diff --git a/.gitignore b/.gitignore
new file mode 100644
index 0000000..07e6e47
--- /dev/null
+++ b/.gitignore
@@ -0,0 +1 @@
+/node_modules
diff --git a/PeerConnection.js b/PeerConnection.js
new file mode 100644
index 0000000..3727dd1
--- /dev/null
+++ b/PeerConnection.js
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+/*  MIT License: https://webrtc-experiment.appspot.com/licence/ 
+ 2013, Muaz Khan<muazkh>--[github.com/muaz-khan]
+
+ Demo & Documentation: http://bit.ly/RTCPeerConnection-Documentation */
+window.moz = !! navigator.mozGetUserMedia;
+var PeerConnection = function (options) {
+    var PeerConnection = window.mozRTCPeerConnection || window.webkitRTCPeerConnection,
+        SessionDescription = window.mozRTCSessionDescription || window.RTCSessionDescription,
+        IceCandidate = window.mozRTCIceCandidate || window.RTCIceCandidate;
+
+    // See https://gist.github.com/zziuni/3741933 for a list of public STUN servers
+    var iceServers = {
+        iceServers: [{ url: 'stun:stunserver.org' }]
+    };
+
+    var optional = {
+        optional: []
+    };
+    if (!moz) {
+        // See http://www.webrtc.org/interop under "Constraints / configurations issues."
+        optional.optional = [{
+                DtlsSrtpKeyAgreement: true
+            }
+        ];
+    }
+
+    var peerConnection = new PeerConnection(iceServers, optional);
+    peerConnection.onicecandidate = function(event) {
+        if (!event.candidate) return;
+        options.onicecandidate(event.candidate);
+    }
+    peerConnection.onaddstream = function(event) {
+        console.log('------------onaddstream');
+        options.onaddstream(event.stream);
+    }
+
+    var constraints = options.constraints || {
+        optional: [],
+        mandatory: {
+            OfferToReceiveAudio: true,
+            OfferToReceiveVideo: true
+        }
+    };
+    if (moz) constraints.mandatory.MozDontOfferDataChannel = true;
+
+    return {
+        createOffer: function (callback) {
+            peerConnection.createOffer(function (sessionDescription) {
+                peerConnection.setLocalDescription(sessionDescription);
+                callback(sessionDescription);
+            }, null, constraints);
+        },
+
+        createAnswer: function (offerSDP, callback) {
+            peerConnection.setRemoteDescription(new SessionDescription(offerSDP));
+            peerConnection.createAnswer(function (sessionDescription) {
+                peerConnection.setLocalDescription(sessionDescription);
+                callback(sessionDescription);
+            }, null, constraints);
+        },
+
+        setRemoteDescription: function (sdp) {
+            console.log('--------adding answer sdp:');
+            console.log(sdp.sdp);
+
+            sdp = new SessionDescription(sdp);
+            peerConnection.setRemoteDescription(sdp);
+        },
+
+        addICECandidate: function (candidate) {
+            console.log("addICE: got candidate: " + candidate.candidate);
+            peerConnection.addIceCandidate(new IceCandidate({
+                        sdpMLineIndex: candidate.sdpMLineIndex,
+                        candidate: candidate.candidate
+                    }));
+        },
+
+        addStream: function(stream) {
+            console.log("stream provided, attaching...");
+            peerConnection.addStream(stream);
+        }
+    };
+};
diff --git a/README b/README
new file mode 100644
index 0000000..5ff06e3
--- /dev/null
+++ b/README
@@ -0,0 +1,55 @@
+This is a *minimal* demo of WebRTC video calling between two browsers.
+Because the signalling layer isn't standardized, you'll need to run a
+small node.js server to pass signalling messages between the two
+browsers.  The media connections are point-to-point, though, once the
+signalling has happened no server is required.
+
+This demo is based on Silvia Pfeiffer's WebRTC via Web Sockets demo,
+and Muaz Khan's WebRTC-Experiment websocket-over-nodejs demo.
+Silvia's demo became obsolete (it uses ROAP signalling while JSEP is
+now the standard) and Muaz's demo was more complex than I liked which
+made it difficult to figure out what the WebRTC API's were doing.
