diff --git a/webrtc/api/java/jni/peerconnection_jni.cc b/webrtc/api/java/jni/peerconnection_jni.cc
index 09e8db2..6c5ba97 100644
--- a/webrtc/api/java/jni/peerconnection_jni.cc
+++ b/webrtc/api/java/jni/peerconnection_jni.cc
@@ -2031,10 +2031,19 @@
           .release());
 }
 
-static bool JavaEncodingToJsepRtpEncodingParameters(
+static void JavaRtpParametersToJsepRtpParameters(
     JNIEnv* jni,
-    jobject j_encodings,
-    std::vector<webrtc::RtpEncodingParameters>* encodings) {
+    jobject j_parameters,
+    webrtc::RtpParameters* parameters) {
+  RTC_CHECK(parameters != nullptr);
+  jclass parameters_class = jni->FindClass("org/webrtc/RtpParameters");
+  jfieldID encodings_id =
+      GetFieldID(jni, parameters_class, "encodings", "Ljava/util/LinkedList;");
+  jfieldID codecs_id =
+      GetFieldID(jni, parameters_class, "codecs", "Ljava/util/LinkedList;");
+
+  // Convert encodings.
+  jobject j_encodings = GetObjectField(jni, j_parameters, encodings_id);
   const int kBitrateUnlimited = -1;
   jclass j_encoding_parameters_class =
       jni->FindClass("org/webrtc/RtpParameters$Encoding");
@@ -2057,15 +2066,11 @@
     } else {
       encoding.max_bitrate_bps = kBitrateUnlimited;
     }
-    encodings->push_back(encoding);
+    parameters->encodings.push_back(encoding);
   }
-  return true;
-}
 
-static bool JavaCodecToJsepRtpCodecParameters(
-    JNIEnv* jni,
-    jobject j_codecs,
-    std::vector<webrtc::RtpCodecParameters>* codecs) {
+  // Convert codecs.
+  jobject j_codecs = GetObjectField(jni, j_parameters, codecs_id);
   jclass codec_class = jni->FindClass("org/webrtc/RtpParameters$Codec");
   jfieldID payload_type_id = GetFieldID(jni, codec_class, "payloadType", "I");
   jfieldID mime_type_id =
@@ -2080,45 +2085,20 @@
         JavaToStdString(jni, GetStringField(jni, j_codec, mime_type_id));
     codec.clock_rate = GetIntField(jni, j_codec, clock_rate_id);
     codec.channels = GetIntField(jni, j_codec, channels_id);
-    codecs->push_back(codec);
+    parameters->codecs.push_back(codec);
   }
-  return true;
 }
 
-JOW(jboolean, RtpSender_nativeSetParameters)
-(JNIEnv* jni, jclass, jlong j_rtp_sender_pointer, jobject j_parameters) {
-  if (IsNull(jni, j_parameters)) {
-    return false;
-  }
-  jclass parameters_class = jni->FindClass("org/webrtc/RtpParameters");
-  jclass encoding_class = jni->FindClass("org/webrtc/RtpParameters$Encoding");
-  jfieldID encodings_id =
-      GetFieldID(jni, parameters_class, "encodings", "Ljava/util/LinkedList;");
-  jfieldID codecs_id =
-      GetFieldID(jni, parameters_class, "codecs", "Ljava/util/LinkedList;");
-
-  jobject j_encodings = GetObjectField(jni, j_parameters, encodings_id);
-  jobject j_codecs = GetObjectField(jni, j_parameters, codecs_id);
-  webrtc::RtpParameters parameters;
-  JavaEncodingToJsepRtpEncodingParameters(jni, j_encodings,
-                                          &parameters.encodings);
-  JavaCodecToJsepRtpCodecParameters(jni, j_codecs, &parameters.codecs);
-  return reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)
-      ->SetParameters(parameters);
-}
-
-JOW(jobject, RtpSender_nativeGetParameters)
-(JNIEnv* jni, jclass, jlong j_rtp_sender_pointer) {
-  webrtc::RtpParameters parameters =
-      reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)
-          ->GetParameters();
-
+static jobject JsepRtpParametersToJavaRtpParameters(
+    JNIEnv* jni,
+    const webrtc::RtpParameters& parameters) {
   jclass parameters_class = jni->FindClass("org/webrtc/RtpParameters");
   jmethodID parameters_ctor =
       GetMethodID(jni, parameters_class, "<init>", "()V");
   jobject j_parameters = jni->NewObject(parameters_class, parameters_ctor);
   CHECK_EXCEPTION(jni) << "error during NewObject";
 
