Add Rtcp parameters for PeerConnection senders
Bug: webrtc:7580
Change-Id: Ibcf5e849a1f11f21fa75f6d006fecf1cd54f8552
Reviewed-on: https://webrtc-review.googlesource.com/78063
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23407}diff --git a/api/ortc/rtptransportinterface.h b/api/ortc/rtptransportinterface.h
index 716a297..8822300 100644
--- a/api/ortc/rtptransportinterface.h
+++ b/api/ortc/rtptransportinterface.h
@@ -17,42 +17,13 @@
#include "api/ortc/packettransportinterface.h"
#include "api/rtcerror.h"
#include "api/rtp_headers.h"
+#include "api/rtpparameters.h"
#include "common_types.h" // NOLINT(build/include)
namespace webrtc {
class RtpTransportAdapter;
-struct RtcpParameters final {
- // The SSRC to be used in the "SSRC of packet sender" field. If not set, one
- // will be chosen by the implementation.
- // TODO(deadbeef): Not implemented.
- rtc::Optional<uint32_t> ssrc;
-
- // The Canonical Name (CNAME) used by RTCP (e.g. in SDES messages).
- //
- // If empty in the construction of the RtpTransport, one will be generated by
- // the implementation, and returned in GetRtcpParameters. Multiple
- // RtpTransports created by the same OrtcFactory will use the same generated
- // CNAME.
- //
- // If empty when passed into SetParameters, the CNAME simply won't be
- // modified.
- std::string cname;
-
- // Send reduced-size RTCP?
- bool reduced_size = false;
-
- // Send RTCP multiplexed on the RTP transport?
- bool mux = true;
-
- bool operator==(const RtcpParameters& o) const {
- return ssrc == o.ssrc && cname == o.cname &&
- reduced_size == o.reduced_size && mux == o.mux;
- }
- bool operator!=(const RtcpParameters& o) const { return !(*this == o); }
-};
-
struct RtpTransportParameters final {
RtcpParameters rtcp;
diff --git a/api/rtpparameters.cc b/api/rtpparameters.cc
index ed48091..5a873de 100644
--- a/api/rtpparameters.cc
+++ b/api/rtpparameters.cc
@@ -65,6 +65,9 @@
RtpCapabilities::RtpCapabilities() {}
RtpCapabilities::~RtpCapabilities() {}
+RtcpParameters::RtcpParameters() {}
+RtcpParameters::~RtcpParameters() {}
+
RtpParameters::RtpParameters() {}
RtpParameters::~RtpParameters() {}
diff --git a/api/rtpparameters.h b/api/rtpparameters.h
index 96df9ce..c12b0c9 100644
--- a/api/rtpparameters.h
+++ b/api/rtpparameters.h
@@ -547,9 +547,40 @@
bool operator!=(const RtpCapabilities& o) const { return !(*this == o); }
};
-// Note that unlike in ORTC, an RtcpParameters structure is not included in
-// RtpParameters, because our API includes an additional "RtpTransport"
-// abstraction on which RTCP parameters are set.
+struct RtcpParameters final {
+ RtcpParameters();
+ ~RtcpParameters();
+
+ // The SSRC to be used in the "SSRC of packet sender" field. If not set, one
+ // will be chosen by the implementation.
+ // TODO(deadbeef): Not implemented.
+ rtc::Optional<uint32_t> ssrc;
+
+ // The Canonical Name (CNAME) used by RTCP (e.g. in SDES messages).
+ //
+ // If empty in the construction of the RtpTransport, one will be generated by
+ // the implementation, and returned in GetRtcpParameters. Multiple
+ // RtpTransports created by the same OrtcFactory will use the same generated
+ // CNAME.
+ //
+ // If empty when passed into SetParameters, the CNAME simply won't be
+ // modified.
+ std::string cname;
+
+ // Send reduced-size RTCP?
+ bool reduced_size = false;
+
+ // Send RTCP multiplexed on the RTP transport?
+ // Not used with PeerConnection senders/receivers
+ bool mux = true;
+
+ bool operator==(const RtcpParameters& o) const {
+ return ssrc == o.ssrc && cname == o.cname &&
+ reduced_size == o.reduced_size && mux == o.mux;
+ }
+ bool operator!=(const RtcpParameters& o) const { return !(*this == o); }
+};
+
struct RtpParameters {
RtpParameters();
~RtpParameters();
@@ -571,6 +602,11 @@
std::vector<RtpEncodingParameters> encodings;
+ // Only available with a Peerconnection RtpSender.
