diff --git a/client/src/contexts/WebRTCProvider.tsx b/client/src/contexts/WebRTCProvider.tsx
new file mode 100644
index 0000000..a18d24d
--- /dev/null
+++ b/client/src/contexts/WebRTCProvider.tsx
@@ -0,0 +1,167 @@
+/*
+ * Copyright (C) 2022 Savoir-faire Linux Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU Affero General Public License as
+ * published by the Free Software Foundation; either version 3 of the
+ * License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU Affero General Public License for more details.
+ *
+ * You should have received a copy of the GNU Affero General Public
+ * License along with this program.  If not, see
+ * <https://www.gnu.org/licenses/>.
+ */
+
+import React, { createContext, useCallback, useRef } from 'react';
+import { connect, Socket } from 'socket.io-client';
+
+import { WithChildren } from '../utils/utils';
+
+/*
+ * TODO: This socket is temporary, it will be replaced by the real socket
+ * for communication with webrtc
+ * */
+const socket = connect('http://192.168.0.12:8080', { transports: ['websocket'] });
+
+interface IWebRTCContext {
+  localVideoRef: React.RefObject<HTMLVideoElement> | null;
+  remoteVideoRef: React.RefObject<HTMLVideoElement> | null;
+  createWebRTCConnection: () => void;
+  sendWebRTCOffer: () => void;
+  sendWebRTCAnswer: (remoteSdp: RTCSessionDescriptionInit) => void;
+  handleWebRTCAnswer: (remoteSdp: RTCSessionDescriptionInit) => void;
+  addIceCandidate: (candidate: RTCIceCandidateInit) => void;
+  socket: Socket;
+}
+
+const DefaultWebRTCContext: IWebRTCContext = {
+  localVideoRef: null,
+  remoteVideoRef: null,
+  createWebRTCConnection: () => {},
+  sendWebRTCOffer: () => {},
+  sendWebRTCAnswer: () => {},
+  handleWebRTCAnswer: () => {},
+  addIceCandidate: () => {},
+  socket: socket,
+};
+
+export const WebRTCContext = createContext<IWebRTCContext>(DefaultWebRTCContext);
+
+export default ({ children }: WithChildren) => {
+  const localVideoRef = useRef<HTMLVideoElement>(null);
+  const remoteVideoRef = useRef<HTMLVideoElement>(null);
+  const webRTCConnectionRef = useRef<RTCPeerConnection>();
+
+  const createWebRTCConnection = useCallback(async () => {
+    //TODO use SFL iceServers
+    const iceConfig = { iceServers: [{ urls: 'stun:stun.l.google.com:19302' }] };
+    webRTCConnectionRef.current = new RTCPeerConnection(iceConfig);
+    const localStream = await navigator.mediaDevices.getUserMedia({
+      video: true,
+      audio: true,
+    });
+
+    if (localVideoRef.current) {
+      localVideoRef.current.srcObject = localStream;
+    }
+
+    localStream.getTracks().forEach((track) => {
+      if (webRTCConnectionRef.current) {
+        webRTCConnectionRef.current.addTrack(track, localStream);
+      }
+    });
+    webRTCConnectionRef.current.addEventListener('icecandidate', (event) => {
+      if (event.candidate && socket) {
+        console.log('webRTCConnection : onicecandidate');
+        socket.emit('candidate', event.candidate);
+      }
+    });
+    webRTCConnectionRef.current.addEventListener('track', async (event) => {
+      if (remoteVideoRef.current) {
+        remoteVideoRef.current.srcObject = event.streams[0];
+        console.log('webRTCConnection : add remotetrack success');
+      }
+    });
+  }, [webRTCConnectionRef, localVideoRef, remoteVideoRef]);
+
+  const sendWebRTCOffer = useCallback(async () => {
+    try {
+      if (webRTCConnectionRef.current && socket) {
+        const sdp = await webRTCConnectionRef.current.createOffer({
+          offerToReceiveAudio: true,
+          offerToReceiveVideo: true,
+        });
+        await webRTCConnectionRef.current.setLocalDescription(new RTCSessionDescription(sdp));
+        socket.emit('offer', sdp);
+      }
+    } catch (e) {
+      console.error(e);
+    }
+  }, [webRTCConnectionRef]);
+
+  const sendWebRTCAnswer = useCallback(
+    async (remoteSdp: RTCSessionDescriptionInit) => {
+      try {
+        if (webRTCConnectionRef.current && socket && remoteSdp) {
+          await webRTCConnectionRef.current.setRemoteDescription(new RTCSessionDescription(remoteSdp));
+          const mySdp = await webRTCConnectionRef.current.createAnswer({
+            offerToReceiveAudio: true,
+            offerToReceiveVideo: true,
+          });
+          await webRTCConnectionRef.current.setLocalDescription(new RTCSessionDescription(mySdp));
+          socket.emit('answer', mySdp);
+        }
+      } catch (e) {
+        console.error(e);
+      }
+    },
+    [webRTCConnectionRef]
+  );
+
+  const handleWebRTCAnswer = useCallback(
+    async (remoteSdp: RTCSessionDescriptionInit) => {
+      try {
+        if (webRTCConnectionRef.current && remoteSdp) {
+          await webRTCConnectionRef.current.setRemoteDescription(new RTCSessionDescription(remoteSdp));
+        }
+      } catch (e) {
+        console.error(e);
+      }
+    },
+    [webRTCConnectionRef]
+  );
+
+  const addIceCandidate = useCallback(
+    async (candidate: RTCIceCandidateInit) => {
+      try {
+        if (webRTCConnectionRef.current) {
+          await webRTCConnectionRef.current.addIceCandidate(new RTCIceCandidate(candidate));
+        }
+      } catch (e) {
+        console.error(e);
+      }
+    },
+    [webRTCConnectionRef]
+  );
+
+  return (
+    <WebRTCContext.Provider
+      value={{
+        localVideoRef,
+        remoteVideoRef,
+        createWebRTCConnection,
+        sendWebRTCOffer,
+        sendWebRTCAnswer,
+        handleWebRTCAnswer,
+        addIceCandidate,
+        socket,
+      }}
+    >
+      {children}
+    </WebRTCContext.Provider>
+  );
+};