diff --git a/Telegram/SourceFiles/calls/calls_call.cpp b/Telegram/SourceFiles/calls/calls_call.cpp
index 2fe91d7304..a855a6471c 100644
--- a/Telegram/SourceFiles/calls/calls_call.cpp
+++ b/Telegram/SourceFiles/calls/calls_call.cpp
@@ -87,7 +87,7 @@ void AppendServer(
 				if (host.isEmpty()) {
 					return;
 				}
-				list.push_back({
+				list.push_back(tgcalls::RtcServer{
 					.host = host.toStdString(),
 					.port = port,
 					.isTurn = false
@@ -100,7 +100,7 @@ void AppendServer(
 		const auto password = qs(data.vpassword());
 		if (data.is_turn() && !username.isEmpty() && !password.isEmpty()) {
 			const auto pushTurn = [&](const QString &host) {
-				list.push_back({
+				list.push_back(tgcalls::RtcServer{
 					.host = host.toStdString(),
 					.port = port,
 					.login = username.toStdString(),
@@ -143,8 +143,8 @@ uint64 ComputeFingerprint(bytes::const_span authKey) {
 	return WrapVersions(tgcalls::Meta::Versions() | ranges::action::reverse);
 }
 
-[[nodiscard]] webrtc::VideoState StartVideoState(bool enabled) {
-	using State = webrtc::VideoState;
+[[nodiscard]] Webrtc::VideoState StartVideoState(bool enabled) {
+	using State = Webrtc::VideoState;
 	return enabled ? State::Active : State::Inactive;
 }
 
@@ -159,8 +159,8 @@ Call::Call(
 , _user(user)
 , _api(&_user->session().mtp())
 , _type(type)
-, _videoIncoming(std::make_unique<webrtc::VideoTrack>(StartVideoState(video)))
-, _videoOutgoing(std::make_unique<webrtc::VideoTrack>(StartVideoState(video))) {
+, _videoIncoming(std::make_unique<Webrtc::VideoTrack>(StartVideoState(video)))
+, _videoOutgoing(std::make_unique<Webrtc::VideoTrack>(StartVideoState(video))) {
 	_discardByTimeoutTimer.setCallback([=] { hangup(); });
 
 	if (_type == Type::Outgoing) {
@@ -345,14 +345,14 @@ void Call::setMuted(bool mute) {
 void Call::setupOutgoingVideo() {
 	const auto started = _videoOutgoing->state();
 	_videoOutgoing->stateValue(
-	) | rpl::start_with_next([=](webrtc::VideoState state) {
+	) | rpl::start_with_next([=](Webrtc::VideoState state) {
 		if (_state.current() != State::Established
 			&& state != started
 			&& !_videoCapture) {
 			_videoOutgoing->setState(started);
-		} else if (state != webrtc::VideoState::Inactive) {
+		} else if (state != Webrtc::VideoState::Inactive) {
 			// Paused not supported right now.
-			Assert(state == webrtc::VideoState::Active);
+			Assert(state == Webrtc::VideoState::Active);
 			if (!_videoCapture) {
 				_videoCapture = tgcalls::VideoCaptureInterface::Create();
 				_videoCapture->setOutput(_videoOutgoing->sink());
@@ -367,11 +367,11 @@ void Call::setupOutgoingVideo() {
 	}, _lifetime);
 }
 
-not_null<webrtc::VideoTrack*> Call::videoIncoming() const {
+not_null<Webrtc::VideoTrack*> Call::videoIncoming() const {
 	return _videoIncoming.get();
 }
 
-not_null<webrtc::VideoTrack*> Call::videoOutgoing() const {
+not_null<Webrtc::VideoTrack*> Call::videoOutgoing() const {
 	return _videoOutgoing.get();
 }
 
