diff --git a/internal/webrtc/manager.go b/internal/webrtc/manager.go
index e3e454a1..7bf18d41 100644
--- a/internal/webrtc/manager.go
+++ b/internal/webrtc/manager.go
@@ -400,6 +400,13 @@ func (manager *WebRTCManagerCtx) CreatePeer(session types.Session, videoID strin
 			})
 	})
 
+	videoTrack.OnRTCP(func(p rtcp.Packet) {
+		switch rtcpPacket := p.(type) {
+		case *rtcp.ReceiverEstimatedMaximumBitrate: // TODO: Deprecated.
+			manager.metrics.SetReceiverEstimatedMaximumBitrate(session, rtcpPacket.Bitrate)
+		}
+	})
+
 	go func() {
 		ticker := time.NewTicker(5 * time.Second)
 		defer ticker.Stop()
diff --git a/internal/webrtc/metrics.go b/internal/webrtc/metrics.go
index 14c99def..066f011b 100644
--- a/internal/webrtc/metrics.go
+++ b/internal/webrtc/metrics.go
@@ -21,6 +21,8 @@ type metrics struct {
 	videoIds   map[string]prometheus.Gauge
 	videoIdsMu *sync.Mutex
 
+	receiverEstimatedMaximumBitrate prometheus.Gauge
+
 	iceBytesSent      prometheus.Gauge
 	iceBytesReceived  prometheus.Gauge
 	sctpBytesSent     prometheus.Gauge
@@ -92,6 +94,16 @@ func (m *metricsCtx) getBySession(session types.Session) metrics {
 		videoIds:   map[string]prometheus.Gauge{},
 		videoIdsMu: &sync.Mutex{},
 
+		receiverEstimatedMaximumBitrate: promauto.NewGauge(prometheus.GaugeOpts{
+			Name:      "receiver_estimated_maximum_bitrate",
+			Namespace: "neko",
+			Subsystem: "webrtc",
+			Help:      "Receiver Estimated Maximum Bitrate from SCTP.",
+			ConstLabels: map[string]string{
+				"session_id": session.ID(),
+			},
+		}),
+
 		iceBytesSent: promauto.NewGauge(prometheus.GaugeOpts{
 			Name:      "ice_bytes_sent",
 			Namespace: "neko",
@@ -205,6 +217,12 @@ func (m *metricsCtx) SetVideoID(session types.Session, videoId string) {
 	}
 }
 
+func (m *metricsCtx) SetReceiverEstimatedMaximumBitrate(session types.Session, bitrate float32) {
+	met := m.getBySession(session)
+
+	met.receiverEstimatedMaximumBitrate.Set(float64(bitrate))
+}
+
 func (m *metricsCtx) SetIceTransportStats(session types.Session, data webrtc.TransportStats) {
 	met := m.getBySession(session)
 
diff --git a/internal/webrtc/peerstreamtrack.go b/internal/webrtc/peerstreamtrack.go
index b77cab2a..2d2fa59b 100644
--- a/internal/webrtc/peerstreamtrack.go
+++ b/internal/webrtc/peerstreamtrack.go
@@ -5,6 +5,7 @@ import (
 	"io"
 	"sync"
 
+	"github.com/pion/rtcp"
 	"github.com/pion/webrtc/v3"
 	"github.com/pion/webrtc/v3/pkg/media"
 	"github.com/rs/zerolog"
@@ -51,6 +52,9 @@ type PeerStreamTrack struct {
 
 	stream   types.StreamSinkManager
 	streamMu sync.Mutex
+
+	onRtcp   func(rtcp.Packet)
+	onRtcpMu sync.RWMutex
 }
 
