diff --git a/samples/js/demos/app.yaml b/samples/js/demos/app.yaml
deleted file mode 100644
index 918e4b7f50..0000000000
--- a/samples/js/demos/app.yaml
+++ /dev/null
@@ -1,30 +0,0 @@
-application: webrtc-demos
-version: 1
-runtime: python27
-api_version: 1
-threadsafe: yes
-
-handlers:
-- url: /favicon\.ico
-  static_files: favicon.ico
-  upload: favicon\.ico
-
-- url: /html
-  static_dir: html
-  secure: always
-
-- url: /images
-  static_dir: images
-  secure: always
-
-- url: /js
-  static_dir: js
-  secure: always
-
-- url: .*
-  script: main.app
-  secure: always
-
-libraries:
-- name: webapp2
-  version: "2.5.1"
diff --git a/samples/js/demos/html/dtmf1.html b/samples/js/demos/html/dtmf1.html
new file mode 100644
index 0000000000..c9e37f97af
--- /dev/null
+++ b/samples/js/demos/html/dtmf1.html
@@ -0,0 +1,187 @@
+<html>
+<head>
+<title>PeerConnection DTMF Demo 1</title>
+<!-- Load the polyfill to switch-hit between Chrome and Firefox -->
+<script src="/js/adapter.js"></script>
+<style>
+button {
+  font: 18px sans-serif;
+  padding: 8px;
+}
+#left { position: absolute; left: 20; top: 0; width: 50%; }
+#right { position: absolute; right: 0; top: 0; width: 50%; }
+</style>
+</head>
+<body onload="onload()">
+<div id="left">
+<audio id="audio1" autoplay="autoplay" muted="true"></audio>
+<h3>Send Dtmf Tones</h3>
+<div id="dialingPad"></div>
+<br><br>
+duration:
+<input type="text" id="dtmf-tones-duration" size="10" value="100"/>
+<br><br>
+tone-gap:
+<input type="text" id="dtmf-tones-gap" size="10" value="50"/>
+<br><br>
+tones:
+<input type="text" id="dtmf-tones" size="10" value="123,abc"/>
+<button id="sendTones" 
+  onclick="sendTone(document.getElementById('dtmf-tones').value)">Send tones
+  </button>
+<br><br>
+<button id="callBtn" onclick="call()">Call</button>
+<button id="hangBtn" onclick="hangup()">Hang Up</button>
+<br><br>
+</div>
+<div id="right">
+<audio id="audio2" autoplay="autoplay"></audio>
+<h3>Sent Tones</h3>
+<textarea id="dtmfTonesSent" rows="12" cols="40" disabled="true">
+</textarea><br>
+</div>
+<script>
+callBtn.disabled = false;
+hangBtn.disabled = true;
+sendTones.disabled = true;
+var pc1 = null,pc2 = null;
+var localstream = null;
+var dtmfSender = null;
+var sdpConstraints = {'mandatory': {
+                      'OfferToReceiveAudio':true, 
+                      'OfferToReceiveVideo':false }};
+
+function gotStream(stream){
+  trace("Received local stream");
+  // Call the polyfill wrapper to attach the media stream to this element.
+  localstream = stream;
+  audioTracks = localstream.getAudioTracks();
+  if (audioTracks.length > 0)
+    trace('Using Audio device: ' + audioTracks[0].label);
+  pc1.addStream(localstream);
+  trace("Adding Local Stream to peer connection");
+  
+  pc1.createOffer(gotDescription1);
+}
+  
+function call() {
+  trace("Starting call");
+  var servers = null;
+  var pc_constraints = {"optional": []};
+  pc1 = new RTCPeerConnection(servers,pc_constraints);
+  trace("Created local peer connection object pc1");
+  pc1.onicecandidate = iceCallback1; 
+  pc2 = new RTCPeerConnection(servers,pc_constraints);
+  trace("Created remote peer connection object pc2");
+  pc2.onicecandidate = iceCallback2;
+  pc2.onaddstream = gotRemoteStream; 
+  
+  trace("Requesting local stream");
+  // Call into getUserMedia via the polyfill (adapter.js).
+  getUserMedia({audio:true, video:false},
+                gotStream, function() {});
+
+  callBtn.disabled = true;
+  hangBtn.disabled = false;
+  sendTones.disabled = false;
+}
+
+function gotDescription1(desc){
+  pc1.setLocalDescription(desc);
+  trace("Offer from pc1 \n" + desc.sdp);
+  pc2.setRemoteDescription(desc);
+  // Since the "remote" side has no media stream we need
+  // to pass in the right constraints in order for it to
+  // accept the incoming offer of audio.
