diff --git a/examples/rtp-to-webrtc/README.md b/examples/rtp-to-webrtc/README.md
index 9298e34d7b..e5acb03607 100644
--- a/examples/rtp-to-webrtc/README.md
+++ b/examples/rtp-to-webrtc/README.md
@@ -51,3 +51,9 @@ Copy the text that `rtp-to-webrtc` just emitted and copy into second text area
 A video should start playing in your browser above the input boxes.
 
 Congrats, you have used Pion WebRTC! Now start building something cool
+
+## Dealing with broken/lossy inputs
+Pion WebRTC also provides a [SampleBuilder](https://pkg.go.dev/github.com/pion/webrtc/v3@v3.0.4/pkg/media/samplebuilder). This consumes RTP packets and returns samples.
+It can be used to re-order and delay for lossy streams. You can see its usage in this example in [e4da70](https://github.com/pion/webrtc/commit/e4da709754639871266ca5e2cc2f945178caeb12).
+
+Currently it isn't working with H264, but is useful for VP8 and Opus. See [#1652](https://github.com/pion/webrtc/issues/1652) for the status of fixing for H264.
diff --git a/examples/rtp-to-webrtc/main.go b/examples/rtp-to-webrtc/main.go
index 71c3be4ce6..1c9fd9213a 100644
--- a/examples/rtp-to-webrtc/main.go
+++ b/examples/rtp-to-webrtc/main.go
@@ -6,11 +6,8 @@ import (
 	"fmt"
 	"net"
 
-	"github.com/pion/rtp"
-	"github.com/pion/rtp/codecs"
 	"github.com/pion/webrtc/v3"
 	"github.com/pion/webrtc/v3/examples/internal/signal"
-	"github.com/pion/webrtc/v3/pkg/media/samplebuilder"
 )
 
 func main() {
@@ -37,7 +34,7 @@ func main() {
 	}()
 
 	// Create a video track
-	videoTrack, err := webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: "video/vp8"}, "video", "pion")
+	videoTrack, err := webrtc.NewTrackLocalStaticRTP(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeVP8}, "video", "pion")
 	if err != nil {
 		panic(err)
 	}
@@ -95,32 +92,16 @@ func main() {
 	// Output the answer in base64 so we can paste it in browser
 	fmt.Println(signal.Encode(*peerConnection.LocalDescription()))
 
-	videoBuilder := samplebuilder.New(10, &codecs.VP8Packet{}, 90000)
-
 	// Read RTP packets forever and send them to the WebRTC Client
+	inboundRTPPacket := make([]byte, 1600) // UDP MTU
 	for {
-		inboundRTPPacket := make([]byte, 1500) // UDP MTU
-		packet := &rtp.Packet{}
-
 		n, _, err := listener.ReadFrom(inboundRTPPacket)
 		if err != nil {
 			panic(fmt.Sprintf("error during read: %s", err))
 		}
 
-		if err = packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
+		if _, err = videoTrack.Write(inboundRTPPacket[:n]); err != nil {
 			panic(err)
 		}
-
-		videoBuilder.Push(packet)
-		for {
-			sample := videoBuilder.Pop()
-			if sample == nil {
-				break
-			}
-
-			if writeErr := videoTrack.WriteSample(*sample); writeErr != nil {
-				panic(writeErr)
-			}
-		}
 	}
 }