@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.
- package/dist/wazo-sdk.js +1 -1
- package/esm/domain/Phone/WebRTCPhone.js +4 -4
- package/esm/index.js +4 -4
- package/esm/simple/Auth.js +4 -4
- package/esm/simple/Configuration.js +4 -4
- package/esm/simple/Phone.js +4 -4
- package/esm/simple/index.js +4 -4
- package/esm/simple/room/LocalParticipant.js +4 -4
- package/esm/simple/room/Participant.js +4 -4
- package/esm/simple/room/RemoteParticipant.js +4 -4
- package/esm/simple/room/Room.js +4 -4
- package/esm/simple/room/SipRoom.js +4 -4
- package/esm/simple/room/Stream.js +4 -4
- package/esm/simple/room/__tests__/RemoteParticipant.test.js +4 -4
- package/esm/simple/utils.js +4 -4
- package/lib/domain/Phone/WebRTCPhone.js +4 -4
- package/lib/index.js +4 -4
- package/lib/simple/Auth.js +4 -4
- package/lib/simple/Configuration.js +4 -4
- package/lib/simple/Phone.js +4 -4
- package/lib/simple/index.js +4 -4
- package/lib/simple/room/LocalParticipant.js +4 -4
- package/lib/simple/room/Participant.js +4 -4
- package/lib/simple/room/RemoteParticipant.js +4 -4
- package/lib/simple/room/Room.js +4 -4
- package/lib/simple/room/SipRoom.js +4 -4
- package/lib/simple/room/Stream.js +4 -4
- package/lib/simple/room/__tests__/RemoteParticipant.test.js +4 -4
- package/lib/simple/utils.js +4 -4
- package/package.json +1 -1
- package/src/domain/Phone/WebRTCPhone.js +4 -4
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@@ -9122,7 +9122,7 @@ class WebRTCPhone extends Emitter {
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// When stopping screenshare and we had video before that, we have to re-upgrade
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// When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
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// We have to downgrade to audio.
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-
const screenshareStopped = this.currentScreenShare.
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+
const screenshareStopped = this.currentScreenShare.hadVideo;
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const constraints = { audio: false, video: screenshareStopped };
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// We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
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@@ -9566,7 +9566,7 @@ class WebRTCPhone extends Emitter {
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logger$8.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
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// Stop
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// Stop screenshare if needed
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if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
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this.lastScreenShare = this.currentScreenShare;
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await this.stopScreenSharing(false, callSession);
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@@ -10142,7 +10142,7 @@ class WebRTCPhone extends Emitter {
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});
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this.client.on(this.client.DISCONNECTED, () => {
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logger$8.
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logger$8.info('WebRTC client disconnected');
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this.eventEmitter.emit(ON_UNREGISTERED);
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@@ -10328,7 +10328,7 @@ class WebRTCPhone extends Emitter {
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endTime: fromSession ? fromSession.endTime : null,
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answered: state === sessionState.SessionState.Established,
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paused: this.client.isCallHeld(sipSession),
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-
isCaller: 'incoming' in extra ? !extra.incoming : false,
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isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
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cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
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number,
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ringing: false,
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package/lib/simple/Phone.js
CHANGED
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@@ -9122,7 +9122,7 @@ class WebRTCPhone extends Emitter {
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// When stopping screenshare and we had video before that, we have to re-upgrade
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9123
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// When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
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// We have to downgrade to audio.
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-
const screenshareStopped = this.currentScreenShare.
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+
const screenshareStopped = this.currentScreenShare.hadVideo;
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const constraints = { audio: false, video: screenshareStopped };
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// We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
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@@ -9566,7 +9566,7 @@ class WebRTCPhone extends Emitter {
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logger$8.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
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-
// Stop
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// Stop screenshare if needed
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if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
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this.lastScreenShare = this.currentScreenShare;
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await this.stopScreenSharing(false, callSession);
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@@ -10142,7 +10142,7 @@ class WebRTCPhone extends Emitter {
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});
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this.client.on(this.client.DISCONNECTED, () => {
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logger$8.