+  pc2.createAnswer(gotDescription2, null, sdpConstraints);
+}
+
+function gotDescription2(desc){
+  pc2.setLocalDescription(desc);
+  trace("Answer from pc2 \n" + desc.sdp);
+  pc1.setRemoteDescription(desc);
+}
+
+function hangup() {
+  trace("Ending call");
+  pc1.close(); 
+  pc2.close();
+  pc1 = null;
+  pc2 = null;
+  localstream = null;
+  dtmfSender = null;
+  callBtn.disabled = false;
+  hangBtn.disabled = true;
+  sendTones.disabled = true;
+  document.getElementById("dtmfTonesSent").value = "Dtmf de-activated\n";
+}
+
+function gotRemoteStream(e){
+  audio2.src = webkitURL.createObjectURL(e.stream);
+  trace("Received remote stream");
+  enableDtmfSender();
+}
+
+function iceCallback1(event){
+  if (event.candidate) {
+    pc2.addIceCandidate(new RTCIceCandidate(event.candidate));
+    trace("Local ICE candidate: \n" + event.candidate.candidate);
+  }
+}
+      
+function iceCallback2(event){
+  if (event.candidate) {
+    pc1.addIceCandidate(new RTCIceCandidate(event.candidate));
+    trace("Remote ICE candidate: \n " + event.candidate.candidate);
+  }
+}
+
+function enableDtmfSender(){
+  document.getElementById("dtmfTonesSent").value = "Dtmf activated\n";
+  if (localstream != null) {
+    var local_audio_track = localstream.getAudioTracks()[0];
+    dtmfSender = pc1.createDTMFSender(local_audio_track);
+    trace("Created DTMF Sender\n");
+    dtmfSender.ontonechange = dtmfOnToneChange;
+  }
+  else {
+    trace("No Local Stream to create DTMF Sender\n");
+  }
+}
+
+function dtmfOnToneChange(tone){
+  if (tone) {
+    trace("Sent Dtmf tone: \t" + tone.tone);
+    document.getElementById("dtmfTonesSent").value += tone.tone + '\t';
+  }
+}
+
+function sendTone(tones){
+  if (dtmfSender) {
+    duration = document.getElementById("dtmf-tones-duration").value;
+    gap = document.getElementById("dtmf-tones-gap").value;
+    dtmfSender.insertDTMF(tones, duration, gap);
+  }
+}
+
+function createDialingPad() {
+  var tones = '1234567890*#ABCD';
+  var dialingPad = document.getElementById('dialingPad');
+  for (var i = 0; i < tones.length; ++i) { 
+    var tone = tones.charAt(i);
+    dialingPad.innerHTML += '<button id="' + 
+      tone + '" onclick="sendTone(\'' + tone +
+      '\')" style="height:40px; width: 30px">' + tone + '</button>';
+    if ((i + 1) % 4 == 0) {
+      dialingPad.innerHTML += '<br>';
+    } 
+  }
+}
+
+function onload() {
+  createDialingPad();
+}
+
+</script>
+</body>
+</html>
diff --git a/samples/js/demos/html/pc1-audio.html b/samples/js/demos/html/pc1-audio.html
new file mode 100644
index 0000000000..ad56329ad8
--- /dev/null
+++ b/samples/js/demos/html/pc1-audio.html
@@ -0,0 +1,110 @@
+<!DOCTYPE html>
+<html>
+<head>
+<title>PeerConnection Audio Only Demo 1</title>
+<!-- Load the polyfill to switch-hit between Chrome and Firefox -->
+<script src="/js/adapter.js"></script>
+<style>
+button {
+  font: 18px sans-serif;
+  padding: 8px;
+}
+</style>
+</head>
+<body>
+<h2>Local-Audio</h2>
+<audio id="audio1" autoplay="autoplay" controls="controls" muted="true"></audio>
+<h2>Remote-Audio</h2>
+<audio id="audio2" autoplay="autoplay" controls="controls"></audio>
+<br><br>
+<button id="btn1" onclick="call()">Call</button>
+<button id="btn2" onclick="hangup()">Hang Up</button>
+<br><br>
+<script>
+btn1.disabled = false;
+btn2.disabled = true;
+var pc1,pc2;
+var localstream;
+var sdpConstraints = {'mandatory': {
+                        'OfferToReceiveAudio':true, 
+                        'OfferToReceiveVideo':false }};
+
+function gotStream(stream){
+  trace("Received local stream");
+  // Call the polyfill wrapper to attach the media stream to this element.
+  localstream = stream;
+  audioTracks = localstream.getAudioTracks();
+  if (audioTracks.length > 0)
+    trace('Using Audio device: ' + audioTracks[0].label);
+  pc1.addStream(localstream);
+  trace("Adding Local Stream to peer connection");
+  
+  pc1.createOffer(gotDescription1);
+}
+  
+function call() {
+  btn1.disabled = true;
+  btn2.disabled = false;
+  trace("Starting call");
+  var servers = null;
+  var pc_constraints = {"optional": []};
+  pc1 = new RTCPeerConnection(servers,pc_constraints);
+  trace("Created local peer connection object pc1");
+  pc1.onicecandidate = iceCallback1; 
+  pc2 = new RTCPeerConnection(servers,pc_constraints);
+  trace("Created remote peer connection object pc2");
+  pc2.onicecandidate = iceCallback2;
+  pc2.onaddstream = gotRemoteStream; 
+  trace("Requesting local stream");
+  // Call into getUserMedia via the polyfill (adapter.js).
