diff --git a/Telegram/SourceFiles/calls/calls_call.cpp b/Telegram/SourceFiles/calls/calls_call.cpp
index ba59556d9..486d5af2a 100644
--- a/Telegram/SourceFiles/calls/calls_call.cpp
+++ b/Telegram/SourceFiles/calls/calls_call.cpp
@@ -420,6 +420,14 @@ void Call::actuallyAnswer() {
 	}).send();
 }
 
+void Call::captureMuteChanged(bool mute) {
+	setMuted(mute);
+}
+
+rpl::producer<Webrtc::DeviceResolvedId> Call::captureMuteDeviceId() {
+	return _captureDeviceId.value();
+}
+
 void Call::setMuted(bool mute) {
 	_muted = mute;
 	if (_instance) {
@@ -1033,6 +1041,20 @@ void Call::createAndStartController(const MTPDphoneCall &call) {
 
 	raw->setIncomingVideoOutput(_videoIncoming->sink());
 	raw->setAudioOutputDuckingEnabled(settings.callAudioDuckingEnabled());
+
+	_state.value() | rpl::start_with_next([=](State state) {
+		const auto track = (state != State::FailedHangingUp)
+			&& (state != State::Failed)
+			&& (state != State::HangingUp)
+			&& (state != State::Ended)
+			&& (state != State::EndedByOtherDevice)
+			&& (state != State::Busy);
+		Core::App().mediaDevices().setCaptureMuteTracker(this, track);
+	}, _instanceLifetime);
+
+	_muted.value() | rpl::start_with_next([=](bool muted) {
+		Core::App().mediaDevices().setCaptureMuted(muted);
+	}, _instanceLifetime);
 }
 
 void Call::handleControllerStateChange(tgcalls::State state) {
@@ -1375,6 +1397,9 @@ void Call::handleControllerError(const QString &error) {
 }
 
 void Call::destroyController() {
+	_instanceLifetime.destroy();
+	Core::App().mediaDevices().setCaptureMuteTracker(this, false);
+
 	if (_instance) {
 		_instance->stop([](tgcalls::FinalState) {
 		});
diff --git a/Telegram/SourceFiles/calls/calls_call.h b/Telegram/SourceFiles/calls/calls_call.h
index 4be689831..69c2d94da 100644
--- a/Telegram/SourceFiles/calls/calls_call.h
+++ b/Telegram/SourceFiles/calls/calls_call.h
@@ -59,7 +59,9 @@ enum class CallType {
 	Outgoing,
 };
 
-class Call : public base::has_weak_ptr {
+class Call final
+	: public base::has_weak_ptr
+	, private Webrtc::CaptureMuteTracker {
 public:
 	class Delegate {
 	public:
@@ -249,6 +251,9 @@ private:
 	void setSignalBarCount(int count);
 	void destroyController();
 
+	void captureMuteChanged(bool mute) override;
+	rpl::producer<Webrtc::DeviceResolvedId> captureMuteDeviceId() override;
+
 	void setupMediaDevices();
 	void setupOutgoingVideo();
 	void updateRemoteMediaState(
@@ -298,6 +303,7 @@ private:
 
 	std::unique_ptr<Media::Audio::Track> _waitingTrack;
 
+	rpl::lifetime _instanceLifetime;
 	rpl::lifetime _lifetime;
 
 };
diff --git a/Telegram/SourceFiles/calls/group/calls_group_call.cpp b/Telegram/SourceFiles/calls/group/calls_group_call.cpp
index 8617aef85..5f7346913 100644
--- a/Telegram/SourceFiles/calls/group/calls_group_call.cpp
+++ b/Telegram/SourceFiles/calls/group/calls_group_call.cpp
@@ -30,6 +30,7 @@ https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
 #include "base/random.h"
 #include "webrtc/webrtc_video_track.h"
 #include "webrtc/webrtc_create_adm.h"
+#include "webrtc/webrtc_environment.h"
 
