@wazo/sdk 0.38.10 → 0.38.11

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Files changed (46) hide show
  1. package/dist/wazo-sdk.js +2 -2
  2. package/esm/__tests__/web-rtc-client.test.js +10 -6
  3. package/esm/checker/Checker.js +10 -6
  4. package/esm/checker/checks/index.js +10 -6
  5. package/esm/checker/checks/webrtc-transport.js +10 -6
  6. package/esm/checker/checks/webrtc.js +10 -6
  7. package/esm/domain/Phone/WebRTCPhone.js +2 -2
  8. package/esm/index.js +12 -8
  9. package/esm/simple/Auth.js +12 -8
  10. package/esm/simple/Configuration.js +12 -8
  11. package/esm/simple/Phone.js +12 -8
  12. package/esm/simple/index.js +12 -8
  13. package/esm/simple/room/LocalParticipant.js +12 -8
  14. package/esm/simple/room/Participant.js +12 -8
  15. package/esm/simple/room/RemoteParticipant.js +12 -8
  16. package/esm/simple/room/Room.js +12 -8
  17. package/esm/simple/room/SipRoom.js +12 -8
  18. package/esm/simple/room/Stream.js +12 -8
  19. package/esm/simple/room/__tests__/Participant.test.js +12 -8
  20. package/esm/simple/room/__tests__/RemoteParticipant.test.js +12 -8
  21. package/esm/simple/utils.js +12 -8
  22. package/esm/web-rtc-client.js +10 -6
  23. package/lib/__tests__/web-rtc-client.test.js +10 -6
  24. package/lib/checker/Checker.js +10 -6
  25. package/lib/checker/checks/index.js +10 -6
  26. package/lib/checker/checks/webrtc-transport.js +10 -6
  27. package/lib/checker/checks/webrtc.js +10 -6
  28. package/lib/domain/Phone/WebRTCPhone.js +2 -2
  29. package/lib/index.js +12 -8
  30. package/lib/simple/Auth.js +12 -8
  31. package/lib/simple/Configuration.js +12 -8
  32. package/lib/simple/Phone.js +12 -8
  33. package/lib/simple/index.js +12 -8
  34. package/lib/simple/room/LocalParticipant.js +12 -8
  35. package/lib/simple/room/Participant.js +12 -8
  36. package/lib/simple/room/RemoteParticipant.js +12 -8
  37. package/lib/simple/room/Room.js +12 -8
  38. package/lib/simple/room/SipRoom.js +12 -8
  39. package/lib/simple/room/Stream.js +12 -8
  40. package/lib/simple/room/__tests__/Participant.test.js +12 -8
  41. package/lib/simple/room/__tests__/RemoteParticipant.test.js +12 -8
  42. package/lib/simple/utils.js +12 -8
  43. package/lib/web-rtc-client.js +10 -6
  44. package/package.json +1 -1
  45. package/src/domain/Phone/WebRTCPhone.js +2 -2
  46. package/src/web-rtc-client.js +10 -6
package/lib/simple/room/Stream.js CHANGED
@@ -7832,9 +7832,11 @@ class WebRTCClient extends Emitter {
7832
7832
  const videoTracks = localStream.getVideoTracks();
7833
7833
 
7834
7834
  // Remove video senders
7835
- pc.getSenders().filter(sender => sender.track && sender.track.kind === 'video').forEach(videoSender => {
7836
- videoSender.replaceTrack(null);
7837
- });
7835
+ if (pc.getSenders) {
7836
+ pc.getSenders().filter(sender => sender.track && sender.track.kind === 'video').forEach(videoSender => {
7837
+ videoSender.replaceTrack(null);
7838
+ });
7839
+ }
7838
7840
 