+  // Add encodings.
   jclass encoding_class = jni->FindClass("org/webrtc/RtpParameters$Encoding");
   jmethodID encoding_ctor = GetMethodID(jni, encoding_class, "<init>", "()V");
   jfieldID encodings_id =
@@ -2153,6 +2133,7 @@
     RTC_CHECK(added);
   }
 
+  // Add codecs.
   jclass codec_class = jni->FindClass("org/webrtc/RtpParameters$Codec");
   jmethodID codec_ctor = GetMethodID(jni, codec_class, "<init>", "()V");
   jfieldID codecs_id =
@@ -2186,6 +2167,25 @@
   return j_parameters;
 }
 
+JOW(jboolean, RtpSender_nativeSetParameters)
+(JNIEnv* jni, jclass, jlong j_rtp_sender_pointer, jobject j_parameters) {
+  if (IsNull(jni, j_parameters)) {
+    return false;
+  }
+  webrtc::RtpParameters parameters;
+  JavaRtpParametersToJsepRtpParameters(jni, j_parameters, &parameters);
+  return reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)
+      ->SetParameters(parameters);
+}
+
+JOW(jobject, RtpSender_nativeGetParameters)
+(JNIEnv* jni, jclass, jlong j_rtp_sender_pointer) {
+  webrtc::RtpParameters parameters =
+      reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)
+          ->GetParameters();
+  return JsepRtpParametersToJavaRtpParameters(jni, parameters);
+}
+
 JOW(jstring, RtpSender_nativeId)(
     JNIEnv* jni, jclass, jlong j_rtp_sender_pointer) {
   return JavaStringFromStdString(
@@ -2206,6 +2206,25 @@
           .release());
 }
 
+JOW(jboolean, RtpReceiver_nativeSetParameters)
+(JNIEnv* jni, jclass, jlong j_rtp_sender_pointer, jobject j_parameters) {
+  if (IsNull(jni, j_parameters)) {
+    return false;
+  }
+  webrtc::RtpParameters parameters;
+  JavaRtpParametersToJsepRtpParameters(jni, j_parameters, &parameters);
+  return reinterpret_cast<RtpReceiverInterface*>(j_rtp_sender_pointer)
+      ->SetParameters(parameters);
+}
+
+JOW(jobject, RtpReceiver_nativeGetParameters)
+(JNIEnv* jni, jclass, jlong j_rtp_sender_pointer) {
+  webrtc::RtpParameters parameters =
+      reinterpret_cast<RtpReceiverInterface*>(j_rtp_sender_pointer)
+          ->GetParameters();
+  return JsepRtpParametersToJavaRtpParameters(jni, parameters);
+}
+
 JOW(jstring, RtpReceiver_nativeId)(
     JNIEnv* jni, jclass, jlong j_rtp_receiver_pointer) {
   return JavaStringFromStdString(
diff --git a/webrtc/api/java/src/org/webrtc/RtpReceiver.java b/webrtc/api/java/src/org/webrtc/RtpReceiver.java
index 04a5d2a..1c4eef3 100644
--- a/webrtc/api/java/src/org/webrtc/RtpReceiver.java
+++ b/webrtc/api/java/src/org/webrtc/RtpReceiver.java
@@ -27,6 +27,14 @@
     return cachedTrack;
   }
 
+  public boolean setParameters(RtpParameters parameters) {
+    return nativeSetParameters(nativeRtpReceiver, parameters);
+  }
+
+  public RtpParameters getParameters() {
+    return nativeGetParameters(nativeRtpReceiver);
+  }
+
   public String id() {
     return nativeId(nativeRtpReceiver);
   }
@@ -40,6 +48,11 @@
   // Will be released in dispose().
   private static native long nativeGetTrack(long nativeRtpReceiver);
 