+Both of their code was very helpful to me, though!
+
+Silvia's Blog post:
+http://blog.gingertech.net/2012/06/04/video-conferencing-in-html5-webrtc-via-web-sockets/
+
+Muaz's WebRTC Experiments web site:
+https://www.webrtc-experiment.com/
+
+Muaz's WebSocket over Node.js demo:
+https://github.com/muaz-khan/WebRTC-Experiment/tree/master/websocket-over-nodejs
+
+Installation:
+
+Install node.js from http://nodejs.org
+
+Install Node's websocket package:
+$ npm install websocket
+
+Install the node-static package to serve static files:
+$ npm install node-static
+
+Usage:
+
+Run the Node server:
+$ node websocket-server.js
+
+Point two browsers at http://your-server-address:1337/.  You'll be
+prompted to allow the use of your camera and microphone.
+
+One of the browsers clicks "Call" and the call should be set up.
+
+To close the call just reload the page.  Like I said: this is a
+minimal demo.
+
+Debugging the Node Server:
+  install https://github.com/node-inspector/node-inspector
+  run the node app in debug mode (will stop at the first line)
+  $ node --debug-brk websocket-server.js
+  then in another terminal
+  $ node node_modules/node-inspector/bin/inspector.js
+   - or -
+  $ node_modules/.bin/node-inspector
+  open chrome to http://127.0.0.1:8080/debug?port=5858
+  step once to go to the first line of the program
diff --git a/call.js b/call.js
new file mode 100644
index 0000000..54fc641
--- /dev/null
+++ b/call.js
@@ -0,0 +1,78 @@
+/* MIT License: https://webrtc-experiment.appspot.com/licence/ */
+
+var call = function (config) {
+    var
+    peerConnection = PeerConnection(makePeerConfig()),
+    webSocket = new WebSocket('ws://' + document.location.host + '/');
+
+    // configure the signalling WebSocket
+    webSocket.onmessage = function (event) {
+        console.log("received a message: " + event.data);
+        onIncomingMessage(JSON.parse(event.data));
+    };
+    webSocket.push = webSocket.send;
+    webSocket.send = function (data) {
+        webSocket.push(JSON.stringify(data));
+    };
+
+    function onIncomingMessage(response) {
+        // the other client has sent me an offer SDP
+        if (response.offerSDP) {
+            console.log("received offerSDP " + response.offerSDP + ", will answer");
+            peerConnection.addStream(config.localStream);
+            peerConnection.createAnswer(response.offerSDP, function (sdp) {
+                console.log("sending answer SDP");
+                webSocket.send({
+                    answerSDP: sdp
+                });
+            });
+        }
+
+        // the other client has sent me an answer SDP
+        if (response.answerSDP) {
+            peerConnection.setRemoteDescription(response.answerSDP);
+        }
+
+        // the other client has sent me an ICE candidate
+        if (response.candidate) {
+            console.log("got a candidate message, passing to RTCPeerConnection");
+            peerConnection.addICECandidate({
+                sdpMLineIndex: response.candidate.sdpMLineIndex,
+                candidate: response.candidate.candidate
+            });
+        }
+    }
+
+    // PeerConnection.js's options structure
+    function makePeerConfig() {
+        return {
+            onicecandidate: function (candidate) {
+                console.log("onICE");
+                webSocket.send({
+                    candidate: {
+                        sdpMLineIndex: candidate.sdpMLineIndex,
+                        candidate: candidate.candidate
+                    }
+                });
+            },
+            onaddstream: function (stream) {
+                console.log("onRemoteStream");
+                config.video['src'] = URL.createObjectURL(stream);
+            }
+        };
+    }
+
+    return {
+        initiateCall: function(localStream) {
+            // attach the stream to the peer connection
+            peerConnection.addStream(localStream);
+            // create the offer SDP and send it when it's ready
+            peerConnection.createOffer(function (sdp) {
+                console.log("sending offer SDP");
+                webSocket.send({
+                    offerSDP: sdp
+                })
+            });
+        }
+    };
+};
