diff --git a/stream/webrtc/webrtc.go b/stream/webrtc/webrtc.go
index 9893602..ff09fac 100644
--- a/stream/webrtc/webrtc.go
+++ b/stream/webrtc/webrtc.go
@@ -1,11 +1,15 @@
 package webrtc
 
 import (
+	"bufio"
 	"fmt"
+	"github.com/pion/rtp"
 	"io"
 	"log"
 	"math/rand"
+	"net"
 	"os"
+	"os/exec"
 	"time"
 
 	"github.com/pion/webrtc/v3"
@@ -30,7 +34,7 @@ type SessionDescription = webrtc.SessionDescription
 
 const (
 	audioFileName = "output.ogg"
-	videoFileName = "output.ivf"
+	videoFileName = "toto.ivf"
 )
 
 var (
@@ -244,16 +248,115 @@ func getPayloadType(m webrtc.MediaEngine, codecType webrtc.RTPCodecType, codecNa
 }
 
 func waitForPackets(inputChannel chan srt.Packet) {
+	// FIXME Clean code
+	var ffmpeg *exec.Cmd
+	var ffmpegInput io.WriteCloser
 	for {
 		var err error = nil
 		packet := <-inputChannel
 		switch packet.PacketType {
 		case "register":
 			log.Printf("WebRTC RegisterStream %s", packet.StreamName)
+
+			// Copied from https://github.com/pion/webrtc/blob/master/examples/rtp-to-webrtc/main.go
+
+			// Open a UDP Listener for RTP Packets on port 5004
+			videoListener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: 5004})
+			if err != nil {
+				panic(err)
+			}
+			audioListener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: 5005})
+			if err != nil {
+				panic(err)
+			}
+			defer func() {
+				if err = videoListener.Close(); err != nil {
+					panic(err)
+				}
+				if err = audioListener.Close(); err != nil {
+					panic(err)
+				}
+			}()
+
+			ffmpeg = exec.Command("ffmpeg", "-re", "-i", "pipe:0",
+				"-an", "-vcodec", "libvpx", //"-cpu-used", "5", "-deadline", "1", "-g", "10", "-error-resilient", "1", "-auto-alt-ref", "1",
+				"-f", "rtp", "rtp://127.0.0.1:5004",
+				"-vn", "-acodec", "libopus", //"-cpu-used", "5", "-deadline", "1", "-g", "10", "-error-resilient", "1", "-auto-alt-ref", "1",
+				"-f", "rtp", "rtp://127.0.0.1:5005")
+
+			fmt.Println("Waiting for RTP Packets, please run GStreamer or ffmpeg now")
+
+			input, err := ffmpeg.StdinPipe()
+			if err != nil {
+				panic(err)
+			}
+			ffmpegInput = input
+			errOutput, err := ffmpeg.StderrPipe()
+			if err != nil {
+				panic(err)
+			}
+
+			if err := ffmpeg.Start(); err != nil {
+				panic(err)
+			}
+
+			// Receive video
+			go func() {
+				for {
+					// Listen for a single RTP Packet, we need this to determine the SSRC
+					inboundRTPPacket := make([]byte, 1500) // UDP MTU
+					n, _, err := videoListener.ReadFromUDP(inboundRTPPacket)
+					if err != nil {
+						panic(err)
+					}
+					packet := &rtp.Packet{}
+					if err := packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
+						panic(err)
+					}
+					log.Printf("[Video] %s", packet)
+					for _, videoTrack := range videoTracks {
+						if writeErr := videoTrack.WriteRTP(packet); writeErr != nil {
+							panic(err)
+						}
+					}
+				}
+			}()
+
+			// Receive audio
+			go func() {
+				for {
+					// Listen for a single RTP Packet, we need this to determine the SSRC
+					inboundRTPPacket := make([]byte, 1500) // UDP MTU
+					n, _, err := audioListener.ReadFromUDP(inboundRTPPacket)
+					if err != nil {
+						panic(err)
+					}
+					packet := &rtp.Packet{}
+					if err := packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
+						panic(err)
+					}
+					log.Printf("[Audio] %s", packet)
+					for _, audioTrack := range audioTracks {
+						if writeErr := audioTrack.WriteRTP(packet); writeErr != nil {
+							panic(err)
+						}
+					}
+				}
+			}()
+
+			go func() {
+				scanner := bufio.NewScanner(errOutput)
+				for scanner.Scan() {
+					log.Printf("[WEBRTC FFMPEG %s] %s", "demo", scanner.Text())
+				}
+			}()
 			break
 		case "sendData":
-			log.Printf("WebRTC SendPacket %s", packet.StreamName)
-			// packet.Data
+			// log.Printf("WebRTC SendPacket %s", packet.StreamName)
+			_, err := ffmpegInput.Write(packet.Data)
+			if err != nil {
+				panic(err)
+			}
 			break
 		case "close":
 			log.Printf("WebRTC CloseConnection %s", packet.StreamName)
@@ -274,8 +377,8 @@ func Serve(remoteSdpChan, localSdpChan chan webrtc.SessionDescription, inputChan
 
 	// FIXME: use data from inputChannel
 	go waitForPackets(inputChannel)
-	go playVideo()
-	go playAudio()
+	// go playVideo()
+	// go playAudio()
 
 	// Handle new connections
 	for {