diff --git a/client/src/contexts/CallProvider.tsx b/client/src/contexts/CallProvider.tsx
index 39aad07..5464228 100644
--- a/client/src/contexts/CallProvider.tsx
+++ b/client/src/contexts/CallProvider.tsx
@@ -24,7 +24,7 @@
 import CallPermissionDenied from '../pages/CallPermissionDenied';
 import { CallRouteParams } from '../router';
 import { callTimeoutMs } from '../utils/constants';
-import { SetState, WithChildren } from '../utils/utils';
+import { AsyncSetState, SetState, WithChildren } from '../utils/utils';
 import { useConversationContext } from './ConversationProvider';
 import { MediaDevicesInfo, MediaInputKind, WebRtcContext } from './WebRtcProvider';
 import { IWebSocketContext, WebSocketContext } from './WebSocketProvider';
@@ -40,6 +40,12 @@
   PermissionsDenied,
 }
 
+export enum VideoStatus {
+  Off,
+  Camera,
+  ScreenShare,
+}
+
 type MediaDeviceIdState = {
   id: string | undefined;
   setId: (id: string | undefined) => void | Promise<void>;
@@ -52,8 +58,8 @@
 
   isAudioOn: boolean;
   setIsAudioOn: SetState<boolean>;
-  isVideoOn: boolean;
-  setIsVideoOn: SetState<boolean>;
+  videoStatus: VideoStatus;
+  updateVideoStatus: AsyncSetState<VideoStatus>;
   isChatShown: boolean;
   setIsChatShown: SetState<boolean>;
   isFullscreen: boolean;
@@ -89,8 +95,8 @@
 
   isAudioOn: false,
   setIsAudioOn: () => {},
-  isVideoOn: false,
-  setIsVideoOn: () => {},
+  videoStatus: VideoStatus.Off,
+  updateVideoStatus: () => Promise.reject(),
   isChatShown: false,
   setIsChatShown: () => {},
   isFullscreen: false,
@@ -122,8 +128,15 @@
   webSocket: IWebSocketContext;
 }) => {
   const { state: routeState } = useUrlParams<CallRouteParams>();
-  const { localStream, sendWebRtcOffer, iceConnectionState, closeConnection, getMediaDevices, updateLocalStream } =
-    useContext(WebRtcContext);
+  const {
+    localStream,
+    updateScreenShare,
+    sendWebRtcOffer,
+    iceConnectionState,
+    closeConnection,
+    getMediaDevices,
+    updateLocalStream,
+  } = useContext(WebRtcContext);
   const { conversationId, conversation } = useConversationContext();
   const navigate = useNavigate();
 
@@ -133,7 +146,7 @@
   const [videoDeviceId, setVideoDeviceId] = useState<string>();
 
   const [isAudioOn, setIsAudioOn] = useState(false);
-  const [isVideoOn, setIsVideoOn] = useState(false);
+  const [videoStatus, setVideoStatus] = useState(VideoStatus.Off);
   const [isChatShown, setIsChatShown] = useState(false);
   const [isFullscreen, setIsFullscreen] = useState(false);
   const [callStatus, setCallStatus] = useState(routeState?.callStatus);
@@ -187,14 +200,35 @@
   useEffect(() => {
     if (localStream) {
       for (const track of localStream.getVideoTracks()) {
-        track.enabled = isVideoOn;
+        track.enabled = videoStatus === VideoStatus.Camera;
         const deviceId = track.getSettings().deviceId;
         if (deviceId) {
           setVideoDeviceId(deviceId);
         }
       }
     }
-  }, [isVideoOn, localStream]);
+  }, [videoStatus, localStream]);
+
+  const updateVideoStatus = useCallback(
+    async (newStatus: ((prevState: VideoStatus) => VideoStatus) | VideoStatus) => {
+      if (typeof newStatus === 'function') {
+        newStatus = newStatus(videoStatus);
+      }
+
+      const stream = await updateScreenShare(newStatus === VideoStatus.ScreenShare);
+      if (stream) {
+        for (const track of stream.getTracks()) {
+          track.addEventListener('ended', () => {
+            console.warn('Browser ended screen sharing');
+            updateVideoStatus(VideoStatus.Off);
+          });
+        }
+      }
+
+      setVideoStatus(newStatus);
+    },
+    [videoStatus, updateScreenShare]
+  );
 
   useEffect(() => {
     const onFullscreenChange = () => {
@@ -220,7 +254,7 @@
           };
 
           setCallStatus(CallStatus.Ringing);
-          setIsVideoOn(withVideoOn);
+          setVideoStatus(withVideoOn ? VideoStatus.Camera : VideoStatus.Off);
           console.info('Sending CallBegin', callBegin);
           webSocket.send(WebSocketMessageType.CallBegin, callBegin);
         })
@@ -241,7 +275,7 @@
             conversationId,
           };
 
-          setIsVideoOn(withVideoOn);
+          setVideoStatus(withVideoOn ? VideoStatus.Camera : VideoStatus.Off);
           setCallStatus(CallStatus.Connecting);
           console.info('Sending CallAccept', callAccept);
           webSocket.send(WebSocketMessageType.CallAccept, callAccept);
@@ -324,7 +358,7 @@
       console.info('ICE connection disconnected or failed, ending call');
       endCall();
     }
-  }, [iceConnectionState, callStatus, isVideoOn, endCall]);
+  }, [iceConnectionState, callStatus, videoStatus, endCall]);
 