+  private static native boolean nativeSetParameters(long nativeRtpReceiver,
+                                                    RtpParameters parameters);
+
+  private static native RtpParameters nativeGetParameters(long nativeRtpReceiver);
+
   private static native String nativeId(long nativeRtpReceiver);
 
   private static native void free(long nativeRtpReceiver);
diff --git a/webrtc/api/mediastreamprovider.h b/webrtc/api/mediastreamprovider.h
index 8c866f0..b23e17b 100644
--- a/webrtc/api/mediastreamprovider.h
+++ b/webrtc/api/mediastreamprovider.h
@@ -63,9 +63,14 @@
       uint32_t ssrc,
       std::unique_ptr<webrtc::AudioSinkInterface> sink) = 0;
 
-  virtual RtpParameters GetAudioRtpParameters(uint32_t ssrc) const = 0;
-  virtual bool SetAudioRtpParameters(uint32_t ssrc,
-                                     const RtpParameters& parameters) = 0;
+  virtual RtpParameters GetAudioRtpSendParameters(uint32_t ssrc) const = 0;
+  virtual bool SetAudioRtpSendParameters(uint32_t ssrc,
+                                         const RtpParameters& parameters) = 0;
+
+  virtual RtpParameters GetAudioRtpReceiveParameters(uint32_t ssrc) const = 0;
+  virtual bool SetAudioRtpReceiveParameters(
+      uint32_t ssrc,
+      const RtpParameters& parameters) = 0;
 
  protected:
   virtual ~AudioProviderInterface() {}
@@ -88,9 +93,14 @@
                             bool enable,
                             const cricket::VideoOptions* options) = 0;
 
-  virtual RtpParameters GetVideoRtpParameters(uint32_t ssrc) const = 0;
-  virtual bool SetVideoRtpParameters(uint32_t ssrc,
-                                     const RtpParameters& parameters) = 0;
+  virtual RtpParameters GetVideoRtpSendParameters(uint32_t ssrc) const = 0;
+  virtual bool SetVideoRtpSendParameters(uint32_t ssrc,
+                                         const RtpParameters& parameters) = 0;
+
+  virtual RtpParameters GetVideoRtpReceiveParameters(uint32_t ssrc) const = 0;
+  virtual bool SetVideoRtpReceiveParameters(
+      uint32_t ssrc,
+      const RtpParameters& parameters) = 0;
 
  protected:
   virtual ~VideoProviderInterface() {}
diff --git a/webrtc/api/rtpparameters.h b/webrtc/api/rtpparameters.h
index 729f841..5c79ab4 100644
--- a/webrtc/api/rtpparameters.h
+++ b/webrtc/api/rtpparameters.h
@@ -25,6 +25,9 @@
   bool operator==(const RtpEncodingParameters& o) const {
     return active == o.active && max_bitrate_bps == o.max_bitrate_bps;
   }
+  bool operator!=(const RtpEncodingParameters& o) const {
+    return !(*this == o);
+  }
 };
 
 struct RtpCodecParameters {
@@ -38,6 +41,7 @@
     return payload_type == o.payload_type && mime_type == o.mime_type &&
            clock_rate == o.clock_rate && channels == o.channels;
   }
+  bool operator!=(const RtpCodecParameters& o) const { return !(*this == o); }
 };
 
 struct RtpParameters {
@@ -47,6 +51,7 @@
   bool operator==(const RtpParameters& o) const {
     return encodings == o.encodings && codecs == o.codecs;
   }
+  bool operator!=(const RtpParameters& o) const { return !(*this == o); }
 };
 
 }  // namespace webrtc
diff --git a/webrtc/api/rtpreceiver.cc b/webrtc/api/rtpreceiver.cc
index 38245dc..1b52ce2 100644
--- a/webrtc/api/rtpreceiver.cc
+++ b/webrtc/api/rtpreceiver.cc
@@ -14,6 +14,7 @@
 #include "webrtc/api/audiotrack.h"
 #include "webrtc/api/videosourceproxy.h"
 #include "webrtc/api/videotrack.h"
+#include "webrtc/base/trace_event.h"
 
 namespace webrtc {
 
@@ -66,6 +67,15 @@
   provider_ = nullptr;
 }
 
+RtpParameters AudioRtpReceiver::GetParameters() const {
+  return provider_->GetAudioRtpReceiveParameters(ssrc_);
+}
+
+bool AudioRtpReceiver::SetParameters(const RtpParameters& parameters) {
+  TRACE_EVENT0("webrtc", "AudioRtpReceiver::SetParameters");
+  return provider_->SetAudioRtpReceiveParameters(ssrc_, parameters);
+}
+
 void AudioRtpReceiver::Reconfigure() {
   if (!provider_) {
     return;
@@ -113,4 +123,13 @@
   provider_ = nullptr;
 }
 
