diff --git a/libwebrtc/src/native/rtp_receiver.rs b/libwebrtc/src/native/rtp_receiver.rs
index b0fe64bc3..ce59e92b4 100644
--- a/libwebrtc/src/native/rtp_receiver.rs
+++ b/libwebrtc/src/native/rtp_receiver.rs
@@ -15,7 +15,10 @@
 use crate::imp::media_stream_track::new_media_stream_track;
 use crate::media_stream_track::MediaStreamTrack;
 use crate::rtp_parameters::RtpParameters;
+use crate::stats::RtcStats;
+use crate::{RtcError, RtcErrorType};
 use cxx::SharedPtr;
+use tokio::sync::oneshot;
 use webrtc_sys::rtp_receiver as sys_rr;
 
 #[derive(Clone)]
@@ -33,6 +36,27 @@ impl RtpReceiver {
         Some(new_media_stream_track(track_handle))
     }
 
+    pub async fn get_stats(&self) -> Result<Vec<RtcStats>, RtcError> {
+        let (tx, rx) = oneshot::channel::<Result<Vec<RtcStats>, RtcError>>();
+        let ctx = Box::new(sys_rr::ReceiverContext(Box::new(tx)));
+
+        self.sys_handle.get_stats(ctx, |ctx, stats| {
+            let tx = ctx
+                .0
+                .downcast::<oneshot::Sender<Result<Vec<RtcStats>, RtcError>>>()
+                .unwrap();
+
+            // Unwrap because it should not happens
+            let vec = serde_json::from_str(&stats).unwrap();
+            let _ = tx.send(Ok(vec));
+        });
+
+        rx.await.map_err(|_| RtcError {
+            error_type: RtcErrorType::Internal,
+            message: "get_stats cancelled".to_owned(),
+        })?
+    }
+
     pub fn parameters(&self) -> RtpParameters {
         self.sys_handle.get_parameters().into()
     }
diff --git a/libwebrtc/src/native/rtp_sender.rs b/libwebrtc/src/native/rtp_sender.rs
index 1af9537de..fcb7b06f4 100644
--- a/libwebrtc/src/native/rtp_sender.rs
+++ b/libwebrtc/src/native/rtp_sender.rs
@@ -14,8 +14,10 @@
 
 use super::media_stream_track::new_media_stream_track;
 use crate::media_stream_track::MediaStreamTrack;
+use crate::stats::RtcStats;
 use crate::{rtp_parameters::RtpParameters, RtcError, RtcErrorType};
 use cxx::SharedPtr;
+use tokio::sync::oneshot;
 use webrtc_sys::rtc_error as sys_err;
 use webrtc_sys::rtp_sender as sys_rs;
 
@@ -34,6 +36,27 @@ impl RtpSender {
         Some(new_media_stream_track(track_handle))
     }
 
+    pub async fn get_stats(&self) -> Result<Vec<RtcStats>, RtcError> {
+        let (tx, rx) = oneshot::channel::<Result<Vec<RtcStats>, RtcError>>();
+        let ctx = Box::new(sys_rs::SenderContext(Box::new(tx)));
+
+        self.sys_handle.get_stats(ctx, |ctx, stats| {
+            let tx = ctx
+                .0
+                .downcast::<oneshot::Sender<Result<Vec<RtcStats>, RtcError>>>()
+                .unwrap();
+
+            // Unwrap because it should not happens
+            let vec = serde_json::from_str(&stats).unwrap();
+            let _ = tx.send(Ok(vec));
+        });
+
+        rx.await.map_err(|_| RtcError {
+            error_type: RtcErrorType::Internal,
+            message: "get_stats cancelled".to_owned(),
+        })?
+    }
+
     pub fn set_track(&self, track: Option<MediaStreamTrack>) -> Result<(), RtcError> {
         if !self
             .sys_handle
diff --git a/libwebrtc/src/rtp_receiver.rs b/libwebrtc/src/rtp_receiver.rs
index 765b329ed..e8b297ffa 100644
--- a/libwebrtc/src/rtp_receiver.rs
+++ b/libwebrtc/src/rtp_receiver.rs
@@ -16,7 +16,7 @@ use std::fmt::Debug;
 
 use crate::{
     imp::rtp_receiver as imp_rr, media_stream_track::MediaStreamTrack,
-    rtp_parameters::RtpParameters,
+    rtp_parameters::RtpParameters, stats::RtcStats, RtcError,
 };
 
 #[derive(Clone)]
@@ -29,6 +29,10 @@ impl RtpReceiver {
         self.handle.track()
     }
 
+    pub async fn get_stats(&self) -> Result<Vec<RtcStats>, RtcError> {
+        self.handle.get_stats().await
+    }
+
     pub fn parameters(&self) -> RtpParameters {
         self.handle.parameters()
     }
diff --git a/libwebrtc/src/rtp_sender.rs b/libwebrtc/src/rtp_sender.rs
index 300c6f87e..3301e2532 100644
--- a/libwebrtc/src/rtp_sender.rs
+++ b/libwebrtc/src/rtp_sender.rs
@@ -16,7 +16,7 @@ use std::fmt::Debug;
 
 use crate::{
     imp::rtp_sender as imp_rs, media_stream_track::MediaStreamTrack, rtp_parameters::RtpParameters,
-    RtcError,
+    stats::RtcStats, RtcError,
 };
 
 #[derive(Clone)]
@@ -29,6 +29,10 @@ impl RtpSender {
         self.handle.track()
     }
 
+    pub async fn get_stats(&self) -> Result<Vec<RtcStats>, RtcError> {
+        self.handle.get_stats().await
+    }
+
     pub fn set_track(&self, track: Option<MediaStreamTrack>) -> Result<(), RtcError> {
         self.handle.set_track(track)
     }
diff --git a/webrtc-sys/include/livekit/jsep.h b/webrtc-sys/include/livekit/jsep.h
index 90092321d..3cb59a0c1 100644
--- a/webrtc-sys/include/livekit/jsep.h
+++ b/webrtc-sys/include/livekit/jsep.h
@@ -132,7 +132,7 @@ class NativeSetRemoteSdpObserver
 };
 
 
-template<class T>
+template<class T> // Context type
 class NativeRtcStatsCollector : public webrtc::RTCStatsCollectorCallback {
  public:
   NativeRtcStatsCollector(rust::Box<T> ctx,