diff --git a/index.html b/index.html
new file mode 100644
index 0000000..3200c6b
--- /dev/null
+++ b/index.html
@@ -0,0 +1,51 @@
+<!DOCTYPE html>
+<html>
+  <head>
+    <title>Example WebRTC Point-to-point Video</title>
+    <style>
+      #video {
+      position: relative;
+      }
+      #local-video {
+      left: 0;
+      position: absolute;
+      top: 0;
+      z-index: 1;
+      }
+    </style>
+  </head>
+  <body>
+    <h1>Example WebRTC Point-to-point Video</h1>
+    <table class="visible">
+      <tr><td><button id="call" onClick="callPeer.initiateCall(config.localStream)">Call</button></td></tr>
+    </table>
+    <div id="video">
+      <video id="remote-video" autoplay="true" controls="true" width="480" height="360"></video>
+      <video id="local-video" autoplay="true" controls="true" muted="true" width="160" height="120"></video>
+    </div>
+    <script src="PeerConnection.js" type="text/javascript"></script>
+    <script src="call.js" type="text/javascript"></script>
+    <script type="text/javascript">
+// temporary kludge to paper over browser differences
+navigator.getUserMedia = navigator.webkitGetUserMedia || navigator.mozGetUserMedia;
+
+var config = {
+    video: document.getElementById('remote-video'),
+};
+
+navigator.getUserMedia({
+        audio: true,
+        video: true
+    },
+    function (localStream) {
+        video = document.getElementById('local-video');
+        video['src'] = URL.createObjectURL(localStream);
+        config.localStream = localStream;
+    },
+    function (e) { console.error(e); }
+);
+
+var callPeer = call(config);
+    </script>
+  </body>
+</html>
diff --git a/websocket-server.js b/websocket-server.js
new file mode 100644
index 0000000..6c3be01
--- /dev/null
+++ b/websocket-server.js
@@ -0,0 +1,69 @@
+// adapted from:
+// http://blog.gingertech.net/2012/06/04/video-conferencing-in-html5-webrtc-via-web-sockets/
+
+var WebSocketServer = require('websocket').server;
+var http = require('http');
+var static = require('node-static');
+var clients = [];
+
+var staticServer = new static.Server('.');
+
+// set up node-static to serve static files in this directory,
+// i.e., http://localhost:1337/index.html
+var server = http.createServer(function(request, response) {
+    request.addListener('end', function() {
+        staticServer.serve(request, response);
+    }).resume();
+});
+server.listen(1337, function() {
+  console.log((new Date()) + " Server is listening on port 1337");
+});
+
+// create the server
+wsServer = new WebSocketServer({
+    httpServer: server
+});
+
+function sendCallback(err) {
+    if (err) console.error((new Date()) + "send() error: " + err);
+}
+
+// This callback function is called every time someone
+// tries to connect to the WebSocket server
+wsServer.on('request', function(request) {
+    console.log((new Date()) + ' Connection from origin ' + request.origin + '.');
+    var connection = request.accept(null, request.origin);
+    console.log((new Date()) + ' Connection remoteAddress ' + connection.remoteAddress);
+    clients.push(connection);
+
+    // This is the most important callback for us, we'll handle
+    // all messages from users here.
+    connection.on('message', function(message) {
+        if (message.type === 'utf8') {
+            // process WebSocket message
+            console.log((new Date()) + ' Received Message ' + message.utf8Data);
+            // broadcast message to all connected clients
+            clients.forEach(function (outputConnection) {
+                if (outputConnection != connection) {
+                  outputConnection.send(message.utf8Data, sendCallback);
+                }
+            });
+        }
+    });
+
+    connection.on('close', function(connection) {
+        // close user connection
+        console.log((new Date()) + " Peer disconnected.");
+        for (var i = 0; i < clients.length; i++) {
+            if (!clients[i].connected)
+                clients.remove(i);
+        }
+    });
+});
+
+// Array Remove - By John Resig (MIT Licensed)
+Array.prototype.remove = function(from, to) {
+  var rest = this.slice((to || from) + 1 || this.length);
+  this.length = from < 0 ? this.length + from : from;
+  return this.push.apply(this, rest);
+};