@@ -589,7 +589,7 @@ void Call::updateRemoteMediaState(
 	}();
 	_videoIncoming->setState([&] {
 		using From = tgcalls::VideoState;
-		using To = webrtc::VideoState;
+		using To = Webrtc::VideoState;
 		switch (video) {
 		case From::Inactive: return To::Inactive;
 		case From::Paused: return To::Paused;
@@ -992,8 +992,8 @@ void Call::finish(FinishType type, const MTPPhoneCallDiscardReason &reason) {
 	auto duration = getDurationMs() / 1000;
 	auto connectionId = _instance ? _instance->getPreferredRelayId() : 0;
 	_finishByTimeoutTimer.call(kHangupTimeoutMs, [this, finalState] { setState(finalState); });
-	const auto flags = ((_videoIncoming->state() != webrtc::VideoState::Inactive)
-		|| (_videoOutgoing->state() != webrtc::VideoState::Inactive))
+	const auto flags = ((_videoIncoming->state() != Webrtc::VideoState::Inactive)
+		|| (_videoOutgoing->state() != Webrtc::VideoState::Inactive))
 		? MTPphone_DiscardCall::Flag::f_video
 		: MTPphone_DiscardCall::Flag(0);
 	_api.request(MTPphone_DiscardCall(
diff --git a/Telegram/SourceFiles/calls/calls_call.h b/Telegram/SourceFiles/calls/calls_call.h
index 43f9928b5d..56b9806dc0 100644
--- a/Telegram/SourceFiles/calls/calls_call.h
+++ b/Telegram/SourceFiles/calls/calls_call.h
@@ -27,10 +27,10 @@ enum class VideoState;
 enum class AudioState;
 } // namespace tgcalls
 
-namespace webrtc {
+namespace Webrtc {
 enum class VideoState;
 class VideoTrack;
-} // namespace webrtc
+} // namespace Webrtc
 
 namespace Calls {
 
@@ -117,11 +117,11 @@ public:
 		return _remoteAudioState.value();
 	}
 
-	[[nodiscard]] webrtc::VideoState remoteVideoState() const {
+	[[nodiscard]] Webrtc::VideoState remoteVideoState() const {
 		return _remoteVideoState.current();
 	}
 	[[nodiscard]] auto remoteVideoStateValue() const
-	-> rpl::producer<webrtc::VideoState> {
+	-> rpl::producer<Webrtc::VideoState> {
 		return _remoteVideoState.value();
 	}
 
@@ -140,8 +140,8 @@ public:
 		return _muted.value();
 	}
 
-	[[nodiscard]] not_null<webrtc::VideoTrack*> videoIncoming() const;
-	[[nodiscard]] not_null<webrtc::VideoTrack*> videoOutgoing() const;
+	[[nodiscard]] not_null<Webrtc::VideoTrack*> videoIncoming() const;
+	[[nodiscard]] not_null<Webrtc::VideoTrack*> videoOutgoing() const;
 
 	crl::time getDurationMs() const;
 	float64 getWaitingSoundPeakValue() const;
@@ -212,7 +212,7 @@ private:
 	Type _type = Type::Outgoing;
 	rpl::variable<State> _state = State::Starting;
 	rpl::variable<RemoteAudioState> _remoteAudioState = RemoteAudioState::Active;
-	rpl::variable<webrtc::VideoState> _remoteVideoState;
+	rpl::variable<Webrtc::VideoState> _remoteVideoState;
 	FinishType _finishAfterRequestingCall = FinishType::None;
 	bool _answerAfterDhConfigReceived = false;
 	rpl::variable<int> _signalBarCount = kSignalBarStarting;
@@ -236,8 +236,8 @@ private:
 
 	std::unique_ptr<tgcalls::Instance> _instance;
 	std::shared_ptr<tgcalls::VideoCaptureInterface> _videoCapture;
-	const std::unique_ptr<webrtc::VideoTrack> _videoIncoming;
-	const std::unique_ptr<webrtc::VideoTrack> _videoOutgoing;
+	const std::unique_ptr<Webrtc::VideoTrack> _videoIncoming;
+	const std::unique_ptr<Webrtc::VideoTrack> _videoOutgoing;
 
 	std::unique_ptr<Media::Audio::Track> _waitingTrack;
 