7839
7841
  videoTracks.forEach(videoTrack => {
7840
7842
  videoTrack.enabled = false;
@@ -8067,7 +8069,8 @@ class WebRTCClient extends Emitter {
8067
8069
  const constraints = { audio: { deviceId: { exact: deviceId } } };
8068
8070
  const stream = await navigator.mediaDevices.getUserMedia(constraints);
8069
8071
  const audioTrack = stream.getAudioTracks()[0];
8070
- const sender = pc && pc.getSenders().find(s => audioTrack && s && s.track && s.track.kind === audioTrack.kind);
8072
+ const sender = pc && pc.getSenders
8073
+ && pc.getSenders().find(s => audioTrack && s && s.track && s.track.kind === audioTrack.kind);
8071
8074
 
8072
8075
  if (sender) {
8073
8076
  if (sender.track) {
@@ -8130,10 +8133,11 @@ class WebRTCClient extends Emitter {
8130
8133
  // $FlowFixMe
8131
8134
  return navigator.mediaDevices.getUserMedia(constraints).then(async stream => {
8132
8135
  const videoTrack = stream.getVideoTracks()[0];
8133
- let sender = pc && pc.getSenders().find(s => videoTrack && s && s.track && s.track.kind === videoTrack.kind);
8136
+ let sender = pc && pc.getSenders
8137
+ && pc.getSenders().find(s => videoTrack && s && s.track && s.track.kind === videoTrack.kind);
8134
8138
  let wasTrackEnabled = false;
8135
8139
  if (!sender) {
8136
- sender = pc && pc.getSenders().find(s => !s.track);
8140
+ sender = pc && pc.getSenders && pc.getSenders().find(s => !s.track);
8137
8141
  }
8138
8142
 
8139
8143
  if (sender) {
@@ -9667,7 +9671,7 @@ class WebRTCPhone extends Emitter {
9667
9671
  const screenTrack = screenShareStream.getVideoTracks()[0];
9668
9672
  const sipSession = this.currentSipSession;
9669
9673
  const pc = this.client.getPeerConnection(this.getSipSessionId(sipSession));
9670
- const sender = pc && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
9674
+ const sender = pc && pc.getSenders && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
9671
9675
  const localStream = this.client.getLocalStream(this.getSipSessionId(sipSession));
9672
9676
  const videoTrack = localStream ? localStream.getTracks().find(track => track.kind === 'video') : null;
9673
9677
  const hadVideo = !!videoTrack;
@@ -9749,7 +9753,7 @@ class WebRTCPhone extends Emitter {
9749
9753
  const screenTrack = screenStream.getVideoTracks()[0];
9750
9754
  const sipSessionId = this.getSipSessionId(sipSession);
9751
9755
  const pc = this.client.getPeerConnection(sipSessionId);
9752
- const sender = pc && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
9756
+ const sender = pc && pc.getSenders && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
9753
9757
 
9754
9758
  logger$7.info('WebRTC phone - on screensharing', {
9755
9759
  hadVideo,
package/lib/simple/room/__tests__/Participant.test.js CHANGED
@@ -7832,9 +7832,11 @@ class WebRTCClient extends Emitter {
7832
7832
  const videoTracks = localStream.getVideoTracks();
7833
7833
 
7834
7834
  // Remove video senders
7835
- pc.getSenders().filter(sender => sender.track && sender.track.kind === 'video').forEach(videoSender => {
7836
- videoSender.replaceTrack(null);
7837
- });
7835
+ if (pc.getSenders) {
7836
+ pc.getSenders().filter(sender => sender.track && sender.track.kind === 'video').forEach(videoSender => {
7837
+ videoSender.replaceTrack(null);
7838
+ });
7839
+ }
7838
7840
 
7839
7841
  videoTracks.forEach(videoTrack => {
7840
7842
  videoTrack.enabled = false;
@@ -8067,7 +8069,8 @@ class WebRTCClient extends Emitter {
8067
8069
  const constraints = { audio: { deviceId: { exact: deviceId } } };
8068
8070
  const stream = await navigator.mediaDevices.getUserMedia(constraints);
8069
8071
  const audioTrack = stream.getAudioTracks()[0];
8070
- const sender = pc && pc.getSenders().find(s => audioTrack && s && s.track && s.track.kind === audioTrack.kind);
8072
+ const sender = pc && pc.getSenders
8073
+ && pc.getSenders().find(s => audioTrack && s && s.track && s.track.kind === audioTrack.kind);
8071
8074
 
