diff --git a/webrtc/api/datachannelinterface.h b/webrtc/api/datachannelinterface.h
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+/*
+ * libjingle
+ * Copyright 2012 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+// This file contains interfaces for DataChannels
+// http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcdatachannel
+
+#ifndef WEBRTC_API_DATACHANNELINTERFACE_H_
+#define WEBRTC_API_DATACHANNELINTERFACE_H_
+
+#include <string>
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/refcount.h"
+
+
+namespace webrtc {
+
+struct DataChannelInit {
+  DataChannelInit()
+      : reliable(false),
+        ordered(true),
+        maxRetransmitTime(-1),
+        maxRetransmits(-1),
+        negotiated(false),
+        id(-1) {
+  }
+
+  bool reliable;           // Deprecated.
+  bool ordered;            // True if ordered delivery is required.
+  int maxRetransmitTime;   // The max period of time in milliseconds in which
+                           // retransmissions will be sent.  After this time, no
+                           // more retransmissions will be sent. -1 if unset.
+  int maxRetransmits;      // The max number of retransmissions. -1 if unset.
+  std::string protocol;    // This is set by the application and opaque to the
+                           // WebRTC implementation.
+  bool negotiated;         // True if the channel has been externally negotiated
+                           // and we do not send an in-band signalling in the
+                           // form of an "open" message.
+  int id;                  // The stream id, or SID, for SCTP data channels. -1
+                           // if unset.
+};
+
+struct DataBuffer {
+  DataBuffer(const rtc::Buffer& data, bool binary)
+      : data(data),
+        binary(binary) {
+  }
+  // For convenience for unit tests.
+  explicit DataBuffer(const std::string& text)
+      : data(text.data(), text.length()),
+        binary(false) {
+  }
+  size_t size() const { return data.size(); }
+
+  rtc::Buffer data;
+  // Indicates if the received data contains UTF-8 or binary data.
+  // Note that the upper layers are left to verify the UTF-8 encoding.
+  // TODO(jiayl): prefer to use an enum instead of a bool.
+  bool binary;
+};
+
+class DataChannelObserver {
+ public:
+  // The data channel state have changed.
+  virtual void OnStateChange() = 0;
+  //  A data buffer was successfully received.
+  virtual void OnMessage(const DataBuffer& buffer) = 0;
+  // The data channel's buffered_amount has changed.
+  virtual void OnBufferedAmountChange(uint64_t previous_amount){};
+
+ protected:
+  virtual ~DataChannelObserver() {}
+};
+
+class DataChannelInterface : public rtc::RefCountInterface {
+ public:
+  // Keep in sync with DataChannel.java:State and
+  // RTCDataChannel.h:RTCDataChannelState.
+  enum DataState {
+    kConnecting,
+    kOpen,  // The DataChannel is ready to send data.
+    kClosing,
+    kClosed
+  };
+
+  static const char* DataStateString(DataState state) {
+    switch (state) {
+      case kConnecting:
+        return "connecting";
+      case kOpen:
+        return "open";
+      case kClosing:
+        return "closing";
+      case kClosed:
+        return "closed";
+    }
+    RTC_CHECK(false) << "Unknown DataChannel state: " << state;
+    return "";
+  }
+
+  virtual void RegisterObserver(DataChannelObserver* observer) = 0;
+  virtual void UnregisterObserver() = 0;
+  // The label attribute represents a label that can be used to distinguish this
+  // DataChannel object from other DataChannel objects.
+  virtual std::string label() const = 0;
+  virtual bool reliable() const = 0;
+
+  // TODO(tommyw): Remove these dummy implementations when all classes have
+  // implemented these APIs. They should all just return the values the
+  // DataChannel was created with.
+  virtual bool ordered() const { return false; }
+  virtual uint16_t maxRetransmitTime() const { return 0; }
+  virtual uint16_t maxRetransmits() const { return 0; }
+  virtual std::string protocol() const { return std::string(); }
+  virtual bool negotiated() const { return false; }
+
+  virtual int id() const = 0;
+  virtual DataState state() const = 0;
+  // The buffered_amount returns the number of bytes of application data
+  // (UTF-8 text and binary data) that have been queued using SendBuffer but
+  // have not yet been transmitted to the network.
+  virtual uint64_t buffered_amount() const = 0;
+  virtual void Close() = 0;
+  // Sends |data| to the remote peer.
+  virtual bool Send(const DataBuffer& buffer) = 0;
+
+ protected:
+  virtual ~DataChannelInterface() {}
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_API_DATACHANNELINTERFACE_H_