diff --git a/pkg/rtc/mediatrack.go b/pkg/rtc/mediatrack.go
index 21191a924..47113a07a 100644
--- a/pkg/rtc/mediatrack.go
+++ b/pkg/rtc/mediatrack.go
@@ -353,11 +353,16 @@ func (t *MediaTrack) AddReceiver(receiver *webrtc.RTPReceiver, track *webrtc.Tra
 		t.SetSimulcast(true)
 	}
 
+	var bitrates int
+	if len(ti.Layers) > int(layer) {
+		bitrates = int(ti.Layers[layer].GetBitrate())
+	}
+
 	if t.IsSimulcast() {
 		t.MediaTrackReceiver.SetLayerSsrc(mime, track.RID(), uint32(track.SSRC()))
 	}
 
-	buff.Bind(receiver.GetParameters(), track.Codec().RTPCodecCapability)
+	buff.Bind(receiver.GetParameters(), track.Codec().RTPCodecCapability, bitrates)
 
 	// if subscriber request fps before fps calculated, update them after fps updated.
 	buff.OnFpsChanged(func() {
diff --git a/pkg/sfu/buffer/buffer.go b/pkg/sfu/buffer/buffer.go
index e48692ba2..06492966a 100644
--- a/pkg/sfu/buffer/buffer.go
+++ b/pkg/sfu/buffer/buffer.go
@@ -194,7 +194,7 @@ func (b *Buffer) SetAudioLossProxying(enable bool) {
 	b.enableAudioLossProxying = enable
 }
 
-func (b *Buffer) Bind(params webrtc.RTPParameters, codec webrtc.RTPCodecCapability) {
+func (b *Buffer) Bind(params webrtc.RTPParameters, codec webrtc.RTPCodecCapability, bitrates int) {
 	b.Lock()
 	defer b.Unlock()
 	if b.bound {
@@ -264,6 +264,14 @@ func (b *Buffer) Bind(params webrtc.RTPParameters, codec webrtc.RTPCodecCapabili
 				}
 			}
 		}
+		if bitrates > 0 {
+			pps := bitrates / 8 / 1200
+			for pps > b.bucket.Capacity() {
+				if b.bucket.Grow() >= b.maxVideoPkts {
+					break
+				}
+			}
+		}
 
 	default:
 		b.codecType = webrtc.RTPCodecType(0)
diff --git a/pkg/sfu/buffer/buffer_test.go b/pkg/sfu/buffer/buffer_test.go
index d27ef3fe2..e97ad1db4 100644
--- a/pkg/sfu/buffer/buffer_test.go
+++ b/pkg/sfu/buffer/buffer_test.go
@@ -68,7 +68,7 @@ func TestNack(t *testing.T) {
 		buff.Bind(webrtc.RTPParameters{
 			HeaderExtensions: nil,
 			Codecs:           []webrtc.RTPCodecParameters{vp8Codec},
-		}, vp8Codec.RTPCodecCapability)
+		}, vp8Codec.RTPCodecCapability, 0)
 		rtt := uint32(20)
 		buff.nacker.SetRTT(rtt)
 		for i := 0; i < 15; i++ {
@@ -127,7 +127,7 @@ func TestNack(t *testing.T) {
 		buff.Bind(webrtc.RTPParameters{
 			HeaderExtensions: nil,
 			Codecs:           []webrtc.RTPCodecParameters{vp8Codec},
-		}, vp8Codec.RTPCodecCapability)
+		}, vp8Codec.RTPCodecCapability, 0)
 		rtt := uint32(30)
 		buff.nacker.SetRTT(rtt)
 		for i := 0; i < 15; i++ {
@@ -193,7 +193,7 @@ func TestNewBuffer(t *testing.T) {
 			buff.Bind(webrtc.RTPParameters{
 				HeaderExtensions: nil,
 				Codecs:           []webrtc.RTPCodecParameters{vp8Codec},
-			}, vp8Codec.RTPCodecCapability)
+			}, vp8Codec.RTPCodecCapability, 0)
 
 			for _, p := range TestPackets {
 				buf, _ := p.Marshal()
@@ -229,7 +229,7 @@ func TestFractionLostReport(t *testing.T) {
 	buff.Bind(webrtc.RTPParameters{
 		HeaderExtensions: nil,
 		Codecs:           []webrtc.RTPCodecParameters{opusCodec},
-	}, opusCodec.RTPCodecCapability)
+	}, opusCodec.RTPCodecCapability, 0)
 	for i := 0; i < 15; i++ {
 		pkt := rtp.Packet{
 			Header:  rtp.Header{SequenceNumber: uint16(i), Timestamp: uint32(i)},
@@ -261,7 +261,7 @@ func TestFractionLostReport(t *testing.T) {
 	buff.Bind(webrtc.RTPParameters{
 		HeaderExtensions: nil,
 		Codecs:           []webrtc.RTPCodecParameters{opusCodec},
-	}, opusCodec.RTPCodecCapability)
+	}, opusCodec.RTPCodecCapability, 0)
 	for i := 0; i < 15; i++ {
 		pkt := rtp.Packet{
 			Header:  rtp.Header{SequenceNumber: uint16(i), Timestamp: uint32(i)},