 #include <tgcalls/group/GroupInstanceCustomImpl.h>
 #include <tgcalls/VideoCaptureInterface.h>
@@ -667,6 +668,8 @@ GroupCall::GroupCall(
 GroupCall::~GroupCall() {
 	destroyScreencast();
 	destroyController();
+
+	Core::App().mediaDevices().setCaptureMuteTracker(this, false);
 }
 
 bool GroupCall::isSharingScreen() const {
@@ -2087,6 +2090,32 @@ void GroupCall::setupMediaDevices() {
 	}) | rpl::start_with_next([=](const Webrtc::DeviceResolvedId &deviceId) {
 		_cameraCapture->switchToDevice(deviceId.value.toStdString(), false);
 	}, _lifetime);
+
+	_muted.value() | rpl::start_with_next([=](MuteState state) {
+		const auto devices = &Core::App().mediaDevices();
+		const auto muted = (state != MuteState::Active)
+			&& (state != MuteState::PushToTalk);
+		const auto track = !muted || (state == MuteState::Muted);
+		devices->setCaptureMuteTracker(this, track);
+		devices->setCaptureMuted(muted);
+	}, _lifetime);
+}
+
+void GroupCall::captureMuteChanged(bool mute) {
+	const auto oldState = muted();
+	if (mute
+		&& (oldState == MuteState::ForceMuted
+			|| oldState == MuteState::RaisedHand
+			|| oldState == MuteState::Muted)) {
+		return;
+	} else if (!mute && oldState != MuteState::Muted) {
+		return;
+	}
+	setMutedAndUpdate(mute ? MuteState::Muted : MuteState::Active);
+}
+
+rpl::producer<Webrtc::DeviceResolvedId> GroupCall::captureMuteDeviceId() {
+	return _captureDeviceId.value();
 }
 
 int GroupCall::activeVideoSendersCount() const {
diff --git a/Telegram/SourceFiles/calls/group/calls_group_call.h b/Telegram/SourceFiles/calls/group/calls_group_call.h
index fd2cd88a0..8f93793bd 100644
--- a/Telegram/SourceFiles/calls/group/calls_group_call.h
+++ b/Telegram/SourceFiles/calls/group/calls_group_call.h
@@ -12,6 +12,7 @@ https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
 #include "base/bytes.h"
 #include "mtproto/sender.h"
 #include "mtproto/mtproto_auth_key.h"
+#include "webrtc/webrtc_device_common.h"
 #include "webrtc/webrtc_device_resolver.h"
 
 class History;
@@ -175,7 +176,9 @@ struct ParticipantVideoParams;
 [[nodiscard]] uint32 GetAdditionalAudioSsrc(
 	const std::shared_ptr<ParticipantVideoParams> &params);
 
-class GroupCall final : public base::has_weak_ptr {
+class GroupCall final
+	: public base::has_weak_ptr
+	, private Webrtc::CaptureMuteTracker {
 public:
 	class Delegate {
 	public:
@@ -550,6 +553,9 @@ private:
 	void applySelfUpdate(const MTPDgroupCallParticipant &data);
 	void applyOtherParticipantUpdate(const MTPDgroupCallParticipant &data);
 
+	void captureMuteChanged(bool mute) override;
+	rpl::producer<Webrtc::DeviceResolvedId> captureMuteDeviceId() override;
+
 	void setupMediaDevices();
 	void setupOutgoingVideo();
 	void setScreenEndpoint(std::string endpoint);
diff --git a/Telegram/lib_webrtc b/Telegram/lib_webrtc
index 5493af61d..1cbf5fa7d 160000
--- a/Telegram/lib_webrtc
+++ b/Telegram/lib_webrtc
@@ -1 +1 @@
-Subproject commit 5493af61df5cb90a30b686296521961763a009e0
+Subproject commit 1cbf5fa7d875074c40e76216a3047bd7c59996d7