diff --git a/src/utils/webrtc/simplewebrtc/localmedia.js b/src/utils/webrtc/simplewebrtc/localmedia.js
index 35c1563a931..bcb69904db8 100644
--- a/src/utils/webrtc/simplewebrtc/localmedia.js
+++ b/src/utils/webrtc/simplewebrtc/localmedia.js
@@ -144,6 +144,10 @@ LocalMedia.prototype.start = function(mediaConstraints, cb, context) {
 		webrtcIndex.mediaDevicesManager.disableDeviceEvents()
 	}
 
+	// The handlers for "change:audioInputId" and "change:videoInputId" events
+	// expect the initial "getUserMedia" call to have been completed before
+	// being used, so they must be set when the promise is resolved or rejected.
+
 	webrtcIndex.mediaDevicesManager.getUserMedia(constraints).then(function(stream) {
 		// Although the promise should be resolved only if all the constraints
 		// are met Edge resolves it if both audio and video are requested but
@@ -187,8 +191,8 @@ LocalMedia.prototype.start = function(mediaConstraints, cb, context) {
 		}
 	}).catch(function(err) {
 		// Fallback for users without a camera or with a camera that can not be
-		// accessed.
-		if (self.config.audioFallback && constraints.video !== false) {
+		// accessed, but only if audio is meant to be used.
+		if (constraints.audio !== false && self.config.audioFallback && constraints.video !== false) {
 			self.emit('localStreamRequestFailedRetryNoVideo', constraints, err)
 			constraints.video = false
 			self.start(constraints, cb, 'retry-no-video')
@@ -197,6 +201,9 @@ LocalMedia.prototype.start = function(mediaConstraints, cb, context) {
 
 		self.emit('localStreamRequestFailed', constraints)
 
+		webrtcIndex.mediaDevicesManager.on('change:audioInputId', self._handleAudioInputIdChangedBound)
+		webrtcIndex.mediaDevicesManager.on('change:videoInputId', self._handleVideoInputIdChangedBound)
+
 		if (cb) {
 			return cb(err, null)
 		}
@@ -484,10 +491,8 @@ LocalMedia.prototype.stop = function(stream) {
 	this.stopStream(stream)
 	this.stopScreenShare(stream)
 
-	if (!this.localStreams.length) {
-		webrtcIndex.mediaDevicesManager.off('change:audioInputId', this._handleAudioInputIdChangedBound)
-		webrtcIndex.mediaDevicesManager.off('change:videoInputId', this._handleVideoInputIdChangedBound)
-	}
+	webrtcIndex.mediaDevicesManager.off('change:audioInputId', this._handleAudioInputIdChangedBound)
+	webrtcIndex.mediaDevicesManager.off('change:videoInputId', this._handleVideoInputIdChangedBound)
 }
 
 LocalMedia.prototype.stopStream = function(stream) {
diff --git a/src/utils/webrtc/webrtc.js b/src/utils/webrtc/webrtc.js
index 5be62030f84..085e32ffe32 100644
--- a/src/utils/webrtc/webrtc.js
+++ b/src/utils/webrtc/webrtc.js
@@ -48,6 +48,7 @@ let callParticipantCollection = null
 let localCallParticipantModel = null
 let showedTURNWarning = false
 let sendCurrentStateWithRepetitionTimeout = null
+let startedWithMedia
 
 function arrayDiff(a, b) {
 	return a.filter(function(i) {
@@ -524,6 +525,8 @@ export default function initWebRTC(signaling, _callParticipantCollection, _local
 	})
 
 	webrtc.startMedia = function(token) {
+		startedWithMedia = undefined
+
 		webrtc.joinCall(token)
 	}
 
@@ -844,6 +847,8 @@ export default function initWebRTC(signaling, _callParticipantCollection, _local
 		// reconnection is forced to start sending it.
 		signaling.setSendVideoIfAvailable(true)
 
+		startedWithMedia = true
+
 		let flags = signaling.getCurrentCallFlags()
 		flags |= PARTICIPANT.CALL_FLAG.WITH_VIDEO
 
@@ -909,9 +914,38 @@ export default function initWebRTC(signaling, _callParticipantCollection, _local
 		}
 	})
 
+	webrtc.on('localTrackReplaced', function(newTrack /*, oldTrack, stream */) {
+		// Device disabled, nothing to do here.
+		if (!newTrack) {
+			return
+		}
+
+		// If the call was started with media the connections will be already
+		// established. If it has not started yet the connections will be
+		// established once started.
+		if (startedWithMedia || startedWithMedia === undefined) {
+			return
+		}
+
+		// If the call was originally started without media the participant
+		// needs to reconnect to establish the sender connections.
+		startedWithMedia = true
+
+		let flags = signaling.getCurrentCallFlags()
+		if (newTrack.kind === 'audio') {
+			flags |= PARTICIPANT.CALL_FLAG.WITH_AUDIO
+		} else if (newTrack.kind === 'video') {
+			flags |= PARTICIPANT.CALL_FLAG.WITH_VIDEO
+		}
+
+		forceReconnect(signaling, flags)
+	})
+
 	webrtc.on('localMediaStarted', function(/* configuration */) {
 		console.info('localMediaStarted')
 
+		startedWithMedia = true
+
 		clearLocalStreamRequestedTimeoutAndHideNotification()
 
 		if (signaling.hasFeature('mcu')) {
@@ -922,8 +956,16 @@ export default function initWebRTC(signaling, _callParticipantCollection, _local
 	webrtc.on('localMediaError', function(error) {
 		console.warn('Access to microphone & camera failed', error)
 
+		startedWithMedia = false
+
 		clearLocalStreamRequestedTimeoutAndHideNotification()
 
+		if (error.name === 'TypeError') {
+			// Both audio and video were explicitly disabled, no need to show an
+			// error.
+			return
+		}
+
 		let message
 		let timeout = TOAST_PERMANENT_TIMEOUT
 		if ((error.name === 'NotSupportedError'