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logger$8.info('WebRTC client disconnected');
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this.eventEmitter.emit(ON_UNREGISTERED);
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@@ -10328,7 +10328,7 @@ class WebRTCPhone extends Emitter {
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endTime: fromSession ? fromSession.endTime : null,
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answered: state === sessionState.SessionState.Established,
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paused: this.client.isCallHeld(sipSession),
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-
isCaller: 'incoming' in extra ? !extra.incoming : false,
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isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
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cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
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number,
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ringing: false,
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package/lib/simple/index.js
CHANGED
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@@ -9404,7 +9404,7 @@ class WebRTCPhone extends Emitter {
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// When stopping screenshare and we had video before that, we have to re-upgrade
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9405
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// When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
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// We have to downgrade to audio.
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-
const screenshareStopped = this.currentScreenShare.
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+
const screenshareStopped = this.currentScreenShare.hadVideo;
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const constraints = { audio: false, video: screenshareStopped };
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// We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
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@@ -9848,7 +9848,7 @@ class WebRTCPhone extends Emitter {
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logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
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-
// Stop
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// Stop screenshare if needed
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if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
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this.lastScreenShare = this.currentScreenShare;
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await this.stopScreenSharing(false, callSession);
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@@ -10424,7 +10424,7 @@ class WebRTCPhone extends Emitter {
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});
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this.client.on(this.client.DISCONNECTED, () => {
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logger$7.
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logger$7.info('WebRTC client disconnected');
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this.eventEmitter.emit(ON_UNREGISTERED);
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@@ -10610,7 +10610,7 @@ class WebRTCPhone extends Emitter {
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endTime: fromSession ? fromSession.endTime : null,
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answered: state === sessionState.SessionState.Established,
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paused: this.client.isCallHeld(sipSession),
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-
isCaller: 'incoming' in extra ? !extra.incoming : false,
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isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
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cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
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number,
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ringing: false,
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@@ -9404,7 +9404,7 @@ class WebRTCPhone extends Emitter {
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9404
9404
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// When stopping screenshare and we had video before that, we have to re-upgrade
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9405
9405
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// When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
|
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9406
9406
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// We have to downgrade to audio.
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9407
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-
const screenshareStopped = this.currentScreenShare.
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+
const screenshareStopped = this.currentScreenShare.hadVideo;
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const constraints = { audio: false, video: screenshareStopped };
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// We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
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@@ -9848,7 +9848,7 @@ class WebRTCPhone extends Emitter {
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logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
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-
// Stop
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+
// Stop screenshare if needed
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if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
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this.lastScreenShare = this.currentScreenShare;
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await this.stopScreenSharing(false, callSession);
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@@ -10424,7 +10424,7 @@ class WebRTCPhone extends Emitter {
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});
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this.client.on(this.client.DISCONNECTED, () => {
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logger$7.
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logger$7.info('WebRTC client disconnected');
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this.eventEmitter.emit(ON_UNREGISTERED);
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@@ -10610,7 +10610,7 @@ class WebRTCPhone extends Emitter {
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endTime: fromSession ? fromSession.endTime : null,
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answered: state === sessionState.SessionState.Established,
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paused: this.client.isCallHeld(sipSession),
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-
isCaller: 'incoming' in extra ? !extra.incoming : false,
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+
isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
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cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
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10615
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number,
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ringing: false,
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@@ -9404,7 +9404,7 @@ class WebRTCPhone extends Emitter {
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9404
9404
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// When stopping screenshare and we had video before that, we have to re-upgrade
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|
9405
9405
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// When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
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9406
9406
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// We have to downgrade to audio.
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9407
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-
const screenshareStopped = this.currentScreenShare.