8072
8075
  if (sender) {
8073
8076
  if (sender.track) {
@@ -8130,10 +8133,11 @@ class WebRTCClient extends Emitter {
8130
8133
  // $FlowFixMe
8131
8134
  return navigator.mediaDevices.getUserMedia(constraints).then(async stream => {
8132
8135
  const videoTrack = stream.getVideoTracks()[0];
8133
- let sender = pc && pc.getSenders().find(s => videoTrack && s && s.track && s.track.kind === videoTrack.kind);
8136
+ let sender = pc && pc.getSenders
8137
+ && pc.getSenders().find(s => videoTrack && s && s.track && s.track.kind === videoTrack.kind);
8134
8138
  let wasTrackEnabled = false;
8135
8139
  if (!sender) {
8136
- sender = pc && pc.getSenders().find(s => !s.track);
8140
+ sender = pc && pc.getSenders && pc.getSenders().find(s => !s.track);
8137
8141
  }
8138
8142
 
8139
8143
  if (sender) {
@@ -9667,7 +9671,7 @@ class WebRTCPhone extends Emitter {
9667
9671
  const screenTrack = screenShareStream.getVideoTracks()[0];
9668
9672
  const sipSession = this.currentSipSession;
9669
9673
  const pc = this.client.getPeerConnection(this.getSipSessionId(sipSession));
9670
- const sender = pc && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
9674
+ const sender = pc && pc.getSenders && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
9671
9675
  const localStream = this.client.getLocalStream(this.getSipSessionId(sipSession));
9672
9676
  const videoTrack = localStream ? localStream.getTracks().find(track => track.kind === 'video') : null;
9673
9677
  const hadVideo = !!videoTrack;
@@ -9749,7 +9753,7 @@ class WebRTCPhone extends Emitter {
9749
9753
  const screenTrack = screenStream.getVideoTracks()[0];
9750
9754
  const sipSessionId = this.getSipSessionId(sipSession);
9751
9755
  const pc = this.client.getPeerConnection(sipSessionId);
9752
- const sender = pc && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
9756
+ const sender = pc && pc.getSenders && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
9753
9757
 
9754
9758
  logger$7.info('WebRTC phone - on screensharing', {
9755
9759
  hadVideo,
package/lib/simple/room/__tests__/RemoteParticipant.test.js CHANGED
@@ -7832,9 +7832,11 @@ class WebRTCClient extends Emitter {
7832
7832
  const videoTracks = localStream.getVideoTracks();
7833
7833
 
7834
7834
  // Remove video senders
7835
- pc.getSenders().filter(sender => sender.track && sender.track.kind === 'video').forEach(videoSender => {
7836
- videoSender.replaceTrack(null);
7837
- });
7835
+ if (pc.getSenders) {
7836
+ pc.getSenders().filter(sender => sender.track && sender.track.kind === 'video').forEach(videoSender => {
7837
+ videoSender.replaceTrack(null);
7838
+ });
7839
+ }
7838
7840
 
7839
7841
  videoTracks.forEach(videoTrack => {
7840
7842
  videoTrack.enabled = false;
@@ -8067,7 +8069,8 @@ class WebRTCClient extends Emitter {
8067
8069
  const constraints = { audio: { deviceId: { exact: deviceId } } };
8068
8070
  const stream = await navigator.mediaDevices.getUserMedia(constraints);
8069
8071
  const audioTrack = stream.getAudioTracks()[0];
8070
- const sender = pc && pc.getSenders().find(s => audioTrack && s && s.track && s.track.kind === audioTrack.kind);
8072
+ const sender = pc && pc.getSenders
8073
+ && pc.getSenders().find(s => audioTrack && s && s.track && s.track.kind === audioTrack.kind);
8071
8074
 