+RtpParameters VideoRtpReceiver::GetParameters() const {
+  return provider_->GetVideoRtpReceiveParameters(ssrc_);
+}
+
+bool VideoRtpReceiver::SetParameters(const RtpParameters& parameters) {
+  TRACE_EVENT0("webrtc", "VideoRtpReceiver::SetParameters");
+  return provider_->SetVideoRtpReceiveParameters(ssrc_, parameters);
+}
+
 }  // namespace webrtc
diff --git a/webrtc/api/rtpreceiver.h b/webrtc/api/rtpreceiver.h
index b581857..2e7339d 100644
--- a/webrtc/api/rtpreceiver.h
+++ b/webrtc/api/rtpreceiver.h
@@ -56,6 +56,9 @@
 
   void Stop() override;
 
+  RtpParameters GetParameters() const override;
+  bool SetParameters(const RtpParameters& parameters) override;
+
  private:
   void Reconfigure();
 
@@ -89,6 +92,9 @@
 
   void Stop() override;
 
+  RtpParameters GetParameters() const override;
+  bool SetParameters(const RtpParameters& parameters) override;
+
  private:
   std::string id_;
   uint32_t ssrc_;
diff --git a/webrtc/api/rtpreceiverinterface.h b/webrtc/api/rtpreceiverinterface.h
index c2a579b..ef4f0e1 100644
--- a/webrtc/api/rtpreceiverinterface.h
+++ b/webrtc/api/rtpreceiverinterface.h
@@ -33,6 +33,12 @@
 
   virtual void Stop() = 0;
 
+  // The WebRTC specification only defines RTCRtpParameters in terms of senders,
+  // but this API also applies them to receivers, similar to ORTC:
+  // http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
+  virtual RtpParameters GetParameters() const = 0;
+  virtual bool SetParameters(const RtpParameters& parameters) = 0;
+
  protected:
   virtual ~RtpReceiverInterface() {}
 };
@@ -42,6 +48,8 @@
 PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
 PROXY_CONSTMETHOD0(std::string, id)
 PROXY_METHOD0(void, Stop)
+PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
+PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
 END_SIGNALING_PROXY()
 
 }  // namespace webrtc
diff --git a/webrtc/api/rtpsender.cc b/webrtc/api/rtpsender.cc
index 360b686..5577b9c 100644
--- a/webrtc/api/rtpsender.cc
+++ b/webrtc/api/rtpsender.cc
@@ -208,12 +208,12 @@
 }
 
 RtpParameters AudioRtpSender::GetParameters() const {
-  return provider_->GetAudioRtpParameters(ssrc_);
+  return provider_->GetAudioRtpSendParameters(ssrc_);
 }
 
 bool AudioRtpSender::SetParameters(const RtpParameters& parameters) {
   TRACE_EVENT0("webrtc", "AudioRtpSender::SetParameters");
-  return provider_->SetAudioRtpParameters(ssrc_, parameters);
+  return provider_->SetAudioRtpSendParameters(ssrc_, parameters);
 }
 
 VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
@@ -350,12 +350,12 @@
 }
 
 RtpParameters VideoRtpSender::GetParameters() const {
-  return provider_->GetVideoRtpParameters(ssrc_);
+  return provider_->GetVideoRtpSendParameters(ssrc_);
 }
 
 bool VideoRtpSender::SetParameters(const RtpParameters& parameters) {
   TRACE_EVENT0("webrtc", "VideoRtpSender::SetParameters");
-  return provider_->SetVideoRtpParameters(ssrc_, parameters);
+  return provider_->SetVideoRtpSendParameters(ssrc_, parameters);
 }
 