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9407
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+
const screenshareStopped = this.currentScreenShare.hadVideo;
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const constraints = { audio: false, video: screenshareStopped };
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// We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
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@@ -9848,7 +9848,7 @@ class WebRTCPhone extends Emitter {
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9848
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logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
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9850
9850
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9851
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-
// Stop
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9851
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+
// Stop screenshare if needed
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9852
9852
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if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
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9853
9853
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this.lastScreenShare = this.currentScreenShare;
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9854
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await this.stopScreenSharing(false, callSession);
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@@ -10424,7 +10424,7 @@ class WebRTCPhone extends Emitter {
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});
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this.client.on(this.client.DISCONNECTED, () => {
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10427
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logger$7.
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+
logger$7.info('WebRTC client disconnected');
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10428
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this.eventEmitter.emit(ON_UNREGISTERED);
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@@ -10610,7 +10610,7 @@ class WebRTCPhone extends Emitter {
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endTime: fromSession ? fromSession.endTime : null,
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answered: state === sessionState.SessionState.Established,
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paused: this.client.isCallHeld(sipSession),
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10613
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-
isCaller: 'incoming' in extra ? !extra.incoming : false,
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10613
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+
isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
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10614
10614
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cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
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10615
10615
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number,
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10616
10616
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ringing: false,
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@@ -9404,7 +9404,7 @@ class WebRTCPhone extends Emitter {
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|
|
9404
9404
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// When stopping screenshare and we had video before that, we have to re-upgrade
|
|
9405
9405
|
// When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
|
|
9406
9406
|
// We have to downgrade to audio.
|
|
9407
|
-
const screenshareStopped = this.currentScreenShare.
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|
9407
|
+
const screenshareStopped = this.currentScreenShare.hadVideo;
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9408
9408
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const constraints = { audio: false, video: screenshareStopped };
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9409
9409
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9410
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// We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
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|
@@ -9848,7 +9848,7 @@ class WebRTCPhone extends Emitter {
|
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9848
9848
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9849
9849
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logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
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9850
9850
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|
|
9851
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-
// Stop
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|
9851
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+
// Stop screenshare if needed
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|
9852
9852
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if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
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9853
9853
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this.lastScreenShare = this.currentScreenShare;
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9854
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await this.stopScreenSharing(false, callSession);
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@@ -10424,7 +10424,7 @@ class WebRTCPhone extends Emitter {
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10424
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});
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10425
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this.client.on(this.client.DISCONNECTED, () => {
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10427
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-
logger$7.
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10427
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+
logger$7.info('WebRTC client disconnected');
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10428
10428
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this.eventEmitter.emit(ON_UNREGISTERED);
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@@ -10610,7 +10610,7 @@ class WebRTCPhone extends Emitter {
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10610
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endTime: fromSession ? fromSession.endTime : null,
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10611
10611
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answered: state === sessionState.SessionState.Established,
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10612
10612
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paused: this.client.isCallHeld(sipSession),
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10613
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-
isCaller: 'incoming' in extra ? !extra.incoming : false,
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10613
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+
isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
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10614
10614
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cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
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10615
10615
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number,
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10616
10616
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ringing: false,
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package/lib/simple/room/Room.js
CHANGED
|
@@ -9406,7 +9406,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9406
9406
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// When stopping screenshare and we had video before that, we have to re-upgrade
|
|
9407
9407
|
// When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
|
|
9408
9408
|
// We have to downgrade to audio.
|
|
9409
|
-
const screenshareStopped = this.currentScreenShare.
|
|
9409
|
+
const screenshareStopped = this.currentScreenShare.hadVideo;
|
|
9410
9410
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const constraints = { audio: false, video: screenshareStopped };
|
|
9411
9411
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|
|
9412
9412
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// We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
|
|
@@ -9850,7 +9850,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9850
9850
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9851
9851
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logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
|
|
9852
9852
|
|
|
9853
|
-
// Stop
|
|
9853
|
+
// Stop screenshare if needed
|
|
9854
9854
|
if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
|
|
9855
9855
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this.lastScreenShare = this.currentScreenShare;
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9856
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await this.stopScreenSharing(false, callSession);
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@@ -10426,7 +10426,7 @@ class WebRTCPhone extends Emitter {
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10426
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});
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10427
10427
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this.client.on(this.client.DISCONNECTED, () => {
|
|
10429
|
-
logger$7.