8072
8075
  if (sender) {
8073
8076
  if (sender.track) {
@@ -8130,10 +8133,11 @@ class WebRTCClient extends Emitter {
8130
8133
  // $FlowFixMe
8131
8134
  return navigator.mediaDevices.getUserMedia(constraints).then(async stream => {
8132
8135
  const videoTrack = stream.getVideoTracks()[0];
8133
- let sender = pc && pc.getSenders().find(s => videoTrack && s && s.track && s.track.kind === videoTrack.kind);
8136
+ let sender = pc && pc.getSenders
8137
+ && pc.getSenders().find(s => videoTrack && s && s.track && s.track.kind === videoTrack.kind);
8134
8138
  let wasTrackEnabled = false;
8135
8139
  if (!sender) {
8136
- sender = pc && pc.getSenders().find(s => !s.track);
8140
+ sender = pc && pc.getSenders && pc.getSenders().find(s => !s.track);
8137
8141
  }
8138
8142
 
8139
8143
  if (sender) {
@@ -9667,7 +9671,7 @@ class WebRTCPhone extends Emitter {
9667
9671
  const screenTrack = screenShareStream.getVideoTracks()[0];
9668
9672
  const sipSession = this.currentSipSession;
9669
9673
  const pc = this.client.getPeerConnection(this.getSipSessionId(sipSession));
9670
- const sender = pc && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
9674
+ const sender = pc && pc.getSenders && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
9671
9675
  const localStream = this.client.getLocalStream(this.getSipSessionId(sipSession));
9672
9676
  const videoTrack = localStream ? localStream.getTracks().find(track => track.kind === 'video') : null;
9673
9677
  const hadVideo = !!videoTrack;
@@ -9749,7 +9753,7 @@ class WebRTCPhone extends Emitter {
9749
9753
  const screenTrack = screenStream.getVideoTracks()[0];
9750
9754
  const sipSessionId = this.getSipSessionId(sipSession);
9751
9755
  const pc = this.client.getPeerConnection(sipSessionId);
9752
- const sender = pc && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
9756
+ const sender = pc && pc.getSenders && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
9753
9757
 
9754
9758
  logger$7.info('WebRTC phone - on screensharing', {
9755
9759
  hadVideo,
package/lib/simple/utils.js CHANGED
@@ -7834,9 +7834,11 @@ class WebRTCClient extends Emitter {
7834
7834
  const videoTracks = localStream.getVideoTracks();
7835
7835
 
7836
7836
  // Remove video senders
7837
- pc.getSenders().filter(sender => sender.track && sender.track.kind === 'video').forEach(videoSender => {
7838
- videoSender.replaceTrack(null);
7839
- });
7837
+ if (pc.getSenders) {
7838
+ pc.getSenders().filter(sender => sender.track && sender.track.kind === 'video').forEach(videoSender => {
7839
+ videoSender.replaceTrack(null);
7840
+ });
7841
+ }
7840
7842
 
7841
7843
  videoTracks.forEach(videoTrack => {
7842
7844
  videoTrack.enabled = false;
@@ -8069,7 +8071,8 @@ class WebRTCClient extends Emitter {
8069
8071
  const constraints = { audio: { deviceId: { exact: deviceId } } };
8070
8072
  const stream = await navigator.mediaDevices.getUserMedia(constraints);
8071
8073
  const audioTrack = stream.getAudioTracks()[0];
8072
- const sender = pc && pc.getSenders().find(s => audioTrack && s && s.track && s.track.kind === audioTrack.kind);
8074
+ const sender = pc && pc.getSenders
8075
+ && pc.getSenders().find(s => audioTrack && s && s.track && s.track.kind === audioTrack.kind);
8073
8076
 
8074
8077
  if (sender) {
8075
8078
  if (sender.track) {
@@ -8132,10 +8135,11 @@ class WebRTCClient extends Emitter {
8132
8135
  // $FlowFixMe
8133
8136
  return navigator.mediaDevices.getUserMedia(constraints).then(async stream => {
8134
8137
  const videoTrack = stream.getVideoTracks()[0];
8135
- let sender = pc && pc.getSenders().find(s => videoTrack && s && s.track && s.track.kind === videoTrack.kind);
8138
+ let sender = pc && pc.getSenders
8139
+ && pc.getSenders().find(s => videoTrack && s && s.track && s.track.kind === videoTrack.kind);
8136
8140
  let wasTrackEnabled = false;
8137
8141
  if (!sender) {
8138
- sender = pc && pc.getSenders().find(s => !s.track);
8142
+ sender = pc && pc.getSenders && pc.getSenders().find(s => !s.track);
8139
8143
  }
8140
8144
 