 }  // namespace webrtc
diff --git a/webrtc/api/rtpsenderreceiver_unittest.cc b/webrtc/api/rtpsenderreceiver_unittest.cc
index f8b968f..3db9d5e 100644
--- a/webrtc/api/rtpsenderreceiver_unittest.cc
+++ b/webrtc/api/rtpsenderreceiver_unittest.cc
@@ -59,8 +59,12 @@
                     const cricket::AudioOptions& options,
                     cricket::AudioSource* source));
   MOCK_METHOD2(SetAudioPlayoutVolume, void(uint32_t ssrc, double volume));
-  MOCK_CONST_METHOD1(GetAudioRtpParameters, RtpParameters(uint32_t ssrc));
-  MOCK_METHOD2(SetAudioRtpParameters,
+  MOCK_CONST_METHOD1(GetAudioRtpSendParameters, RtpParameters(uint32_t ssrc));
+  MOCK_METHOD2(SetAudioRtpSendParameters,
+               bool(uint32_t ssrc, const RtpParameters&));
+  MOCK_CONST_METHOD1(GetAudioRtpReceiveParameters,
+                     RtpParameters(uint32_t ssrc));
+  MOCK_METHOD2(SetAudioRtpReceiveParameters,
                bool(uint32_t ssrc, const RtpParameters&));
 
   void SetRawAudioSink(
@@ -88,8 +92,12 @@
                     bool enable,
                     const cricket::VideoOptions* options));
 
-  MOCK_CONST_METHOD1(GetVideoRtpParameters, RtpParameters(uint32_t ssrc));
-  MOCK_METHOD2(SetVideoRtpParameters,
+  MOCK_CONST_METHOD1(GetVideoRtpSendParameters, RtpParameters(uint32_t ssrc));
+  MOCK_METHOD2(SetVideoRtpSendParameters,
+               bool(uint32_t ssrc, const RtpParameters&));
+  MOCK_CONST_METHOD1(GetVideoRtpReceiveParameters,
+                     RtpParameters(uint32_t ssrc));
+  MOCK_METHOD2(SetVideoRtpReceiveParameters,
                bool(uint32_t ssrc, const RtpParameters&));
 };
 
@@ -504,9 +512,9 @@
 TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParameters) {
   CreateAudioRtpSender();
 
-  EXPECT_CALL(audio_provider_, GetAudioRtpParameters(kAudioSsrc))
+  EXPECT_CALL(audio_provider_, GetAudioRtpSendParameters(kAudioSsrc))
       .WillOnce(Return(RtpParameters()));
-  EXPECT_CALL(audio_provider_, SetAudioRtpParameters(kAudioSsrc, _))
+  EXPECT_CALL(audio_provider_, SetAudioRtpSendParameters(kAudioSsrc, _))
       .WillOnce(Return(true));
   RtpParameters params = audio_rtp_sender_->GetParameters();
   EXPECT_TRUE(audio_rtp_sender_->SetParameters(params));
@@ -517,9 +525,9 @@
 TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) {
   CreateVideoRtpSender();
 
-  EXPECT_CALL(video_provider_, GetVideoRtpParameters(kVideoSsrc))
+  EXPECT_CALL(video_provider_, GetVideoRtpSendParameters(kVideoSsrc))
       .WillOnce(Return(RtpParameters()));
-  EXPECT_CALL(video_provider_, SetVideoRtpParameters(kVideoSsrc, _))
+  EXPECT_CALL(video_provider_, SetVideoRtpSendParameters(kVideoSsrc, _))
       .WillOnce(Return(true));
   RtpParameters params = video_rtp_sender_->GetParameters();
   EXPECT_TRUE(video_rtp_sender_->SetParameters(params));
@@ -527,4 +535,30 @@
   DestroyVideoRtpSender();
 }
 
+TEST_F(RtpSenderReceiverTest, AudioReceiverCanSetParameters) {
+  CreateAudioRtpReceiver();
+
+  EXPECT_CALL(audio_provider_, GetAudioRtpReceiveParameters(kAudioSsrc))
+      .WillOnce(Return(RtpParameters()));
+  EXPECT_CALL(audio_provider_, SetAudioRtpReceiveParameters(kAudioSsrc, _))
+      .WillOnce(Return(true));
+  RtpParameters params = audio_rtp_receiver_->GetParameters();
+  EXPECT_TRUE(audio_rtp_receiver_->SetParameters(params));
+
+  DestroyAudioRtpReceiver();
+}
+
+TEST_F(RtpSenderReceiverTest, VideoReceiverCanSetParameters) {
+  CreateVideoRtpReceiver();
+
+  EXPECT_CALL(video_provider_, GetVideoRtpReceiveParameters(kVideoSsrc))
+      .WillOnce(Return(RtpParameters()));
+  EXPECT_CALL(video_provider_, SetVideoRtpReceiveParameters(kVideoSsrc, _))
+      .WillOnce(Return(true));
+  RtpParameters params = video_rtp_receiver_->GetParameters();
+  EXPECT_TRUE(video_rtp_receiver_->SetParameters(params));
+
+  DestroyVideoRtpReceiver();
+}
+
 }  // namespace webrtc
diff --git a/webrtc/api/webrtcsession.cc b/webrtc/api/webrtcsession.cc
index 2b146c6..96442c8 100644
--- a/webrtc/api/webrtcsession.cc
+++ b/webrtc/api/webrtcsession.cc
@@ -1219,21 +1219,39 @@
   voice_channel_->SetRawAudioSink(ssrc, std::move(sink));
 }
 
