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diff --git a/third_party/libwebrtc/webrtc/api/rtpsenderinterface.h b/third_party/libwebrtc/webrtc/api/rtpsenderinterface.h
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+++ b/third_party/libwebrtc/webrtc/api/rtpsenderinterface.h
@@ -0,0 +1,84 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file contains interfaces for RtpSenders
+// http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
+
+#ifndef API_RTPSENDERINTERFACE_H_
+#define API_RTPSENDERINTERFACE_H_
+
+#include <string>
+#include <vector>
+
+#include "api/dtmfsenderinterface.h"
+#include "api/mediastreaminterface.h"
+#include "api/mediatypes.h"
+#include "api/proxy.h"
+#include "api/rtpparameters.h"
+#include "rtc_base/refcount.h"
+#include "rtc_base/scoped_ref_ptr.h"
+
+namespace webrtc {
+
+class RtpSenderInterface : public rtc::RefCountInterface {
+ public:
+  // Returns true if successful in setting the track.
+  // Fails if an audio track is set on a video RtpSender, or vice-versa.
+  virtual bool SetTrack(MediaStreamTrackInterface* track) = 0;
+  virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
+
+  // Returns primary SSRC used by this sender for sending media.
+  // Returns 0 if not yet determined.
+  // TODO(deadbeef): Change to rtc::Optional.
+  // TODO(deadbeef): Remove? With GetParameters this should be redundant.
+  virtual uint32_t ssrc() const = 0;
+
+  // Audio or video sender?
+  virtual cricket::MediaType media_type() const = 0;
+
+  // Not to be confused with "mid", this is a field we can temporarily use
+  // to uniquely identify a receiver until we implement Unified Plan SDP.
+  virtual std::string id() const = 0;
+
+  // Returns a list of streams associated with this sender's track. Although we
+  // only support one track per stream, in theory the API allows for multiple.
+  virtual std::vector<std::string> stream_ids() const = 0;
+
+  virtual RtpParameters GetParameters() const = 0;
+  // Note that only a subset of the parameters can currently be changed. See
+  // rtpparameters.h
+  virtual bool SetParameters(const RtpParameters& parameters) = 0;
+
+  // Returns null for a video sender.
+  virtual rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const = 0;
+
+ protected:
+  virtual ~RtpSenderInterface() {}
+};
+
+// Define proxy for RtpSenderInterface.
+// TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
+// are called on is an implementation detail.
+BEGIN_SIGNALING_PROXY_MAP(RtpSender)
+  PROXY_SIGNALING_THREAD_DESTRUCTOR()
+  PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
+  PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
+  PROXY_CONSTMETHOD0(uint32_t, ssrc)
+  PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
+  PROXY_CONSTMETHOD0(std::string, id)
+  PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
+  PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
+  PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
+  PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtmfSenderInterface>, GetDtmfSender);
+END_PROXY_MAP()
+
+}  // namespace webrtc
+
+#endif  // API_RTPSENDERINTERFACE_H_