diff --git a/sdk/objc/api/peerconnection/RTCConfiguration.h b/sdk/objc/api/peerconnection/RTCConfiguration.h
index f1400813d6..f9e6edfd97 100644
--- a/sdk/objc/api/peerconnection/RTCConfiguration.h
+++ b/sdk/objc/api/peerconnection/RTCConfiguration.h
@@ -201,6 +201,16 @@ RTC_OBJC_EXPORT
  */
 @property(nonatomic, nullable) RTCCryptoOptions *cryptoOptions;
 
+/**
+ * Time interval between audio RTCP reports.
+ */
+@property(nonatomic, assign) int rtcpAudioReportIntervalMs;
+
+/**
+ * Time interval between video RTCP reports.
+ */
+@property(nonatomic, assign) int rtcpVideoReportIntervalMs;
+
 - (instancetype)init;
 
 @end
diff --git a/sdk/objc/api/peerconnection/RTCConfiguration.mm b/sdk/objc/api/peerconnection/RTCConfiguration.mm
index 55bf26f38d..4041f8399f 100644
--- a/sdk/objc/api/peerconnection/RTCConfiguration.mm
+++ b/sdk/objc/api/peerconnection/RTCConfiguration.mm
@@ -53,6 +53,8 @@
 @synthesize useMediaTransport = _useMediaTransport;
 @synthesize useMediaTransportForDataChannels = _useMediaTransportForDataChannels;
 @synthesize cryptoOptions = _cryptoOptions;
+@synthesize rtcpAudioReportIntervalMs = _rtcpAudioReportIntervalMs;
+@synthesize rtcpVideoReportIntervalMs = _rtcpVideoReportIntervalMs;
 
 - (instancetype)init {
   // Copy defaults.
@@ -130,6 +132,8 @@
                     sframeRequireFrameEncryption:config.crypto_options->sframe
                                                      .require_frame_encryption];
     }
+    _rtcpAudioReportIntervalMs = config.audio_rtcp_report_interval_ms();
+    _rtcpVideoReportIntervalMs = config.video_rtcp_report_interval_ms();
   }
   return self;
 }
@@ -261,7 +265,8 @@
         _cryptoOptions.sframeRequireFrameEncryption ? true : false;
     nativeConfig->crypto_options = absl::optional<webrtc::CryptoOptions>(nativeCryptoOptions);
   }
-
+  nativeConfig->set_audio_rtcp_report_interval_ms(_rtcpAudioReportIntervalMs);
+  nativeConfig->set_video_rtcp_report_interval_ms(_rtcpVideoReportIntervalMs);
   return nativeConfig.release();
 }
 
diff --git a/sdk/objc/unittests/RTCConfigurationTest.mm b/sdk/objc/unittests/RTCConfigurationTest.mm
index 67a303e550..f31fcfd858 100644
--- a/sdk/objc/unittests/RTCConfigurationTest.mm
+++ b/sdk/objc/unittests/RTCConfigurationTest.mm
@@ -54,6 +54,8 @@
                                                  srtpEnableAes128Sha1_32CryptoCipher:YES
                                               srtpEnableEncryptedRtpHeaderExtensions:YES
                                                         sframeRequireFrameEncryption:YES];
+  config.rtcpAudioReportIntervalMs = 2500;
+  config.rtcpVideoReportIntervalMs = 3750;
 
   std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration>
       nativeConfig([config createNativeConfiguration]);
@@ -86,6 +88,8 @@
   EXPECT_EQ(true, nativeConfig->crypto_options->srtp.enable_aes128_sha1_32_crypto_cipher);
   EXPECT_EQ(true, nativeConfig->crypto_options->srtp.enable_encrypted_rtp_header_extensions);
   EXPECT_EQ(true, nativeConfig->crypto_options->sframe.require_frame_encryption);
+  EXPECT_EQ(2500, nativeConfig->audio_rtcp_report_interval_ms());
+  EXPECT_EQ(3750, nativeConfig->video_rtcp_report_interval_ms());
 }
 
 - (void)testNativeConversionToConfiguration {
@@ -115,6 +119,8 @@
                                                  srtpEnableAes128Sha1_32CryptoCipher:NO
                                               srtpEnableEncryptedRtpHeaderExtensions:NO
                                                         sframeRequireFrameEncryption:NO];
+  config.rtcpAudioReportIntervalMs = 1500;
+  config.rtcpVideoReportIntervalMs = 2150;
 
   webrtc::PeerConnectionInterface::RTCConfiguration *nativeConfig =
       [config createNativeConfiguration];
@@ -150,6 +156,8 @@
             newConfig.cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions);
   EXPECT_EQ(config.cryptoOptions.sframeRequireFrameEncryption,
             newConfig.cryptoOptions.sframeRequireFrameEncryption);
+  EXPECT_EQ(config.rtcpAudioReportIntervalMs, newConfig.rtcpAudioReportIntervalMs);
+  EXPECT_EQ(config.rtcpVideoReportIntervalMs, newConfig.rtcpVideoReportIntervalMs);
 }
 
 - (void)testDefaultValues {