diff --git a/Telegram/SourceFiles/calls/calls_panel.cpp b/Telegram/SourceFiles/calls/calls_panel.cpp
index 413da3a521..97de27e4c8 100644
--- a/Telegram/SourceFiles/calls/calls_panel.cpp
+++ b/Telegram/SourceFiles/calls/calls_panel.cpp
@@ -312,9 +312,9 @@ void Panel::initControls() {
 	_camera->setClickedCallback([=] {
 		if (_call) {
 			_call->videoOutgoing()->setState(
-				(_call->videoOutgoing()->state() == webrtc::VideoState::Active)
-				? webrtc::VideoState::Inactive
-				: webrtc::VideoState::Active);
+				(_call->videoOutgoing()->state() == Webrtc::VideoState::Active)
+				? Webrtc::VideoState::Inactive
+				: Webrtc::VideoState::Active);
 		}
 	});
 
@@ -399,8 +399,8 @@ void Panel::reinitWithCall(Call *call) {
 	}, _callLifetime);
 
 	_call->videoOutgoing()->stateValue(
-	) | rpl::start_with_next([=](webrtc::VideoState state) {
-		_camera->setIconOverride((state == webrtc::VideoState::Active)
+	) | rpl::start_with_next([=](Webrtc::VideoState state) {
+		_camera->setIconOverride((state == Webrtc::VideoState::Active)
 			? nullptr
 			: &st::callNoCameraIcon);
 	}, _callLifetime);
@@ -547,7 +547,7 @@ void Panel::initGeometry() {
 
 void Panel::refreshOutgoingPreviewInBody(State state) {
 	const auto inBody = (state != State::Established)
-		&& (_call->videoOutgoing()->state() != webrtc::VideoState::Inactive)
+		&& (_call->videoOutgoing()->state() != Webrtc::VideoState::Inactive)
 		&& !_call->videoOutgoing()->frameSize().isEmpty();
 	if (_outgoingPreviewInBody == inBody) {
 		return;
@@ -722,7 +722,7 @@ void Panel::paintEvent(QPaintEvent *e) {
 	}
 
 	const auto incomingFrame = _call
-		? _call->videoIncoming()->frame(webrtc::FrameRequest())
+		? _call->videoIncoming()->frame(Webrtc::FrameRequest())
 		: QImage();
 	if (!incomingFrame.isNull()) {
 		const auto to = rect().marginsRemoved(_padding);
@@ -877,7 +877,7 @@ void Panel::stateChanged(State state) {
 }
 
 bool Panel::hasActiveVideo() const {
-	const auto inactive = webrtc::VideoState::Inactive;
+	const auto inactive = Webrtc::VideoState::Inactive;
 	return (_call->videoIncoming()->state() != inactive)
 		|| (_call->videoOutgoing()->state() != inactive);
 }
diff --git a/Telegram/SourceFiles/calls/calls_video_bubble.cpp b/Telegram/SourceFiles/calls/calls_video_bubble.cpp
index 35d50e5598..7cba5529f1 100644
--- a/Telegram/SourceFiles/calls/calls_video_bubble.cpp
+++ b/Telegram/SourceFiles/calls/calls_video_bubble.cpp
@@ -18,10 +18,10 @@ namespace Calls {
 
 VideoBubble::VideoBubble(
 	not_null<QWidget*> parent,
-	not_null<webrtc::VideoTrack*> track)
+	not_null<Webrtc::VideoTrack*> track)
 : _content(parent)
 , _track(track)
-, _state(webrtc::VideoState::Inactive) {
+, _state(Webrtc::VideoState::Inactive) {
 	setup();
 }
 
@@ -35,7 +35,7 @@ void VideoBubble::setup() {
 	}, lifetime());
 
 	_track->stateValue(
-	) | rpl::start_with_next([=](webrtc::VideoState state) {
+	) | rpl::start_with_next([=](Webrtc::VideoState state) {
 		setState(state);
 	}, lifetime());
 
@@ -137,7 +137,7 @@ void VideoBubble::prepareFrame() {
 		* cIntRetinaFactor();
 
