diff --git a/api/BUILD.gn b/api/BUILD.gn
index dc83db9..375cf9d 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -368,21 +368,6 @@
   ]
 }
 
-rtc_source_set("ortc_api") {
-  visibility = [ "*" ]
-  sources = [
-    "ortc/packet_transport_interface.h",
-    "ortc/rtp_transport_interface.h",
-    "ortc/srtp_transport_interface.h",
-  ]
-
-  deps = [
-    ":libjingle_peerconnection_api",
-    ":rtp_headers",
-    "//third_party/abseil-cpp/absl/types:optional",
-  ]
-}
-
 rtc_source_set("rtc_stats_api") {
   visibility = [ "*" ]
   cflags = []
diff --git a/api/ortc/packet_transport_interface.h b/api/ortc/packet_transport_interface.h
deleted file mode 100644
index 78e280a..0000000
--- a/api/ortc/packet_transport_interface.h
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- *  Copyright 2017 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.
- */
-
-#ifndef API_ORTC_PACKET_TRANSPORT_INTERFACE_H_
-#define API_ORTC_PACKET_TRANSPORT_INTERFACE_H_
-
-namespace rtc {
-
-class PacketTransportInternal;
-
-}  // namespace rtc
-
-namespace webrtc {
-
-// Base class for different packet-based transports.
-class PacketTransportInterface {
- public:
-  virtual ~PacketTransportInterface() {}
-
- protected:
-  // Only for internal use. Returns a pointer to an internal interface, for use
-  // by the implementation.
-  virtual rtc::PacketTransportInternal* GetInternal() = 0;
-
-  // Classes that can use this internal interface.
-  friend class RtpTransportControllerAdapter;
-};
-
-}  // namespace webrtc
-
-#endif  // API_ORTC_PACKET_TRANSPORT_INTERFACE_H_
diff --git a/api/ortc/rtp_transport_interface.h b/api/ortc/rtp_transport_interface.h
deleted file mode 100644
index 2e16885..0000000
--- a/api/ortc/rtp_transport_interface.h
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- *  Copyright 2017 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.
- */
-
-#ifndef API_ORTC_RTP_TRANSPORT_INTERFACE_H_
-#define API_ORTC_RTP_TRANSPORT_INTERFACE_H_
-
-#include <string>
-
-#include "absl/types/optional.h"
-#include "api/ortc/packet_transport_interface.h"
-#include "api/rtc_error.h"
-#include "api/rtp_headers.h"
-#include "api/rtp_parameters.h"
-
-namespace webrtc {
-
-struct RtpTransportParameters final {
-  RtcpParameters rtcp;
-
-  bool operator==(const RtpTransportParameters& o) const {
-    return rtcp == o.rtcp;
-  }
-  bool operator!=(const RtpTransportParameters& o) const {
-    return !(*this == o);
-  }
-};
-
-// Base class for different types of RTP transports that can be created by an
-// OrtcFactory. Used by RtpSenders/RtpReceivers.
-//
-// This is not present in the standard ORTC API, but exists here for a few
-// reasons. Firstly, it allows different types of RTP transports to be used:
-// DTLS-SRTP (which is required for the web), but also SDES-SRTP and
-// unencrypted RTP. It also simplifies the handling of RTCP muxing, and
-// provides a better API point for it.
-//
-// Note that Edge's implementation of ORTC provides a similar API point, called
-// RTCSrtpSdesTransport:
-// https://msdn.microsoft.com/en-us/library/mt502527(v=vs.85).aspx
-class RtpTransportInterface {
- public:
-  virtual ~RtpTransportInterface() {}
-
-  // Returns packet transport that's used to send RTP packets.
-  virtual PacketTransportInterface* GetRtpPacketTransport() const = 0;