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|
10429
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+
logger$7.info('WebRTC client disconnected');
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10430
10430
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10431
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this.eventEmitter.emit(ON_UNREGISTERED);
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10432
10432
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|
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@@ -10612,7 +10612,7 @@ class WebRTCPhone extends Emitter {
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10612
10612
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endTime: fromSession ? fromSession.endTime : null,
|
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10613
10613
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answered: state === sessionState.SessionState.Established,
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10614
10614
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paused: this.client.isCallHeld(sipSession),
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10615
|
-
isCaller: 'incoming' in extra ? !extra.incoming : false,
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10615
|
+
isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
|
|
10616
10616
|
cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
|
|
10617
10617
|
number,
|
|
10618
10618
|
ringing: false,
|
|
@@ -9404,7 +9404,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9404
9404
|
// When stopping screenshare and we had video before that, we have to re-upgrade
|
|
9405
9405
|
// When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
|
|
9406
9406
|
// We have to downgrade to audio.
|
|
9407
|
-
const screenshareStopped = this.currentScreenShare.
|
|
9407
|
+
const screenshareStopped = this.currentScreenShare.hadVideo;
|
|
9408
9408
|
const constraints = { audio: false, video: screenshareStopped };
|
|
9409
9409
|
|
|
9410
9410
|
// We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
|
|
@@ -9848,7 +9848,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9848
9848
|
|
|
9849
9849
|
logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
|
|
9850
9850
|
|
|
9851
|
-
// Stop
|
|
9851
|
+
// Stop screenshare if needed
|
|
9852
9852
|
if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
|
|
9853
9853
|
this.lastScreenShare = this.currentScreenShare;
|
|
9854
9854
|
await this.stopScreenSharing(false, callSession);
|
|
@@ -10424,7 +10424,7 @@ class WebRTCPhone extends Emitter {
|
|
|
10424
10424
|
});
|
|
10425
10425
|
|
|
10426
10426
|
this.client.on(this.client.DISCONNECTED, () => {
|
|
10427
|
-
logger$7.
|
|
10427
|
+
logger$7.info('WebRTC client disconnected');
|
|
10428
10428
|
|
|
10429
10429
|
this.eventEmitter.emit(ON_UNREGISTERED);
|
|
10430
10430
|
|
|
@@ -10610,7 +10610,7 @@ class WebRTCPhone extends Emitter {
|
|
|
10610
10610
|
endTime: fromSession ? fromSession.endTime : null,
|
|
10611
10611
|
answered: state === sessionState.SessionState.Established,
|
|
10612
10612
|
paused: this.client.isCallHeld(sipSession),
|
|
10613
|
-
isCaller: 'incoming' in extra ? !extra.incoming : false,
|
|
10613
|
+
isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
|
|
10614
10614
|
cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
|
|
10615
10615
|
number,
|
|
10616
10616
|
ringing: false,
|
|
@@ -9404,7 +9404,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9404
9404
|
// When stopping screenshare and we had video before that, we have to re-upgrade
|
|
9405
9405
|
// When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
|
|
9406
9406
|
// We have to downgrade to audio.
|
|
9407
|
-
const screenshareStopped = this.currentScreenShare.
|
|
9407
|
+
const screenshareStopped = this.currentScreenShare.hadVideo;
|
|
9408
9408
|
const constraints = { audio: false, video: screenshareStopped };
|
|
9409
9409
|
|
|
9410
9410
|
// We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
|
|
@@ -9848,7 +9848,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9848
9848
|
|
|
9849
9849
|
logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
|
|
9850
9850
|
|
|
9851
|
-
// Stop
|
|
9851
|
+
// Stop screenshare if needed
|
|
9852
9852
|
if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
|
|
9853
9853
|
this.lastScreenShare = this.currentScreenShare;
|
|
9854
9854
|
await this.stopScreenSharing(false, callSession);
|
|
@@ -10424,7 +10424,7 @@ class WebRTCPhone extends Emitter {
|
|
|
10424
10424
|
});
|
|
10425
10425
|
|
|
10426
10426
|
this.client.on(this.client.DISCONNECTED, () => {
|
|
10427
|
-
logger$7.