-RtpParameters WebRtcSession::GetAudioRtpParameters(uint32_t ssrc) const {
+RtpParameters WebRtcSession::GetAudioRtpSendParameters(uint32_t ssrc) const {
   ASSERT(signaling_thread()->IsCurrent());
   if (voice_channel_) {
-    return voice_channel_->GetRtpParameters(ssrc);
+    return voice_channel_->GetRtpSendParameters(ssrc);
   }
   return RtpParameters();
 }
 
-bool WebRtcSession::SetAudioRtpParameters(uint32_t ssrc,
-                                          const RtpParameters& parameters) {
+bool WebRtcSession::SetAudioRtpSendParameters(uint32_t ssrc,
+                                              const RtpParameters& parameters) {
   ASSERT(signaling_thread()->IsCurrent());
   if (!voice_channel_) {
     return false;
   }
-  return voice_channel_->SetRtpParameters(ssrc, parameters);
+  return voice_channel_->SetRtpSendParameters(ssrc, parameters);
+}
+
+RtpParameters WebRtcSession::GetAudioRtpReceiveParameters(uint32_t ssrc) const {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (voice_channel_) {
+    return voice_channel_->GetRtpReceiveParameters(ssrc);
+  }
+  return RtpParameters();
+}
+
+bool WebRtcSession::SetAudioRtpReceiveParameters(
+    uint32_t ssrc,
+    const RtpParameters& parameters) {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (!voice_channel_) {
+    return false;
+  }
+  return voice_channel_->SetRtpReceiveParameters(ssrc, parameters);
 }
 
 bool WebRtcSession::SetSource(
@@ -1284,21 +1302,39 @@
   }
 }
 
-RtpParameters WebRtcSession::GetVideoRtpParameters(uint32_t ssrc) const {
+RtpParameters WebRtcSession::GetVideoRtpSendParameters(uint32_t ssrc) const {
   ASSERT(signaling_thread()->IsCurrent());
   if (video_channel_) {
-    return video_channel_->GetRtpParameters(ssrc);
+    return video_channel_->GetRtpSendParameters(ssrc);
   }
   return RtpParameters();
 }
 
-bool WebRtcSession::SetVideoRtpParameters(uint32_t ssrc,
-                                          const RtpParameters& parameters) {
+bool WebRtcSession::SetVideoRtpSendParameters(uint32_t ssrc,
+                                              const RtpParameters& parameters) {
   ASSERT(signaling_thread()->IsCurrent());
   if (!video_channel_) {
     return false;
   }
-  return video_channel_->SetRtpParameters(ssrc, parameters);
+  return video_channel_->SetRtpSendParameters(ssrc, parameters);
+}
+
+RtpParameters WebRtcSession::GetVideoRtpReceiveParameters(uint32_t ssrc) const {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (video_channel_) {
+    return video_channel_->GetRtpReceiveParameters(ssrc);
+  }
+  return RtpParameters();
+}
+
+bool WebRtcSession::SetVideoRtpReceiveParameters(
+    uint32_t ssrc,
+    const RtpParameters& parameters) {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (!video_channel_) {
+    return false;
+  }
+  return video_channel_->SetRtpReceiveParameters(ssrc, parameters);
 }
 
 bool WebRtcSession::CanInsertDtmf(const std::string& track_id) {
diff --git a/webrtc/api/webrtcsession.h b/webrtc/api/webrtcsession.h
index 8a32d78f..d0a5cd4 100644
--- a/webrtc/api/webrtcsession.h
+++ b/webrtc/api/webrtcsession.h
@@ -243,9 +243,12 @@
   void SetRawAudioSink(uint32_t ssrc,
                        std::unique_ptr<AudioSinkInterface> sink) override;
 
