endTime : null,
2654
2654
  answered: state === sessionState.SessionState.Established,
2655
2655
  paused: this.client.isCallHeld(sipSession),
2656
- isCaller: 'incoming' in extra ? !extra.incoming : false,
2656
+ isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
2657
2657
  cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
2658
2658
  number,
2659
2659
  ringing: false,
package/lib/index.js CHANGED
@@ -9846,7 +9846,7 @@ class WebRTCPhone extends Emitter {
9846
9846
  // When stopping screenshare and we had video before that, we have to re-upgrade
9847
9847
  // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
9848
9848
  // We have to downgrade to audio.
9849
- const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
9849
+ const screenshareStopped = this.currentScreenShare.hadVideo;
9850
9850
  const constraints = { audio: false, video: screenshareStopped };
9851
9851
 
9852
9852
  // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -10290,7 +10290,7 @@ class WebRTCPhone extends Emitter {
10290
10290
 
10291
10291
  logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
10292
10292
 
10293
- // Stop screenshre if needed
10293
+ // Stop screenshare if needed
10294
10294
  if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
10295
10295
  this.lastScreenShare = this.currentScreenShare;
10296
10296
  await this.stopScreenSharing(false, callSession);
@@ -10866,7 +10866,7 @@ class WebRTCPhone extends Emitter {
10866
10866
  });
10867
10867
 
10868
10868
  this.client.on(this.client.DISCONNECTED, () => {
10869
- logger$7.warn('WebRTC client disconnected');
10869
+ logger$7.info('WebRTC client disconnected');
10870
10870
 
10871
10871
  this.eventEmitter.emit(ON_UNREGISTERED);
10872
10872
 
@@ -11052,7 +11052,7 @@ class WebRTCPhone extends Emitter {
11052
11052
  endTime: fromSession ? fromSession.endTime : null,
11053
11053
  answered: state === sessionState.SessionState.Established,
11054
11054
  paused: this.client.isCallHeld(sipSession),
11055
- isCaller: 'incoming' in extra ? !extra.incoming : false,
11055
+ isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
11056
11056
  cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
11057
11057
  number,
11058
11058
  ringing: false,
package/lib/simple/Auth.js CHANGED
@@ -9124,7 +9124,7 @@ class WebRTCPhone extends Emitter {
9124
9124
  // When stopping screenshare and we had video before that, we have to re-upgrade
9125
9125
  // When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
9126
9126
  // We have to downgrade to audio.
9127
- const screenshareStopped = this.currentScreenShare.sender && this.currentScreenShare.hadVideo;
9127
+ const screenshareStopped = this.currentScreenShare.hadVideo;
9128
9128
  const constraints = { audio: false, video: screenshareStopped };
9129
9129
 
9130
9130
  // We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
@@ -9568,7 +9568,7 @@ class WebRTCPhone extends Emitter {
9568
9568
 
9569
9569
  logger$8.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
9570
9570
 
9571
- // Stop screenshre if needed
9571
+ // Stop screenshare if needed
9572
9572
  if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
9573
9573
  this.lastScreenShare = this.currentScreenShare;
9574
9574
  await this.stopScreenSharing(false, callSession);
@@ -10144,7 +10144,7 @@ class WebRTCPhone extends Emitter {
10144
10144
  });
10145
10145
 
10146
10146
  this.client.on(this.client.DISCONNECTED, () => {
10147
- logger$8.warn('WebRTC client disconnected');
10147
+ logger$8.info('WebRTC client disconnected');
10148
10148
 
10149
10149
  this.eventEmitter.emit(ON_UNREGISTERED);
10150
10150
 
@@ -10330,7 +10330,7 @@ class WebRTCPhone extends Emitter {
10330
10330
  endTime: fromSession ? fromSession.endTime : null,
10331
10331
  answered: state === sessionState.SessionState.Established,
10332
10332
  paused: this.client.isCallHeld(sipSession),
10333
- isCaller: 'incoming' in extra ? !extra.incoming : false,
10333
+ isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
10334
10334
  cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
10335
10335
  number,
10336
10336
  ringing: false,