+ // In ORTC, our API includes an additional "RtpTransport"
+ // abstraction on which RTCP parameters are set.
+ RtcpParameters rtcp;
+
// TODO(deadbeef): Not implemented.
DegradationPreference degradation_preference =
DegradationPreference::BALANCED;
@@ -578,7 +614,7 @@
bool operator==(const RtpParameters& o) const {
return mid == o.mid && codecs == o.codecs &&
header_extensions == o.header_extensions &&
- encodings == o.encodings &&
+ encodings == o.encodings && rtcp == o.rtcp &&
degradation_preference == o.degradation_preference;
}
bool operator!=(const RtpParameters& o) const { return !(*this == o); }
diff --git a/media/base/mediaengine.cc b/media/base/mediaengine.cc
index d40f765..b0fede3 100644
--- a/media/base/mediaengine.cc
+++ b/media/base/mediaengine.cc
@@ -33,6 +33,7 @@
}
webrtc::RtpParameters parameters;
parameters.encodings = encodings;
+ parameters.rtcp.cname = sp.cname;
return parameters;
}
diff --git a/media/engine/webrtcvideoengine.cc b/media/engine/webrtcvideoengine.cc
index 532f57e..0b89e1f 100644
--- a/media/engine/webrtcvideoengine.cc
+++ b/media/engine/webrtcvideoengine.cc
@@ -1642,6 +1642,8 @@
: webrtc::RtcpMode::kCompound;
parameters_.config.rtp.mid = send_params.mid;
+ rtp_parameters_.rtcp.reduced_size = send_params.rtcp.reduced_size;
+
if (codec_settings) {
SetCodec(*codec_settings);
}
@@ -1761,6 +1763,8 @@
bool recreate_stream = false;
if (params.rtcp_mode) {
parameters_.config.rtp.rtcp_mode = *params.rtcp_mode;
+ rtp_parameters_.rtcp.reduced_size =
+ parameters_.config.rtp.rtcp_mode == webrtc::RtcpMode::kReducedSize;
recreate_stream = true;
}
if (params.rtp_header_extensions) {
@@ -1847,6 +1851,11 @@
RTCErrorType::INVALID_MODIFICATION,
"Attempted to set RtpParameters with different encoding count");
}
+ if (rtp_parameters.rtcp != rtp_parameters_.rtcp) {
+ LOG_AND_RETURN_ERROR(
+ RTCErrorType::INVALID_MODIFICATION,
+ "Attempted to set RtpParameters with modified RTCP parameters");
+ }
if (rtp_parameters.encodings[0].ssrc != rtp_parameters_.encodings[0].ssrc) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
"Attempted to set RtpParameters with modified SSRC");
diff --git a/media/engine/webrtcvideoengine_unittest.cc b/media/engine/webrtcvideoengine_unittest.cc
index 6ecf750..7d640a6 100644
--- a/media/engine/webrtcvideoengine_unittest.cc
+++ b/media/engine/webrtcvideoengine_unittest.cc
@@ -4439,12 +4439,17 @@
// Create stream, expecting that default mode is "compound".
FakeVideoSendStream* stream1 = AddSendStream();
EXPECT_EQ(webrtc::RtcpMode::kCompound, stream1->GetConfig().rtp.rtcp_mode);
+ webrtc::RtpParameters rtp_parameters =
+ channel_->GetRtpSendParameters(last_ssrc_);
+ EXPECT_FALSE(rtp_parameters.rtcp.reduced_size);
// Now enable reduced size mode.
send_parameters_.rtcp.reduced_size = true;
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
stream1 = fake_call_->GetVideoSendStreams()[0];
EXPECT_EQ(webrtc::RtcpMode::kReducedSize, stream1->GetConfig().rtp.rtcp_mode);
+ rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
+ EXPECT_TRUE(rtp_parameters.rtcp.reduced_size);
// Create a new stream and ensure it picks up the reduced size mode.
FakeVideoSendStream* stream2 = AddSendStream();
@@ -5543,6 +5548,16 @@
rtp_parameters.codecs[1]);
}
+// Test that GetRtpSendParameters returns the currently configured RTCP CNAME.
+TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersRtcpCname) {
+ StreamParams params = StreamParams::CreateLegacy(kSsrc);
+ params.cname = "rtcpcname";
+ AddSendStream(params);
+
+ webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrc);
+ EXPECT_STREQ("rtcpcname", rtp_parameters.rtcp.cname.c_str());
+}
+
// Test that RtpParameters for send stream has one encoding and it has
// the correct SSRC.
TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersSsrc) {
diff --git a/media/engine/webrtcvoiceengine.cc b/media/engine/webrtcvoiceengine.cc
index bb1c9c3..cb82bbd 100644
--- a/media/engine/webrtcvoiceengine.cc
+++ b/media/engine/webrtcvoiceengine.cc
@@ -776,6 +776,7 @@
config_.codec_pair_id = codec_pair_id;
config_.track_id = track_id;
rtp_parameters_.encodings[0].ssrc = ssrc;
+ rtp_parameters_.rtcp.cname = c_name;
if (send_codec_spec) {
UpdateSendCodecSpec(*send_codec_spec);
@@ -945,6 +946,11 @@
RTCErrorType::INVALID_MODIFICATION,
"Attempted to set RtpParameters with different encoding count");
}
+ if (rtp_parameters.rtcp != rtp_parameters_.rtcp) {
+ LOG_AND_RETURN_ERROR(
+ RTCErrorType::INVALID_MODIFICATION,
+ "Attempted to set RtpParameters with modified RTCP parameters");
+ }
if (rtp_parameters.encodings[0].ssrc != rtp_parameters_.encodings[0].ssrc) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
"Attempted to set RtpParameters with modified SSRC");
@@ -992,6 +998,10 @@
if (reconfigure_send_stream) {
ReconfigureAudioSendStream();
}
+
+ rtp_parameters_.rtcp.cname = config_.rtp.c_name;
+ rtp_parameters_.rtcp.reduced_size = false;
+
// parameters.encodings[0].active could have changed.
UpdateSendState();
return webrtc::RTCError::OK();
diff --git a/media/engine/webrtcvoiceengine_unittest.cc b/media/engine/webrtcvoiceengine_unittest.cc
index d574a72..966ca6b 100644
--- a/media/engine/webrtcvoiceengine_unittest.cc
+++ b/media/engine/webrtcvoiceengine_unittest.cc
@@ -217,10 +217,14 @@
}
bool SetupSendStream() {
+ return SetupSendStream(cricket::StreamParams::CreateLegacy(kSsrcX));
+ }
+
+ bool SetupSendStream(const cricket::StreamParams& sp) {
if (!SetupChannel()) {
return false;
}
- if (!channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX))) {
+ if (!channel_->AddSendStream(sp)) {
return false;
}
EXPECT_CALL(*apm_, set_output_will_be_muted(false));
@@ -1131,6 +1135,16 @@
EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]);
}
+// Test that GetRtpSendParameters returns the currently configured RTCP CNAME.
+TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersRtcpCname) {
+ cricket::StreamParams params = cricket::StreamParams::CreateLegacy(kSsrcX);
+ params.cname = "rtcpcname";
+ EXPECT_TRUE(SetupSendStream(params));
+
+ webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
+ EXPECT_STREQ("rtcpcname", rtp_parameters.rtcp.cname.c_str());
+}
+
// Test that GetRtpSendParameters returns an SSRC.
TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersSsrc) {
EXPECT_TRUE(SetupSendStream());
diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
index 3f70c75..f8bac07 100644
--- a/sdk/BUILD.gn
+++ b/sdk/BUILD.gn
@@ -683,6 +683,8 @@
"objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory.mm",
"objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryOptions+Private.h",
"objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryOptions.mm",
+ "objc/Framework/Classes/PeerConnection/RTCRtcpParameters+Private.h",
+ "objc/Framework/Classes/PeerConnection/RTCRtcpParameters.mm",
"objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters+Private.h",
"objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters.mm",
"objc/Framework/Classes/PeerConnection/RTCRtpEncodingParameters+Private.h",
@@ -718,6 +720,7 @@
"objc/Framework/Headers/WebRTC/RTCPeerConnection.h",
"objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h",
"objc/Framework/Headers/WebRTC/RTCPeerConnectionFactoryOptions.h",
+ "objc/Framework/Headers/WebRTC/RTCRtcpParameters.h",
"objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h",
"objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h",
"objc/Framework/Headers/WebRTC/RTCRtpParameters.h",
diff --git a/sdk/android/api/org/webrtc/RtpParameters.java b/sdk/android/api/org/webrtc/RtpParameters.java
index 0e893bd..e376c08 100644
--- a/sdk/android/api/org/webrtc/RtpParameters.java
+++ b/sdk/android/api/org/webrtc/RtpParameters.java
@@ -118,8 +118,33 @@
}
}
+ public static class Rtcp {
+ /** The Canonical Name used by RTCP */
+ private final String cname;
+ /** Whether reduced size RTCP is configured or compound RTCP */
+ private final boolean reducedSize;
+
+ @CalledByNative("Rtcp")
+ Rtcp(String cname, boolean reducedSize) {
+ this.cname = cname;
+ this.reducedSize = reducedSize;
+ }
+
+ @CalledByNative("Rtcp")
+ public String getCname() {
+ return cname;
+ }
+
+ @CalledByNative("Rtcp")
+ public boolean getReducedSize() {
+ return reducedSize;
+ }
+ }
+
public final String transactionId;
+ private final Rtcp rtcp;
+
public final List<Encoding> encodings;
// Codec parameters can't currently be changed between getParameters and
// setParameters. Though in the future it will be possible to reorder them or
@@ -127,8 +152,9 @@
public final List<Codec> codecs;
@CalledByNative
- RtpParameters(String transactionId, List<Encoding> encodings, List<Codec> codecs) {
+ RtpParameters(String transactionId, Rtcp rtcp, List<Encoding> encodings, List<Codec> codecs) {
this.transactionId = transactionId;
+ this.rtcp = rtcp;
this.encodings = encodings;
this.codecs = codecs;
}
@@ -139,6 +165,11 @@
}
@CalledByNative
+ public Rtcp getRtcp() {
+ return rtcp;
+ }
+
+ @CalledByNative
List<Encoding> getEncodings() {
return encodings;
}
diff --git a/sdk/android/src/jni/pc/rtpparameters.cc b/sdk/android/src/jni/pc/rtpparameters.cc
index b0b83eb..385eb8a 100644
--- a/sdk/android/src/jni/pc/rtpparameters.cc
+++ b/sdk/android/src/jni/pc/rtpparameters.cc
@@ -39,6 +39,13 @@
NativeToJavaStringMap(env, codec.parameters));
}
+ScopedJavaLocalRef<jobject> NativeToJavaRtpRtcpParameters(
+ JNIEnv* env,
+ const RtcpParameters& rtcp) {
+ return Java_Rtcp_Constructor(env, NativeToJavaString(env, rtcp.cname),
+ rtcp.reduced_size);
+}
+
} // namespace
RtpEncodingParameters JavaToNativeRtpEncodingParameters(
@@ -64,6 +71,13 @@
Java_RtpParameters_getTransactionId(jni, j_parameters);
parameters.transaction_id = JavaToNativeString(jni, j_transaction_id);
+ ScopedJavaLocalRef<jobject> j_rtcp =
+ Java_RtpParameters_getRtcp(jni, j_parameters);
+ ScopedJavaLocalRef<jstring> j_rtcp_cname = Java_Rtcp_getCname(jni, j_rtcp);
+ jboolean j_rtcp_reduced_size = Java_Rtcp_getReducedSize(jni, j_rtcp);
+ parameters.rtcp.cname = JavaToNativeString(jni, j_rtcp_cname);
+ parameters.rtcp.reduced_size = j_rtcp_reduced_size;
+
// Convert encodings.
ScopedJavaLocalRef<jobject> j_encodings =
Java_RtpParameters_getEncodings(jni, j_parameters);
@@ -99,6 +113,7 @@
const RtpParameters& parameters) {
return Java_RtpParameters_Constructor(
env, NativeToJavaString(env, parameters.transaction_id),
+ NativeToJavaRtpRtcpParameters(env, parameters.rtcp),
NativeToJavaList(env, parameters.encodings,
&NativeToJavaRtpEncodingParameter),
NativeToJavaList(env, parameters.codecs, &NativeToJavaRtpCodecParameter));
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCRtcpParameters+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCRtcpParameters+Private.h
new file mode 100644
index 0000000..4157ffe
--- /dev/null
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCRtcpParameters+Private.h
@@ -0,0 +1,27 @@
+/*
+ * Copyright 2018 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.