-  RtpParameters GetAudioRtpParameters(uint32_t ssrc) const override;
-  bool SetAudioRtpParameters(uint32_t ssrc,
-                             const RtpParameters& parameters) override;
+  RtpParameters GetAudioRtpSendParameters(uint32_t ssrc) const override;
+  bool SetAudioRtpSendParameters(uint32_t ssrc,
+                                 const RtpParameters& parameters) override;
+  RtpParameters GetAudioRtpReceiveParameters(uint32_t ssrc) const override;
+  bool SetAudioRtpReceiveParameters(uint32_t ssrc,
+                                    const RtpParameters& parameters) override;
 
   // Implements VideoMediaProviderInterface.
   bool SetSource(
@@ -259,9 +262,12 @@
                     bool enable,
                     const cricket::VideoOptions* options) override;
 
-  RtpParameters GetVideoRtpParameters(uint32_t ssrc) const override;
-  bool SetVideoRtpParameters(uint32_t ssrc,
-                             const RtpParameters& parameters) override;
+  RtpParameters GetVideoRtpSendParameters(uint32_t ssrc) const override;
+  bool SetVideoRtpSendParameters(uint32_t ssrc,
+                                 const RtpParameters& parameters) override;
+  RtpParameters GetVideoRtpReceiveParameters(uint32_t ssrc) const override;
+  bool SetVideoRtpReceiveParameters(uint32_t ssrc,
+                                    const RtpParameters& parameters) override;
 
   // Implements DtmfProviderInterface.
   bool CanInsertDtmf(const std::string& track_id) override;
diff --git a/webrtc/api/webrtcsession_unittest.cc b/webrtc/api/webrtcsession_unittest.cc
index cd5e784..d81aece 100644
--- a/webrtc/api/webrtcsession_unittest.cc
+++ b/webrtc/api/webrtcsession_unittest.cc
@@ -3358,19 +3358,19 @@
   ASSERT_TRUE(channel != NULL);
   uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
   EXPECT_EQ(-1, channel->max_bps());
-  webrtc::RtpParameters params = session_->GetAudioRtpParameters(send_ssrc);
+  webrtc::RtpParameters params = session_->GetAudioRtpSendParameters(send_ssrc);
   EXPECT_EQ(1, params.encodings.size());
   EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps);
   params.encodings[0].max_bitrate_bps = 1000;
-  EXPECT_TRUE(session_->SetAudioRtpParameters(send_ssrc, params));
+  EXPECT_TRUE(session_->SetAudioRtpSendParameters(send_ssrc, params));
 
   // Read back the parameters and verify they have been changed.
-  params = session_->GetAudioRtpParameters(send_ssrc);
+  params = session_->GetAudioRtpSendParameters(send_ssrc);
   EXPECT_EQ(1, params.encodings.size());
   EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
 
   // Verify that the audio channel received the new parameters.
-  params = channel->GetRtpParameters(send_ssrc);
+  params = channel->GetRtpSendParameters(send_ssrc);
   EXPECT_EQ(1, params.encodings.size());
   EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
 
@@ -3452,19 +3452,19 @@
   ASSERT_TRUE(channel != NULL);
   uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
   EXPECT_EQ(-1, channel->max_bps());
-  webrtc::RtpParameters params = session_->GetVideoRtpParameters(send_ssrc);
+  webrtc::RtpParameters params = session_->GetVideoRtpSendParameters(send_ssrc);
   EXPECT_EQ(1, params.encodings.size());
   EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps);
   params.encodings[0].max_bitrate_bps = 1000;
-  EXPECT_TRUE(session_->SetVideoRtpParameters(send_ssrc, params));
+  EXPECT_TRUE(session_->SetVideoRtpSendParameters(send_ssrc, params));
 
   // Read back the parameters and verify they have been changed.
-  params = session_->GetVideoRtpParameters(send_ssrc);
+  params = session_->GetVideoRtpSendParameters(send_ssrc);
   EXPECT_EQ(1, params.encodings.size());
   EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
 
   // Verify that the video channel received the new parameters.
-  params = channel->GetRtpParameters(send_ssrc);
+  params = channel->GetRtpSendParameters(send_ssrc);
   EXPECT_EQ(1, params.encodings.size());
   EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);