 	// Should we check 'original' and 'size' aspect ratios?..
-	const auto request = webrtc::FrameRequest{
+	const auto request = Webrtc::FrameRequest{
 		.resize = size,
 		.outer = size,
 	};
@@ -165,13 +165,13 @@ void VideoBubble::prepareFrame() {
 		QRect(QPoint(), size));
 }
 
-void VideoBubble::setState(webrtc::VideoState state) {
-	if (state == webrtc::VideoState::Paused) {
+void VideoBubble::setState(Webrtc::VideoState state) {
+	if (state == Webrtc::VideoState::Paused) {
 		using namespace Images;
 		static constexpr auto kRadius = 24;
 		_pausedFrame = Images::BlurLargeImage(_track->frame({}), kRadius);
 		if (_pausedFrame.isNull()) {
-			state = webrtc::VideoState::Inactive;
+			state = Webrtc::VideoState::Inactive;
 		}
 	}
 	_state = state;
@@ -240,7 +240,7 @@ void VideoBubble::setInnerSize(QSize size) {
 
 void VideoBubble::updateVisibility() {
 	const auto size = _track->frameSize();
-	const auto visible = (_state != webrtc::VideoState::Inactive)
+	const auto visible = (_state != Webrtc::VideoState::Inactive)
 		&& !size.isEmpty();
 	if (visible) {
 		updateSizeToFrame(size);
diff --git a/Telegram/SourceFiles/calls/calls_video_bubble.h b/Telegram/SourceFiles/calls/calls_video_bubble.h
index a8682c6e8b..17a7d18b62 100644
--- a/Telegram/SourceFiles/calls/calls_video_bubble.h
+++ b/Telegram/SourceFiles/calls/calls_video_bubble.h
@@ -9,10 +9,10 @@ https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
 
 #include "ui/rp_widget.h"
 
-namespace webrtc {
+namespace Webrtc {
 class VideoTrack;
 enum class VideoState;
-} // namespace webrtc
+} // namespace Webrtc
 
 namespace Calls {
 
@@ -20,7 +20,7 @@ class VideoBubble final {
 public:
 	VideoBubble(
 		not_null<QWidget*> parent,
-		not_null<webrtc::VideoTrack*> track);
+		not_null<Webrtc::VideoTrack*> track);
 
 	enum class DragMode {
 		None,
@@ -39,7 +39,7 @@ public:
 private:
 	void setup();
 	void paint();
-	void setState(webrtc::VideoState state);
+	void setState(Webrtc::VideoState state);
 	void applyDragMode(DragMode mode);
 	void applyBoundingRect(QRect rect);
 	void applySizeConstraints(QSize min, QSize max);
@@ -49,8 +49,8 @@ private:
 	void prepareFrame();
 
 	Ui::RpWidget _content;
-	const not_null<webrtc::VideoTrack*> _track;
-	webrtc::VideoState _state = webrtc::VideoState();
+	const not_null<Webrtc::VideoTrack*> _track;
+	Webrtc::VideoState _state = Webrtc::VideoState();
 	QImage _frame, _pausedFrame;
 	QSize _min, _max, _size, _lastDraggableSize, _lastFrameSize;
 	QRect _boundingRect;
diff --git a/Telegram/ThirdParty/tgcalls b/Telegram/ThirdParty/tgcalls
index 9de3547711..82172d1b7b 160000
--- a/Telegram/ThirdParty/tgcalls
+++ b/Telegram/ThirdParty/tgcalls
@@ -1 +1 @@
-Subproject commit 9de3547711c5bb7056c101fc47e1b912d411c2f4
+Subproject commit 82172d1b7bef85c4dba984dc0d9e681b23dbc1a7
diff --git a/Telegram/cmake/lib_tgcalls.cmake b/Telegram/cmake/lib_tgcalls.cmake
index ca0187ca60..bb7609765e 100644
--- a/Telegram/cmake/lib_tgcalls.cmake
+++ b/Telegram/cmake/lib_tgcalls.cmake
@@ -5,7 +5,13 @@
 # https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
 
 add_library(lib_tgcalls STATIC)
+
+if (LINUX)
+init_target(lib_tgcalls) # All C++20 on Linux, because otherwise ODR violation.
+else()
 init_target(lib_tgcalls cxx_std_14)
+endif()
+
 add_library(tdesktop::lib_tgcalls ALIAS lib_tgcalls)
 
 set(tgcalls_dir ${third_party_loc}/tgcalls)
@@ -43,6 +49,8 @@ PRIVATE
     platform/PlatformInterface.h
 