8141
8145
  if (sender) {
@@ -9669,7 +9673,7 @@ class WebRTCPhone extends Emitter {
9669
9673
  const screenTrack = screenShareStream.getVideoTracks()[0];
9670
9674
  const sipSession = this.currentSipSession;
9671
9675
  const pc = this.client.getPeerConnection(this.getSipSessionId(sipSession));
9672
- const sender = pc && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
9676
+ const sender = pc && pc.getSenders && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
9673
9677
  const localStream = this.client.getLocalStream(this.getSipSessionId(sipSession));
9674
9678
  const videoTrack = localStream ? localStream.getTracks().find(track => track.kind === 'video') : null;
9675
9679
  const hadVideo = !!videoTrack;
@@ -9751,7 +9755,7 @@ class WebRTCPhone extends Emitter {
9751
9755
  const screenTrack = screenStream.getVideoTracks()[0];
9752
9756
  const sipSessionId = this.getSipSessionId(sipSession);
9753
9757
  const pc = this.client.getPeerConnection(sipSessionId);
9754
- const sender = pc && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
9758
+ const sender = pc && pc.getSenders && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
9755
9759
 
9756
9760
  logger$7.info('WebRTC phone - on screensharing', {
9757
9761
  hadVideo,
package/lib/web-rtc-client.js CHANGED
@@ -6423,9 +6423,11 @@ class WebRTCClient extends Emitter {
6423
6423
  const videoTracks = localStream.getVideoTracks();
6424
6424
 
6425
6425
  // Remove video senders
6426
- pc.getSenders().filter(sender => sender.track && sender.track.kind === 'video').forEach(videoSender => {
6427
- videoSender.replaceTrack(null);
6428
- });
6426
+ if (pc.getSenders) {
6427
+ pc.getSenders().filter(sender => sender.track && sender.track.kind === 'video').forEach(videoSender => {
6428
+ videoSender.replaceTrack(null);
6429
+ });
6430
+ }
6429
6431
 
6430
6432
  videoTracks.forEach(videoTrack => {
6431
6433
  videoTrack.enabled = false;
@@ -6658,7 +6660,8 @@ class WebRTCClient extends Emitter {
6658
6660
  const constraints = { audio: { deviceId: { exact: deviceId } } };
6659
6661
  const stream = await navigator.mediaDevices.getUserMedia(constraints);
6660
6662
  const audioTrack = stream.getAudioTracks()[0];
6661
- const sender = pc && pc.getSenders().find(s => audioTrack && s && s.track && s.track.kind === audioTrack.kind);
6663
+ const sender = pc && pc.getSenders
6664
+ && pc.getSenders().find(s => audioTrack && s && s.track && s.track.kind === audioTrack.kind);
6662
6665
 
6663
6666
  if (sender) {
6664
6667
  if (sender.track) {
@@ -6721,10 +6724,11 @@ class WebRTCClient extends Emitter {
6721
6724
  // $FlowFixMe
6722
6725
  return navigator.mediaDevices.getUserMedia(constraints).then(async stream => {
6723
6726
  const videoTrack = stream.getVideoTracks()[0];
6724
- let sender = pc && pc.getSenders().find(s => videoTrack && s && s.track && s.track.kind === videoTrack.kind);
6727
+ let sender = pc && pc.getSenders
6728
+ && pc.getSenders().find(s => videoTrack && s && s.track && s.track.kind === videoTrack.kind);
6725
6729
  let wasTrackEnabled = false;
6726
6730
  if (!sender) {
6727
- sender = pc && pc.getSenders().find(s => !s.track);
6731
+ sender = pc && pc.getSenders && pc.getSenders().find(s => !s.track);
6728
6732
  }
6729
6733
 