 func (peer *PeerStreamTrack) SetStream(stream types.StreamSinkManager) error {
@@ -90,8 +94,30 @@ func (peer *PeerStreamTrack) AddToConnection(connection *webrtc.PeerConnection)
 	go func() {
 		rtcpBuf := make([]byte, 1500)
 		for {
-			if _, _, err := sender.Read(rtcpBuf); err != nil {
-				return
+			n, _, err := sender.Read(rtcpBuf)
+			if err != nil {
+				if err == io.EOF || err == io.ErrClosedPipe {
+					return
+				}
+
+				peer.logger.Err(err).Msg("RTCP read error")
+				continue
+			}
+
+			packets, err := rtcp.Unmarshal(rtcpBuf[:n])
+			if err != nil {
+				peer.logger.Err(err).Msg("RTCP unmarshal error")
+				continue
+			}
+
+			peer.onRtcpMu.RLock()
+			handler := peer.onRtcp
+			peer.onRtcpMu.RUnlock()
+
+			for _, packet := range packets {
+				if handler != nil {
+					go handler(packet)
+				}
 			}
 		}
 	}()
@@ -102,3 +128,10 @@ func (peer *PeerStreamTrack) AddToConnection(connection *webrtc.PeerConnection)
 func (peer *PeerStreamTrack) SetPaused(paused bool) {
 	peer.paused = paused
 }
+
+func (peer *PeerStreamTrack) OnRTCP(f func(rtcp.Packet)) {
+	peer.onRtcpMu.Lock()
+	defer peer.onRtcpMu.Unlock()
+
+	peer.onRtcp = f
+}
diff --git a/pkg/types/codec/codecs.go b/pkg/types/codec/codecs.go
index cc8b9bd7..d53a5509 100644
--- a/pkg/types/codec/codecs.go
+++ b/pkg/types/codec/codecs.go
@@ -6,6 +6,18 @@ import (
 	"github.com/pion/webrtc/v3"
 )
 
+var RTCPFeedback = []webrtc.RTCPFeedback{
+	{Type: webrtc.TypeRTCPFBTransportCC, Parameter: ""},
+	{Type: webrtc.TypeRTCPFBGoogREMB, Parameter: ""},
+
+	// https://www.iana.org/assignments/sdp-parameters/sdp-parameters.xhtml#sdp-parameters-19
+	{Type: webrtc.TypeRTCPFBCCM, Parameter: "fir"},
+
+	// https://www.iana.org/assignments/sdp-parameters/sdp-parameters.xhtml#sdp-parameters-15
+	{Type: webrtc.TypeRTCPFBNACK, Parameter: "pli"},
+	{Type: webrtc.TypeRTCPFBNACK, Parameter: ""},
+}
+
 func ParseRTC(codec webrtc.RTPCodecParameters) (RTPCodec, bool) {
 	codecName := strings.Split(codec.RTPCodecCapability.MimeType, "/")[1]
 	return ParseStr(codecName)
@@ -61,7 +73,7 @@ func VP8() RTPCodec {
 			ClockRate:    90000,
 			Channels:     0,
 			SDPFmtpLine:  "",
-			RTCPFeedback: []webrtc.RTCPFeedback{},
+			RTCPFeedback: RTCPFeedback,
 		},
 		// https://gstreamer.freedesktop.org/documentation/vpx/vp8enc.html
 		// gstreamer1.0-plugins-good
@@ -80,7 +92,7 @@ func VP9() RTPCodec {
 			ClockRate:    90000,
 			Channels:     0,
 			SDPFmtpLine:  "profile-id=0",
-			RTCPFeedback: []webrtc.RTCPFeedback{},
+			RTCPFeedback: RTCPFeedback,
 		},
 		// https://gstreamer.freedesktop.org/documentation/vpx/vp9enc.html
 		// gstreamer1.0-plugins-good
@@ -99,7 +111,7 @@ func H264() RTPCodec {
 			ClockRate:    90000,
 			Channels:     0,
 			SDPFmtpLine:  "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f",
-			RTCPFeedback: []webrtc.RTCPFeedback{},
+			RTCPFeedback: RTCPFeedback,
 		},
 		// https://gstreamer.freedesktop.org/documentation/x264/index.html
 		// gstreamer1.0-plugins-ugly