-
-  // Returns separate packet transport that's used to send RTCP packets. If
-  // RTCP multiplexing is being used, returns null.
-  virtual PacketTransportInterface* GetRtcpPacketTransport() const = 0;
-
-  // Set/get RTP/RTCP transport params. Can be used to enable RTCP muxing or
-  // reduced-size RTCP if initially not enabled.
-  //
-  // Changing |mux| from "true" to "false" is not allowed, and changing the
-  // CNAME is currently unsupported.
-  // RTP keep-alive settings need to be set before before an RtpSender has
-  // started sending, altering the payload type or timeout interval after this
-  // point is not supported. The parameters must also match across all RTP
-  // transports for a given RTP transport controller.
-  virtual RTCError SetParameters(const RtpTransportParameters& parameters) = 0;
-  // Returns last set or constructed-with parameters. If |cname| was empty in
-  // construction, the generated CNAME will be present in the returned
-  // parameters (see above).
-  virtual RtpTransportParameters GetParameters() const = 0;
-
- protected:
-  // Classes that can use this internal interface.
-  friend class OrtcFactory;
-  friend class OrtcRtpSenderAdapter;
-  friend class OrtcRtpReceiverAdapter;
-  friend class RtpTransportControllerAdapter;
-};
-
-}  // namespace webrtc
-
-#endif  // API_ORTC_RTP_TRANSPORT_INTERFACE_H_
diff --git a/api/ortc/srtp_transport_interface.h b/api/ortc/srtp_transport_interface.h
deleted file mode 100644
index 65ef1ef..0000000
--- a/api/ortc/srtp_transport_interface.h
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- *  Copyright 2017 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.
- */
-
-#ifndef API_ORTC_SRTP_TRANSPORT_INTERFACE_H_
-#define API_ORTC_SRTP_TRANSPORT_INTERFACE_H_
-
-#include "api/crypto_params.h"
-#include "api/ortc/rtp_transport_interface.h"
-#include "api/rtc_error.h"
-
-namespace webrtc {
-
-// The subclass of the RtpTransport which uses SRTP. The keying information
-// is explicitly passed in from the application.
-//
-// If using SDP and SDES (RFC4568) for signaling, then after applying the
-// answer, the negotiated keying information from the offer and answer would be
-// set and the SRTP would be active.
-//
-// Note that Edge's implementation of ORTC provides a similar API point, called
-// RTCSrtpSdesTransport:
-// https://msdn.microsoft.com/en-us/library/mt502527(v=vs.85).aspx
-class SrtpTransportInterface : public RtpTransportInterface {
- public:
-  virtual ~SrtpTransportInterface() {}
-
-  // There are some limitations of the current implementation:
-  //  1. Send and receive keys must use the same crypto suite.
-  //  2. The keys can't be changed after initially set.
-  //  3. The keys must be set before creating a sender/receiver using the SRTP
-  //     transport.
-  // Set the SRTP keying material for sending RTP and RTCP.
-  virtual RTCError SetSrtpSendKey(const cricket::CryptoParams& params) = 0;
-
-  // Set the SRTP keying material for receiving RTP and RTCP.
-  virtual RTCError SetSrtpReceiveKey(const cricket::CryptoParams& params) = 0;
-};
-
-}  // namespace webrtc
-
-#endif  // API_ORTC_SRTP_TRANSPORT_INTERFACE_H_
diff --git a/p2p/BUILD.gn b/p2p/BUILD.gn
index e404896..c389e81 100644
--- a/p2p/BUILD.gn
+++ b/p2p/BUILD.gn
@@ -87,7 +87,6 @@
 