|
|
10427
|
+
logger$7.info('WebRTC client disconnected');
|
|
10428
10428
|
|
|
10429
10429
|
this.eventEmitter.emit(ON_UNREGISTERED);
|
|
10430
10430
|
|
|
@@ -10610,7 +10610,7 @@ class WebRTCPhone extends Emitter {
|
|
|
10610
10610
|
endTime: fromSession ? fromSession.endTime : null,
|
|
10611
10611
|
answered: state === sessionState.SessionState.Established,
|
|
10612
10612
|
paused: this.client.isCallHeld(sipSession),
|
|
10613
|
-
isCaller: 'incoming' in extra ? !extra.incoming : false,
|
|
10613
|
+
isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
|
|
10614
10614
|
cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
|
|
10615
10615
|
number,
|
|
10616
10616
|
ringing: false,
|
|
@@ -9404,7 +9404,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9404
9404
|
// When stopping screenshare and we had video before that, we have to re-upgrade
|
|
9405
9405
|
// When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
|
|
9406
9406
|
// We have to downgrade to audio.
|
|
9407
|
-
const screenshareStopped = this.currentScreenShare.
|
|
9407
|
+
const screenshareStopped = this.currentScreenShare.hadVideo;
|
|
9408
9408
|
const constraints = { audio: false, video: screenshareStopped };
|
|
9409
9409
|
|
|
9410
9410
|
// We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
|
|
@@ -9848,7 +9848,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9848
9848
|
|
|
9849
9849
|
logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
|
|
9850
9850
|
|
|
9851
|
-
// Stop
|
|
9851
|
+
// Stop screenshare if needed
|
|
9852
9852
|
if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
|
|
9853
9853
|
this.lastScreenShare = this.currentScreenShare;
|
|
9854
9854
|
await this.stopScreenSharing(false, callSession);
|
|
@@ -10424,7 +10424,7 @@ class WebRTCPhone extends Emitter {
|
|
|
10424
10424
|
});
|
|
10425
10425
|
|
|
10426
10426
|
this.client.on(this.client.DISCONNECTED, () => {
|
|
10427
|
-
logger$7.
|
|
10427
|
+
logger$7.info('WebRTC client disconnected');
|
|
10428
10428
|
|
|
10429
10429
|
this.eventEmitter.emit(ON_UNREGISTERED);
|
|
10430
10430
|
|
|
@@ -10610,7 +10610,7 @@ class WebRTCPhone extends Emitter {
|
|
|
10610
10610
|
endTime: fromSession ? fromSession.endTime : null,
|
|
10611
10611
|
answered: state === sessionState.SessionState.Established,
|
|
10612
10612
|
paused: this.client.isCallHeld(sipSession),
|
|
10613
|
-
isCaller: 'incoming' in extra ? !extra.incoming : false,
|
|
10613
|
+
isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
|
|
10614
10614
|
cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
|
|
10615
10615
|
number,
|
|
10616
10616
|
ringing: false,
|
package/lib/simple/utils.js
CHANGED
|
@@ -9406,7 +9406,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9406
9406
|
// When stopping screenshare and we had video before that, we have to re-upgrade
|
|
9407
9407
|
// When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
|
|
9408
9408
|
// We have to downgrade to audio.
|
|
9409
|
-
const screenshareStopped = this.currentScreenShare.