6730
6734
  if (sender) {
package/package.json CHANGED
@@ -1,6 +1,6 @@
1
1
  {
2
2
  "name": "@wazo/sdk",
3
- "version": "0.38.10",
3
+ "version": "0.38.11",
4
4
  "description": "Wazo's JavaScript Software Development Kit.",
5
5
  "main": "dist/wazo-sdk.js",
6
6
  "author": "Wazo (http://wazo.io)",
package/src/domain/Phone/WebRTCPhone.js CHANGED
@@ -493,7 +493,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
493
493
  const screenTrack = screenShareStream.getVideoTracks()[0];
494
494
  const sipSession = this.currentSipSession;
495
495
  const pc = this.client.getPeerConnection(this.getSipSessionId(sipSession));
496
- const sender = pc && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
496
+ const sender = pc && pc.getSenders && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
497
497
  const localStream = this.client.getLocalStream(this.getSipSessionId(sipSession));
498
498
  const videoTrack = localStream ? localStream.getTracks().find(track => track.kind === 'video') : null;
499
499
  const hadVideo = !!videoTrack;
@@ -575,7 +575,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
575
575
  const screenTrack = screenStream.getVideoTracks()[0];
576
576
  const sipSessionId = this.getSipSessionId(sipSession);
577
577
  const pc = this.client.getPeerConnection(sipSessionId);
578
- const sender = pc && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
578
+ const sender = pc && pc.getSenders && pc.getSenders().find(s => s && s.track && s.track.kind === 'video');
579
579
 
580
580
  logger.info('WebRTC phone - on screensharing', {
581
581
  hadVideo,
package/src/web-rtc-client.js CHANGED
@@ -877,9 +877,11 @@ export default class WebRTCClient extends Emitter {
877
877
  const videoTracks = localStream.getVideoTracks();
878
878
 
879
879
  // Remove video senders
880
- pc.getSenders().filter(sender => sender.track && sender.track.kind === 'video').forEach(videoSender => {
881
- videoSender.replaceTrack(null);
882
- });
880
+ if (pc.getSenders) {
881
+ pc.getSenders().filter(sender => sender.track && sender.track.kind === 'video').forEach(videoSender => {
882
+ videoSender.replaceTrack(null);
883
+ });
884
+ }
883
885
 
884
886
  videoTracks.forEach(videoTrack => {
885
887
  videoTrack.enabled = false;
@@ -1112,7 +1114,8 @@ export default class WebRTCClient extends Emitter {
1112
1114
  const constraints = { audio: { deviceId: { exact: deviceId } } };
1113
1115
  const stream = await navigator.mediaDevices.getUserMedia(constraints);
1114
1116
  const audioTrack = stream.getAudioTracks()[0];
1115
- const sender = pc && pc.getSenders().find(s => audioTrack && s && s.track && s.track.kind === audioTrack.kind);
1117
+ const sender = pc && pc.getSenders
1118
+ && pc.getSenders().find(s => audioTrack && s && s.track && s.track.kind === audioTrack.kind);
1116
1119
 
1117
1120
  if (sender) {
1118
1121
  if (sender.track) {
@@ -1175,10 +1178,11 @@ export default class WebRTCClient extends Emitter {
1175
1178
  // $FlowFixMe
1176
1179
  return navigator.mediaDevices.getUserMedia(constraints).then(async stream => {
1177
1180
  const videoTrack = stream.getVideoTracks()[0];
1178
- let sender = pc && pc.getSenders().find(s => videoTrack && s && s.track && s.track.kind === videoTrack.kind);
1181
+ let sender = pc && pc.getSenders
1182
+ && pc.getSenders().find(s => videoTrack && s && s.track && s.track.kind === videoTrack.kind);
1179
1183
  let wasTrackEnabled = false;
1180
1184
  if (!sender) {
1181
- sender = pc && pc.getSenders().find(s => !s.track);
1185
+ sender = pc && pc.getSenders && pc.getSenders().find(s => !s.track);
1182
1186
  }
1183
1187
 
1184
1188
  if (sender) {