diff --git a/internal/capture/gst/gst.go b/internal/capture/gst/gst.go
index d41608a4..9527c257 100644
--- a/internal/capture/gst/gst.go
+++ b/internal/capture/gst/gst.go
@@ -8,19 +8,18 @@ package gst
 import "C"
 import (
 	"fmt"
+	"time"
 	"sync"
 	"unsafe"
 
-	"github.com/pion/webrtc/v2"
-
 	"demodesk/neko/internal/types"
+	"demodesk/neko/internal/types/codec"
 )
 
 // Pipeline is a wrapper for a GStreamer Pipeline
 type Pipeline struct {
 	Pipeline  *C.GstElement
 	Sample    chan types.Sample
-	ClockRate float32
 	Src       string
 	id        int
 }
@@ -30,9 +29,6 @@ var pipelinesLock sync.Mutex
 var registry *C.GstRegistry
 
 const (
-	videoClockRate = 90000
-	audioClockRate = 48000
-	pcmClockRate   = 8000
 	videoSrc       = "ximagesrc display-name=%s show-pointer=false use-damage=false ! video/x-raw ! videoconvert ! queue ! "
 	audioSrc       = "pulsesrc device=%s ! audio/x-raw,channels=2 ! audioconvert ! "
 	appSink        = " ! appsink name=appsink"
@@ -55,7 +51,7 @@ func CreateRTMPPipeline(pipelineDevice string, pipelineDisplay string, pipelineS
 		pipelineStr = fmt.Sprintf("flvmux name=mux ! rtmpsink location='%s live=1' %s audio/x-raw,channels=2 ! audioconvert ! voaacenc ! mux. %s x264enc bframes=0 key-int-max=60 byte-stream=true tune=zerolatency speed-preset=veryfast ! mux.", pipelineRTMP, audio, video)
 	}
 
-	return CreatePipeline(pipelineStr, 0)
+	return CreatePipeline(pipelineStr)
 }
 
 // CreateJPEGPipeline creates a GStreamer Pipeline
@@ -67,16 +63,15 @@ func CreateJPEGPipeline(pipelineDisplay string, pipelineSrc string, rate string,
 		pipelineStr = fmt.Sprintf("ximagesrc display-name=%s show-pointer=true use-damage=false ! videoconvert ! videoscale ! videorate ! video/x-raw,framerate=%s ! jpegenc quality=%s" + appSink, pipelineDisplay, rate, quality)
 	}
 
-	return CreatePipeline(pipelineStr, 0)
+	return CreatePipeline(pipelineStr)
 }
 
 // CreateAppPipeline creates a GStreamer Pipeline
-func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc string) (*Pipeline, error) {
-	var clockRate float32
+func CreateAppPipeline(codecRTP codec.RTP, pipelineDevice string, pipelineSrc string) (*Pipeline, error) {
 	var pipelineStr string
 
-	switch codecName {
-	case webrtc.VP8:
+	switch codecRTP.Name {
+	case codec.VP8:
 		// https://gstreamer.freedesktop.org/documentation/vpx/vp8enc.html?gi-language=c
 		// gstreamer1.0-plugins-good
 		// vp8enc error-resilient=partitions keyframe-max-dist=10 auto-alt-ref=true cpu-used=5 deadline=1
@@ -84,9 +79,8 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
 			return nil, err
 		}
 
-		clockRate = videoClockRate
 		pipelineStr = fmt.Sprintf(videoSrc + "vp8enc cpu-used=8 threads=2 deadline=1 error-resilient=partitions keyframe-max-dist=10 auto-alt-ref=true" + appSink, pipelineDevice)
-	case webrtc.VP9:
+	case codec.VP9:
 		// https://gstreamer.freedesktop.org/documentation/vpx/vp9enc.html?gi-language=c
 		// gstreamer1.0-plugins-good
 		// vp9enc
@@ -94,9 +88,8 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
 			return nil, err
 		}
 