   deps = [
     "../api:libjingle_peerconnection_api",
-    "../api:ortc_api",
     "../api:scoped_refptr",
     "../api/transport:enums",
     "../logging:ice_log",
@@ -159,7 +158,6 @@
       ":p2p_server_utils",
       ":rtc_p2p",
       "../api:libjingle_peerconnection_api",
-      "../api:ortc_api",
       "../rtc_base",
       "../rtc_base:gunit_helpers",
       "../rtc_base:rtc_base_approved",
@@ -205,7 +203,6 @@
       ":p2p_test_utils",
       ":rtc_p2p",
       "../api:libjingle_peerconnection_api",
-      "../api:ortc_api",
       "../api:scoped_refptr",
       "../api/units:time_delta",
       "../rtc_base",
diff --git a/p2p/base/fake_packet_transport.h b/p2p/base/fake_packet_transport.h
index 6f79aad..f59aa39 100644
--- a/p2p/base/fake_packet_transport.h
+++ b/p2p/base/fake_packet_transport.h
@@ -13,7 +13,6 @@
 
 #include <string>
 
-#include "api/ortc/packet_transport_interface.h"
 #include "p2p/base/packet_transport_internal.h"
 #include "rtc_base/async_invoker.h"
 #include "rtc_base/copy_on_write_buffer.h"
diff --git a/p2p/base/packet_transport_internal.cc b/p2p/base/packet_transport_internal.cc
index 8fda899..0904cb2 100644
--- a/p2p/base/packet_transport_internal.cc
+++ b/p2p/base/packet_transport_internal.cc
@@ -16,10 +16,6 @@
 
 PacketTransportInternal::~PacketTransportInternal() = default;
 
-PacketTransportInternal* PacketTransportInternal::GetInternal() {
-  return this;
-}
-
 bool PacketTransportInternal::GetOption(rtc::Socket::Option opt, int* value) {
   return false;
 }
diff --git a/p2p/base/packet_transport_internal.h b/p2p/base/packet_transport_internal.h
index 24723e9..a532183 100644
--- a/p2p/base/packet_transport_internal.h
+++ b/p2p/base/packet_transport_internal.h
@@ -15,8 +15,6 @@
 #include <vector>
 
 #include "absl/types/optional.h"
-// This is included for PacketOptions.
-#include "api/ortc/packet_transport_interface.h"
 #include "p2p/base/port.h"
 #include "rtc_base/async_packet_socket.h"
 #include "rtc_base/network_route.h"
@@ -28,9 +26,7 @@
 struct PacketOptions;
 struct SentPacket;
 
-class RTC_EXPORT PacketTransportInternal
-    : public virtual webrtc::PacketTransportInterface,
-      public sigslot::has_slots<> {
+class RTC_EXPORT PacketTransportInternal : public sigslot::has_slots<> {
  public:
   virtual const std::string& transport_name() const = 0;
 
@@ -102,8 +98,6 @@
  protected:
   PacketTransportInternal();
   ~PacketTransportInternal() override;
-
-  PacketTransportInternal* GetInternal() override;
 };
 
 }  // namespace rtc
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 62a8186..120d3f8 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -77,7 +77,6 @@
     "../api:audio_options_api",
     "../api:call_api",
     "../api:libjingle_peerconnection_api",
-    "../api:ortc_api",
     "../api:rtp_headers",
     "../api:scoped_refptr",
     "../api/video:builtin_video_bitrate_allocator_factory",
diff --git a/pc/rtp_transport.cc b/pc/rtp_transport.cc
index e6129b5..b7d3a9d 100644
--- a/pc/rtp_transport.cc
+++ b/pc/rtp_transport.cc
@@ -164,26 +164,6 @@
   return true;
 }
 
-RTCError RtpTransport::SetParameters(const RtpTransportParameters& parameters) {
-  if (parameters_.rtcp.mux && !parameters.rtcp.mux) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
-                         "Disabling RTCP muxing is not allowed.");
-  }
-
-  RtpTransportParameters new_parameters = parameters;
-
-  if (new_parameters.rtcp.cname.empty()) {
-    new_parameters.rtcp.cname = parameters_.rtcp.cname;
-  }
-
-  parameters_ = new_parameters;
-  return RTCError::OK();
-}
-
-RtpTransportParameters RtpTransport::GetParameters() const {
-  return parameters_;
-}
-
 void RtpTransport::DemuxPacket(rtc::CopyOnWriteBuffer packet,
                                int64_t packet_time_us) {
   webrtc::RtpPacketReceived parsed_packet(&header_extension_map_);
diff --git a/pc/rtp_transport.h b/pc/rtp_transport.h
index 4573d3c..269a61a 100644
--- a/pc/rtp_transport.h
+++ b/pc/rtp_transport.h
@@ -49,17 +49,6 @@
   }
   void SetRtcpPacketTransport(rtc::PacketTransportInternal* rtcp) override;
 
-  PacketTransportInterface* GetRtpPacketTransport() const override {
-    return rtp_packet_transport_;
-  }
-  PacketTransportInterface* GetRtcpPacketTransport() const override {
-    return rtcp_packet_transport_;
-  }
-
-  // TODO(zstein): Use these RtcpParameters for configuration elsewhere.
-  RTCError SetParameters(const RtpTransportParameters& parameters) override;
-  RtpTransportParameters GetParameters() const override;
-
   bool IsReadyToSend() const override { return ready_to_send_; }
 
   bool IsWritable(bool rtcp) const override;
@@ -119,18 +108,6 @@
 
   bool IsTransportWritable();
 