   useEffect(() => {
     const checkStatusTimeout = () => {
@@ -386,8 +420,8 @@
         currentMediaDeviceIds,
         isAudioOn,
         setIsAudioOn,
-        isVideoOn,
-        setIsVideoOn,
+        videoStatus,
+        updateVideoStatus,
         isChatShown,
         setIsChatShown,
         isFullscreen,
diff --git a/client/src/contexts/WebRtcProvider.tsx b/client/src/contexts/WebRtcProvider.tsx
index 59999b9..1ce3ff6 100644
--- a/client/src/contexts/WebRtcProvider.tsx
+++ b/client/src/contexts/WebRtcProvider.tsx
@@ -33,9 +33,11 @@
   iceConnectionState: RTCIceConnectionState | undefined;
 
   localStream: MediaStream | undefined;
+  screenShareLocalStream: MediaStream | undefined;
   remoteStreams: readonly MediaStream[] | undefined;
   getMediaDevices: () => Promise<MediaDevicesInfo>;
   updateLocalStream: (mediaDeviceIds?: MediaInputIds) => Promise<void>;
+  updateScreenShare: (active: boolean) => Promise<MediaStream | undefined>;
 
   sendWebRtcOffer: () => Promise<void>;
   closeConnection: () => void;
@@ -44,9 +46,11 @@
 const defaultWebRtcContext: IWebRtcContext = {
   iceConnectionState: undefined,
   localStream: undefined,
+  screenShareLocalStream: undefined,
   remoteStreams: undefined,
   getMediaDevices: async () => Promise.reject(),
   updateLocalStream: async () => Promise.reject(),
+  updateScreenShare: async () => Promise.reject(),
   sendWebRtcOffer: async () => Promise.reject(),
   closeConnection: () => {},
 };
@@ -101,6 +105,7 @@
 }) => {
   const { conversation, conversationId } = useConversationContext();
   const [localStream, setLocalStream] = useState<MediaStream>();
+  const [screenShareLocalStream, setScreenShareLocalStream] = useState<MediaStream>();
   const [remoteStreams, setRemoteStreams] = useState<readonly MediaStream[]>();
   const [iceConnectionState, setIceConnectionState] = useState<RTCIceConnectionState | undefined>();
 
@@ -183,14 +188,37 @@
     [getMediaDevices]
   );
 
+  const updateScreenShare = useCallback(
+    async (isOn: boolean) => {
+      if (isOn) {
+        const stream = await navigator.mediaDevices.getDisplayMedia({
+          video: true,
+          audio: false,
+        });
+
+        setScreenShareLocalStream(stream);
+        return stream;
+      } else {
+        if (screenShareLocalStream) {
+          for (const track of screenShareLocalStream.getTracks()) {
+            track.stop();
+          }
+        }
+
+        setScreenShareLocalStream(undefined);
+      }
+    },
+    [screenShareLocalStream]
+  );
+
   useEffect(() => {
-    if (!localStream || !webRtcConnection) {
+    if ((!localStream && !screenShareLocalStream) || !webRtcConnection) {
       return;
     }
 
-    const updateTracks = async (kind: 'audio' | 'video') => {
+    const updateTracks = async (stream: MediaStream, kind: 'audio' | 'video') => {
       const senders = kind === 'audio' ? audioRtcRtpSenders : videoRtcRtpSenders;
-      const tracks = kind === 'audio' ? localStream.getAudioTracks() : localStream.getVideoTracks();
+      const tracks = kind === 'audio' ? stream.getAudioTracks() : stream.getVideoTracks();
       if (senders) {
         const promises: Promise<void>[] = [];
         for (let i = 0; i < senders.length; i++) {
@@ -210,7 +238,7 @@
       // TODO: Currently, we do not support adding new devices. To enable this feature, we would need to implement
       //       the "Perfect negotiation" pattern to renegotiate after `addTrack`.
       //       https://blog.mozilla.org/webrtc/perfect-negotiation-in-webrtc/
-      const newSenders = tracks.map((track) => webRtcConnection.addTrack(track, localStream));
+      const newSenders = tracks.map((track) => webRtcConnection.addTrack(track, stream));
       if (kind === 'audio') {
         setAudioRtcRtpSenders(newSenders);
       } else {
@@ -218,9 +246,15 @@
       }
     };
 
-    updateTracks('audio');
-    updateTracks('video');
-  }, [localStream, webRtcConnection, audioRtcRtpSenders, videoRtcRtpSenders]);
+    if (localStream) {
+      updateTracks(localStream, 'audio');
+      updateTracks(localStream, 'video');
+    }
+
+    if (screenShareLocalStream) {
+      updateTracks(screenShareLocalStream, 'video');
+    }
+  }, [localStream, screenShareLocalStream, webRtcConnection, audioRtcRtpSenders, videoRtcRtpSenders]);
 
   const sendWebRtcOffer = useCallback(async () => {
     const sdp = await webRtcConnection.createOffer({
@@ -375,24 +409,35 @@
   }, [webRtcConnection]);
 
   const closeConnection = useCallback(() => {
-    const localTracks = localStream?.getTracks();
-    if (localTracks) {
-      for (const track of localTracks) {
-        track.stop();
+    const stopStream = (stream: MediaStream) => {
+      const localTracks = stream.getTracks();
+      if (localTracks) {
+        for (const track of localTracks) {
+          track.stop();
+        }
       }
+    };
+
+    if (localStream) {
+      stopStream(localStream);
+    }
+    if (screenShareLocalStream) {
+      stopStream(screenShareLocalStream);
     }
 
     webRtcConnection.close();
-  }, [webRtcConnection, localStream]);
+  }, [webRtcConnection, localStream, screenShareLocalStream]);
 
   return (
     <WebRtcContext.Provider
       value={{
         iceConnectionState,
         localStream,
+        screenShareLocalStream,
         remoteStreams,
         getMediaDevices,
         updateLocalStream,
+        updateScreenShare,
         sendWebRtcOffer,
         closeConnection,
       }}