+ */
+
+#import "WebRTC/RTCRtcpParameters.h"
+
+#include "api/rtpparameters.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCRtcpParameters ()
+
+/** Returns the equivalent native RtcpParameters structure. */
+@property(nonatomic, readonly) webrtc::RtcpParameters nativeParameters;
+
+/** Initialize the object with a native RtcpParameters structure. */
+- (instancetype)initWithNativeParameters:(const webrtc::RtcpParameters &)nativeParameters;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCRtcpParameters.mm b/sdk/objc/Framework/Classes/PeerConnection/RTCRtcpParameters.mm
new file mode 100644
index 0000000..1c8a31b
--- /dev/null
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCRtcpParameters.mm
@@ -0,0 +1,39 @@
+/*
+ * Copyright 2018 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.
+ */
+
+#import "RTCRtcpParameters+Private.h"
+
+#import "NSString+StdString.h"
+
+@implementation RTCRtcpParameters
+
+@synthesize cname = _cname;
+@synthesize isReducedSize = _isReducedSize;
+
+- (instancetype)init {
+ return [super init];
+}
+
+- (instancetype)initWithNativeParameters:(const webrtc::RtcpParameters &)nativeParameters {
+ if (self = [self init]) {
+ _cname = [NSString stringForStdString:nativeParameters.cname];
+ _isReducedSize = nativeParameters.reduced_size;
+ }
+ return self;
+}
+
+- (webrtc::RtcpParameters)nativeParameters {
+ webrtc::RtcpParameters parameters;
+ parameters.cname = [NSString stdStringForString:_cname];
+ parameters.reduced_size = _isReducedSize;
+ return parameters;
+}
+
+@end
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCRtpParameters.mm b/sdk/objc/Framework/Classes/PeerConnection/RTCRtpParameters.mm
index d18eba6..7c00b08 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCRtpParameters.mm
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCRtpParameters.mm
@@ -11,12 +11,14 @@
#import "RTCRtpParameters+Private.h"
#import "NSString+StdString.h"
+#import "RTCRtcpParameters+Private.h"
#import "RTCRtpCodecParameters+Private.h"
#import "RTCRtpEncodingParameters+Private.h"
@implementation RTCRtpParameters
@synthesize transactionId = _transactionId;
+@synthesize rtcp = _rtcp;
@synthesize encodings = _encodings;
@synthesize codecs = _codecs;
@@ -28,6 +30,7 @@
(const webrtc::RtpParameters &)nativeParameters {
if (self = [self init]) {
_transactionId = [NSString stringForStdString:nativeParameters.transaction_id];
+ _rtcp = [[RTCRtcpParameters alloc] initWithNativeParameters:nativeParameters.rtcp];
NSMutableArray *encodings = [[NSMutableArray alloc] init];
for (const auto &encoding : nativeParameters.encodings) {
[encodings addObject:[[RTCRtpEncodingParameters alloc]
@@ -48,6 +51,7 @@
- (webrtc::RtpParameters)nativeParameters {
webrtc::RtpParameters parameters;
parameters.transaction_id = [NSString stdStringForString:_transactionId];
+ parameters.rtcp = [_rtcp nativeParameters];
for (RTCRtpEncodingParameters *encoding in _encodings) {
parameters.encodings.push_back(encoding.nativeParameters);
}
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCRtcpParameters.h b/sdk/objc/Framework/Headers/WebRTC/RTCRtcpParameters.h
new file mode 100644
index 0000000..54b254c
--- /dev/null
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCRtcpParameters.h
@@ -0,0 +1,30 @@
+/*
+ * Copyright 2018 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.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import <WebRTC/RTCMacros.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXPORT
+@interface RTCRtcpParameters : NSObject
+
+/** The Canonical Name used by RTCP. */
+@property(nonatomic, readonly, copy) NSString *cname;
+
+/** Whether reduced size RTCP is configured or compound RTCP. */
+@property(nonatomic, assign) BOOL isReducedSize;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h b/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
index b6e1f01..02ce981 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
@@ -11,6 +11,7 @@
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
+#import <WebRTC/RTCRtcpParameters.h>
#import <WebRTC/RTCRtpCodecParameters.h>
#import <WebRTC/RTCRtpEncodingParameters.h>
@@ -22,6 +23,9 @@
/** A unique identifier for the last set of parameters applied. */
@property(nonatomic, copy) NSString *transactionId;
+/** Parameters used for RTCP. */
+@property(nonatomic, readonly, copy) RTCRtcpParameters *rtcp;
+
/** The currently active encodings in the order of preference. */
@property(nonatomic, copy) NSArray<RTCRtpEncodingParameters *> *encodings;