     # Teleram Desktop
+    platform/tdesktop/DesktopInterface.cpp
+    platform/tdesktop/DesktopInterface.h
     platform/tdesktop/VideoCapturerInterfaceImpl.cpp
     platform/tdesktop/VideoCapturerInterfaceImpl.h
     platform/tdesktop/VideoCapturerTrackSource.cpp
@@ -50,10 +58,6 @@ PRIVATE
     platform/tdesktop/VideoCameraCapturer.cpp
     platform/tdesktop/VideoCameraCapturer.h
 
-    # Windows
-    platform/windows/WindowsInterface.cpp
-    platform/windows/WindowsInterface.h
-
     # iOS / macOS
     platform/darwin/DarwinInterface.h
     platform/darwin/DarwinInterface.mm
@@ -84,8 +88,6 @@ PRIVATE
     platform/darwin/VideoMetalViewMac.h
     platform/darwin/VideoMetalViewMac.mm
 
-    # Linux
-
     # POSIX
 
     reference/InstanceImplReference.cpp
@@ -121,6 +123,8 @@ elseif (APPLE)
         platform/darwin/VideoMetalView.mm
         platform/darwin/VideoMetalViewMac.h
         platform/darwin/VideoMetalViewMac.mm
+        platform/tdesktop/DesktopInterface.cpp
+        platform/tdesktop/DesktopInterface.h
         platform/tdesktop/VideoCapturerTrackSource.cpp
         platform/tdesktop/VideoCapturerTrackSource.h
         platform/tdesktop/VideoCapturerInterfaceImpl.cpp
@@ -146,7 +150,13 @@ PRIVATE
 )
 
 add_library(lib_tgcalls_legacy STATIC)
-init_target(lib_tgcalls_legacy cxx_std_14)
+
+if (LINUX)
+    init_target(lib_tgcalls_legacy) # All C++20 on Linux, because otherwise ODR violation.
+else()
+    init_target(lib_tgcalls_legacy cxx_std_14)
+endif()
+
 add_library(tdesktop::lib_tgcalls_legacy ALIAS lib_tgcalls_legacy)
 
 nice_target_sources(lib_tgcalls_legacy ${tgcalls_loc}
diff --git a/Telegram/cmake/lib_tgvoip.cmake b/Telegram/cmake/lib_tgvoip.cmake
index a5b1d04958..7f8cdb0bcc 100644
--- a/Telegram/cmake/lib_tgvoip.cmake
+++ b/Telegram/cmake/lib_tgvoip.cmake
@@ -14,7 +14,13 @@ if (TDESKTOP_USE_PACKAGED_TGVOIP AND NOT DESKTOP_APP_USE_PACKAGED_LAZY)
     target_link_libraries(lib_tgvoip INTERFACE PkgConfig::TGVOIP)
 else()
     add_library(lib_tgvoip STATIC)
-    init_target(lib_tgvoip cxx_std_14)
+
+    if (LINUX)
+        init_target(lib_tgvoip) # All C++20 on Linux, because otherwise ODR violation.
+    else()
+        init_target(lib_tgvoip cxx_std_14)
+    endif()
+
     add_library(tdesktop::lib_tgvoip ALIAS lib_tgvoip)
 
     set(tgvoip_loc ${third_party_loc}/libtgvoip)
diff --git a/Telegram/lib_webrtc b/Telegram/lib_webrtc
index a8e19691c5..183b7ae329 160000
--- a/Telegram/lib_webrtc
+++ b/Telegram/lib_webrtc
@@ -1 +1 @@
-Subproject commit a8e19691c5d653310f7ac5f75d69e45b34771fa4
+Subproject commit 183b7ae329d57663131c3019a384516f5f51898f