+  getUserMedia({audio:true, video:false},
+                gotStream, function() {});
+}
+
+function gotDescription1(desc){
+  pc1.setLocalDescription(desc);
+  trace("Offer from pc1 \n" + desc.sdp);
+  pc2.setRemoteDescription(desc);
+  // Since the "remote" side has no media stream we need
+  // to pass in the right constraints in order for it to
+  // accept the incoming offer of audio.
+  pc2.createAnswer(gotDescription2, null, sdpConstraints);
+}
+
+function gotDescription2(desc){
+  pc2.setLocalDescription(desc);
+  trace("Answer from pc2 \n" + desc.sdp);
+  pc1.setRemoteDescription(desc);
+}
+
+function hangup() {
+  trace("Ending call");
+  pc1.close(); 
+  pc2.close();
+  pc1 = null;
+  pc2 = null;
+  btn2.disabled = true;
+  btn1.disabled = false;
+}
+
+function gotRemoteStream(e){
+  audio2.src = webkitURL.createObjectURL(e.stream);
+  trace("Received remote stream");
+}
+
+function iceCallback1(event){
+  if (event.candidate) {
+    pc2.addIceCandidate(new RTCIceCandidate(event.candidate));
+    trace("Local ICE candidate: \n" + event.candidate.candidate);
+  }
+}
+      
+function iceCallback2(event){
+  if (event.candidate) {
+    pc1.addIceCandidate(new RTCIceCandidate(event.candidate));
+    trace("Remote ICE candidate: \n " + event.candidate.candidate);
+  }
+}
+</script>
+</body>
+</html>
diff --git a/samples/js/demos/index.html b/samples/js/demos/index.html
index 78cba2460e..809148ebe4 100644
--- a/samples/js/demos/index.html
+++ b/samples/js/demos/index.html
@@ -50,6 +50,12 @@
 					<td>
 						Shows how to set up a simple 1:1 audio/video call.</td>
 				</tr>
+        <tr>
+          <td>
+            <a href="html/pc1-audio.html">pc1.html</a></td>
+          <td>
+            Shows how to set up a simple 1:1 audio only call.</td>
+        </tr>
 				<tr>
 					<td>
 						<a href="html/pc1-deprecated.html">pc1-deprecated.html</a></td>
@@ -68,6 +74,24 @@
 					<td>
 						Shows how to pass constraints into the PeerConnection API, and query it for statistics.</td>
 				</tr>
+        <tr>
+          <td>
+            <a href="html/dtmf1.html">dtmf1.html</a></td>
+          <td>
+            Shows how to send DTMF tones using PeerConnection API.</td>
+        </tr>
+         <tr>
+          <td>
+            <a href="html/dc1.html">dc1.html</a></td>
+          <td>
+            Shows how to send Data using PeerConnection API.</td>
+        </tr>
+        <tr>
+          <td>
+            <a href="html/local-audio-rendering.html">local-audio-rendering.html</a></td>
+          <td>
+            Shows usage of a local media stream connected to an HTML5 audio tag.</td>
+        </tr>
 			</tbody>
 		</table>
 		<p>
diff --git a/samples/js/demos/js/demosite/app.yaml b/samples/js/demos/js/demosite/app.yaml
new file mode 100644
index 0000000000..e1f043bcd0
--- /dev/null
+++ b/samples/js/demos/js/demosite/app.yaml
@@ -0,0 +1,14 @@
+application: webrtc-demos
+version: 1
+runtime: python27
+api_version: 1
+threadsafe: yes
+
+handlers:
+- url: .*
+  script: main.app
+  secure: always
+
+libraries:
+- name: webapp2
+  version: "2.5.1"
diff --git a/samples/js/demos/favicon.ico b/samples/js/demos/js/demosite/favicon.ico
similarity index 100%
rename from samples/js/demos/favicon.ico
rename to samples/js/demos/js/demosite/favicon.ico
diff --git a/samples/js/demos/main.py b/samples/js/demos/js/demosite/main.py
similarity index 56%
rename from samples/js/demos/main.py
rename to samples/js/demos/js/demosite/main.py
index 7be9df6a62..a91fda932c 100644
--- a/samples/js/demos/main.py
+++ b/samples/js/demos/js/demosite/main.py
@@ -18,10 +18,20 @@ import webapp2
 import os
 from google.appengine.ext.webapp import template
 
-class MainHandler(webapp2.RequestHandler):
-  def get(self):
-    path = os.path.join(os.path.dirname(__file__), 'index.html')
-    self.response.out.write(template.render(path, {}))
+class PageHandler(webapp2.RequestHandler):
+ def get(self):
+   base_url = self.request.path
+   if self.request.path == '/':
+     self.redirect("http://webrtc.googlecode.com/svn/trunk/" 
+                    + "samples/js/demos/index.html"
+                    , permanent=True)
+   else:
+     self.redirect("http://webrtc.googlecode.com/svn/trunk/" 
+                    + "samples/js/demos"
+                    + base_url, 
+                    permanent=True)
+
+app = webapp2.WSGIApplication([
+  (r'/*.*', PageHandler),
+ ], debug=True)
 
-app = webapp2.WSGIApplication([('/', MainHandler)],
-                              debug=True)
