foo.patch

diff --git a/examples/gstreamer-send/gst/gst.go b/examples/gstreamer-send/gst/gst.go
index 2887df2..20feaaf 100644
--- a/examples/gstreamer-send/gst/gst.go
+++ b/examples/gstreamer-send/gst/gst.go
@@ -39,7 +39,7 @@ func CreatePipeline(codec webrtc.TrackType, in chan<- webrtc.RTCSample) *Pipelin
 	case webrtc.VP9:
 		pipelineStr = "videotestsrc ! vp9enc ! " + pipelineStr
 	case webrtc.H264:
-		pipelineStr = "videotestsrc ! x264enc speed-preset=veryfast key-int-max=60 ! video/x-h264,stream-format=byte-stream ! " + pipelineStr
+		pipelineStr = "videotestsrc ! x264enc speed-preset=veryfast key-int-max=2 aud=false ! video/x-h264,profile=baseline ! rtph264pay pt=100 ssrc=2147483647 config-interval=1 ! " + pipelineStr
 	case webrtc.Opus:
 		pipelineStr = "audiotestsrc ! opusenc ! " + pipelineStr
 	default:
diff --git a/examples/gstreamer-send/main.go b/examples/gstreamer-send/main.go
index 4d057c3..53fedb4 100644
--- a/examples/gstreamer-send/main.go
+++ b/examples/gstreamer-send/main.go
@@ -30,12 +30,6 @@ func main() {
 	// Create a new RTCPeerConnection
 	peerConnection := &webrtc.RTCPeerConnection{}
 
-	// Create a audio track
-	opusIn, err := peerConnection.AddTrack(webrtc.Opus, 48000)
-	if err != nil {
-		panic(err)
-	}
-
 	// Create a video track
 	vp8In, err := peerConnection.AddTrack(webrtc.H264, 90000)
 	if err != nil {
@@ -63,7 +57,6 @@ func main() {
 	fmt.Println(base64.StdEncoding.EncodeToString([]byte(localDescriptionStr)))
 
 	// Start pushing buffers on these tracks
-	gst.CreatePipeline(webrtc.Opus, opusIn).Start()
 	gst.CreatePipeline(webrtc.H264, vp8In).Start()
 	select {}
 }
diff --git a/internal/sdp/util.go b/internal/sdp/util.go
index 3d3508f..673a20d 100644
--- a/internal/sdp/util.go
+++ b/internal/sdp/util.go
@@ -65,6 +65,7 @@ func BaseSessionDescription(b *SessionBuilder) *SessionDescription {
 			"rtpmap:96 VP8/90000",
 			"rtpmap:98 VP9/90000",
 			"rtpmap:100 H264/90000",
+			"a=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f",
 		},
 	}
 
diff --git a/rtcpeerconnection.go b/rtcpeerconnection.go
index 74c6fd2..1e09be0 100644
--- a/rtcpeerconnection.go
+++ b/rtcpeerconnection.go
@@ -12,7 +12,6 @@ import (
 	"github.com/pions/webrtc/internal/util"
 	"github.com/pions/webrtc/pkg/ice"
 	"github.com/pions/webrtc/pkg/rtp"
-	"github.com/pions/webrtc/pkg/rtp/codecs"
 
 	"github.com/pkg/errors"
 )
@@ -136,34 +135,18 @@ func (r *RTCPeerConnection) AddTrack(mediaType TrackType, clockRate uint32) (sam
 
 	trackInput := make(chan RTCSample, 15)
 	go func() {
-		ssrc := rand.Uint32()
-		sdpTrack := &sdp.SessionBuilderTrack{SSRC: ssrc}
-		var payloader rtp.Payloader
-		var payloadType uint8
-		switch mediaType {
-		case Opus:
-			sdpTrack.IsAudio = true
-			payloader = &codecs.OpusPayloader{}
-			payloadType = 111
-
-		case VP8:
-			payloader = &codecs.VP8Payloader{}
-			payloadType = 96
-
-		case H264:
-			payloader = &codecs.H264Payloader{}
-			payloadType = 100
-		}
-
+		sdpTrack := &sdp.SessionBuilderTrack{SSRC: 2147483647}
 		r.localTracks = append(r.localTracks, sdpTrack)
-		packetizer := rtp.NewPacketizer(1500, payloadType, ssrc, payloader, rtp.NewRandomSequencer(), clockRate)
 		for {
+			var packet rtp.Packet
+
 			in := <-trackInput
-			packets := packetizer.Packetize(in.Data, in.Samples)
-			for _, p := range packets {
-				for _, port := range r.ports {
-					port.Send(p)
-				}
+			err := packet.Unmarshal(in.Data)
+			if err != nil {
+				panic(err)
+			}
+			for _, port := range r.ports {
+				port.Send(&packet)
 			}
 		}
 	}()