-  // SRTP specific methods.
-  // TODO(zhihuang): Improve the inheritance model so that the RtpTransport
-  // doesn't need to implement SRTP specfic methods.
-  RTCError SetSrtpSendKey(const cricket::CryptoParams& params) override {
-    RTC_NOTREACHED();
-    return RTCError::OK();
-  }
-  RTCError SetSrtpReceiveKey(const cricket::CryptoParams& params) override {
-    RTC_NOTREACHED();
-    return RTCError::OK();
-  }
-
   bool rtcp_mux_enabled_;
 
   rtc::PacketTransportInternal* rtp_packet_transport_ = nullptr;
@@ -140,7 +117,6 @@
   bool rtp_ready_to_send_ = false;
   bool rtcp_ready_to_send_ = false;
 
-  RtpTransportParameters parameters_;
   RtpDemuxer rtp_demuxer_;
 
   // Used for identifying the MID for RtpDemuxer.
diff --git a/pc/rtp_transport_internal.h b/pc/rtp_transport_internal.h
index f8a5f2b..fce1f9b 100644
--- a/pc/rtp_transport_internal.h
+++ b/pc/rtp_transport_internal.h
@@ -13,7 +13,6 @@
 
 #include <string>
 
-#include "api/ortc/srtp_transport_interface.h"
 #include "call/rtp_demuxer.h"
 #include "p2p/base/ice_transport_internal.h"
 #include "pc/session_description.h"
@@ -32,9 +31,10 @@
 // it is not accessible to API consumers but is accessible to internal classes
 // in order to send and receive RTP and RTCP packets belonging to a single RTP
 // session. Additional convenience and configuration methods are also provided.
-class RtpTransportInternal : public SrtpTransportInterface,
-                             public sigslot::has_slots<> {
+class RtpTransportInternal : public sigslot::has_slots<> {
  public:
+  virtual ~RtpTransportInternal() = default;
+
   virtual void SetRtcpMuxEnabled(bool enable) = 0;
 
   // TODO(zstein): Remove PacketTransport setters. Clients should pass these
diff --git a/pc/rtp_transport_unittest.cc b/pc/rtp_transport_unittest.cc
index 54f182d..4248ba7 100644
--- a/pc/rtp_transport_unittest.cc
+++ b/pc/rtp_transport_unittest.cc
@@ -31,27 +31,6 @@
 constexpr int kLastPacketId = 100;
 constexpr int kTransportOverheadPerPacket = 28;  // Ipv4(20) + UDP(8).
 
-TEST(RtpTransportTest, SetRtcpParametersCantDisableRtcpMux) {
-  RtpTransport transport(kMuxDisabled);
-  RtpTransportParameters params;
-  transport.SetParameters(params);
-  params.rtcp.mux = false;
-  EXPECT_FALSE(transport.SetParameters(params).ok());
-}
-
-TEST(RtpTransportTest, SetRtcpParametersEmptyCnameUsesExisting) {
-  static const char kName[] = "name";
-  RtpTransport transport(kMuxDisabled);
-  RtpTransportParameters params_with_name;
-  params_with_name.rtcp.cname = kName;
-  transport.SetParameters(params_with_name);
-  EXPECT_EQ(transport.GetParameters().rtcp.cname, kName);
-
-  RtpTransportParameters params_without_name;
-  transport.SetParameters(params_without_name);
-  EXPECT_EQ(transport.GetParameters().rtcp.cname, kName);
-}
-
 class SignalObserver : public sigslot::has_slots<> {
  public:
   explicit SignalObserver(RtpTransport* transport) {
diff --git a/pc/srtp_transport.h b/pc/srtp_transport.h
index e725733..ed92379 100644
--- a/pc/srtp_transport.h
+++ b/pc/srtp_transport.h
@@ -40,8 +40,8 @@
   virtual ~SrtpTransport() = default;
 
   // SrtpTransportInterface specific implementation.
-  RTCError SetSrtpSendKey(const cricket::CryptoParams& params) override;
-  RTCError SetSrtpReceiveKey(const cricket::CryptoParams& params) override;
+  virtual RTCError SetSrtpSendKey(const cricket::CryptoParams& params);
+  virtual RTCError SetSrtpReceiveKey(const cricket::CryptoParams& params);
 
   bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
                      const rtc::PacketOptions& options,