|
|
9409
|
+
const screenshareStopped = this.currentScreenShare.hadVideo;
|
|
9410
9410
|
const constraints = { audio: false, video: screenshareStopped };
|
|
9411
9411
|
|
|
9412
9412
|
// We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
|
|
@@ -9850,7 +9850,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9850
9850
|
|
|
9851
9851
|
logger$7.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
|
|
9852
9852
|
|
|
9853
|
-
// Stop
|
|
9853
|
+
// Stop screenshare if needed
|
|
9854
9854
|
if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
|
|
9855
9855
|
this.lastScreenShare = this.currentScreenShare;
|
|
9856
9856
|
await this.stopScreenSharing(false, callSession);
|
|
@@ -10426,7 +10426,7 @@ class WebRTCPhone extends Emitter {
|
|
|
10426
10426
|
});
|
|
10427
10427
|
|
|
10428
10428
|
this.client.on(this.client.DISCONNECTED, () => {
|
|
10429
|
-
logger$7.
|
|
10429
|
+
logger$7.info('WebRTC client disconnected');
|
|
10430
10430
|
|
|
10431
10431
|
this.eventEmitter.emit(ON_UNREGISTERED);
|
|
10432
10432
|
|
|
@@ -10612,7 +10612,7 @@ class WebRTCPhone extends Emitter {
|
|
|
10612
10612
|
endTime: fromSession ? fromSession.endTime : null,
|
|
10613
10613
|
answered: state === sessionState.SessionState.Established,
|
|
10614
10614
|
paused: this.client.isCallHeld(sipSession),
|
|
10615
|
-
isCaller: 'incoming' in extra ? !extra.incoming : false,
|
|
10615
|
+
isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
|
|
10616
10616
|
cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
|
|
10617
10617
|
number,
|
|
10618
10618
|
ringing: false,
|
package/package.json
CHANGED
|
@@ -543,7 +543,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
|
|
|
543
543
|
// When stopping screenshare and we had video before that, we have to re-upgrade
|
|
544
544
|
// When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
|
|
545
545
|
// We have to downgrade to audio.
|
|
546
|
-
const screenshareStopped = this.currentScreenShare.
|
|
546
|
+
const screenshareStopped = this.currentScreenShare.hadVideo;
|
|
547
547
|
const constraints = { audio: false, video: screenshareStopped };
|
|
548
548
|
|
|
549
549
|
// We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
|
|
@@ -987,7 +987,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
|
|
|
987
987
|
|
|
988
988
|
logger.info('WebRTC hold sip session', { sipId: sipSession.id, hasVideo });
|
|
989
989
|
|
|
990
|
-
// Stop
|
|
990
|
+
// Stop screenshare if needed
|
|
991
991
|
if (this.currentScreenShare && this.currentScreenShare.sipSessionId === sessionId) {
|
|
992
992
|
this.lastScreenShare = this.currentScreenShare;
|
|
993
993
|
await this.stopScreenSharing(false, callSession);
|
|
@@ -1563,7 +1563,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
|
|
|
1563
1563
|
});
|
|
1564
1564
|
|
|
1565
1565
|
this.client.on(this.client.DISCONNECTED, () => {
|
|
1566
|
-
logger.
|
|
1566
|
+
logger.info('WebRTC client disconnected');
|
|
1567
1567
|
|
|
1568
1568
|
this.eventEmitter.emit(ON_UNREGISTERED);
|
|
1569
1569
|
|
|
@@ -1751,7 +1751,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
|
|
|
1751
1751
|
endTime: fromSession ? fromSession.endTime : null,
|
|
1752
1752
|
answered: state === SessionState.Established,
|
|
1753
1753
|
paused: this.client.isCallHeld(sipSession),
|
|
1754
|
-
isCaller: 'incoming' in extra ? !extra.incoming : false,
|
|
1754
|
+
isCaller: fromSession ? fromSession.isCaller : ('incoming' in extra ? !extra.incoming : false),
|
|
1755
1755
|
cameraEnabled: fromSession ? fromSession.isCameraEnabled() : this.client.hasVideo(sessionId),
|
|
1756
1756
|
number,
|
|
1757
1757
|
ringing: false,
|