-		clockRate = videoClockRate
 		pipelineStr = fmt.Sprintf(videoSrc + "vp9enc" + appSink, pipelineDevice)
-	case webrtc.H264:
+	case codec.H264:
 		if err := CheckPlugins([]string{"ximagesrc"}); err != nil {
 			return nil, err
 		}
@@ -106,7 +99,6 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
 		// gstreamer1.0-plugins-bad
 		// openh264enc multi-thread=4 complexity=high bitrate=3072000 max-bitrate=4096000
 		if err = CheckPlugins([]string{"openh264"}); err == nil {
-			clockRate = videoClockRate
 			pipelineStr = fmt.Sprintf(videoSrc + "openh264enc multi-thread=4 complexity=high bitrate=3072000 max-bitrate=4096000 ! video/x-h264,stream-format=byte-stream" + appSink, pipelineDevice)
 			break
 		}
@@ -115,13 +107,12 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
 		// gstreamer1.0-plugins-ugly
 		// video/x-raw,format=I420 ! x264enc bframes=0 key-int-max=60 byte-stream=true tune=zerolatency speed-preset=veryfast ! video/x-h264,stream-format=byte-stream
 		if err = CheckPlugins([]string{"x264"}); err == nil {
-			clockRate = videoClockRate
 			pipelineStr = fmt.Sprintf(videoSrc + "video/x-raw,format=I420 ! x264enc bframes=0 key-int-max=60 byte-stream=true tune=zerolatency speed-preset=veryfast ! video/x-h264,stream-format=byte-stream" + appSink, pipelineDevice)
 			break
 		}
 
 		return nil, err
-	case webrtc.Opus:
+	case codec.Opus:
 		// https://gstreamer.freedesktop.org/documentation/opus/opusenc.html
 		// gstreamer1.0-plugins-base
 		// opusenc
@@ -129,9 +120,8 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
 			return nil, err
 		}
 
-		clockRate = audioClockRate
 		pipelineStr = fmt.Sprintf(audioSrc + "opusenc" + appSink, pipelineDevice)
-	case webrtc.G722:
+	case codec.G722:
 		// https://gstreamer.freedesktop.org/documentation/libav/avenc_g722.html?gi-language=c
 		// gstreamer1.0-libav
 		// avenc_g722
@@ -139,9 +129,8 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
 			return nil, err
 		}
 
-		clockRate = audioClockRate
 		pipelineStr = fmt.Sprintf(audioSrc + "avenc_g722" + appSink, pipelineDevice)
-	case webrtc.PCMU:
+	case codec.PCMU:
 		// https://gstreamer.freedesktop.org/documentation/mulaw/mulawenc.html?gi-language=c
 		// gstreamer1.0-plugins-good
 		// audio/x-raw, rate=8000 ! mulawenc
@@ -149,9 +138,8 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
 			return nil, err
 		}
 
-		clockRate = pcmClockRate
 		pipelineStr = fmt.Sprintf(audioSrc + "audio/x-raw, rate=8000 ! mulawenc" + appSink, pipelineDevice)
-	case webrtc.PCMA:
+	case codec.PCMA:
 		// https://gstreamer.freedesktop.org/documentation/alaw/alawenc.html?gi-language=c
 		// gstreamer1.0-plugins-good
 		// audio/x-raw, rate=8000 ! alawenc
@@ -159,21 +147,20 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
 			return nil, err
 		}
 
-		clockRate = pcmClockRate
 		pipelineStr = fmt.Sprintf(audioSrc + "audio/x-raw, rate=8000 ! alawenc" + appSink, pipelineDevice)
 	default:
-		return nil, fmt.Errorf("unknown codec %s", codecName)
+		return nil, fmt.Errorf("unknown codec %s", codecRTP.Name)
 	}
 
 	if pipelineSrc != "" {
 		pipelineStr = fmt.Sprintf(pipelineSrc + appSink, pipelineDevice)
 	}
 
-	return CreatePipeline(pipelineStr, clockRate)
+	return CreatePipeline(pipelineStr)
 }
 
 // CreatePipeline creates a GStreamer Pipeline
-func CreatePipeline(pipelineStr string, clockRate float32) (*Pipeline, error) {
+func CreatePipeline(pipelineStr string) (*Pipeline, error) {
 	pipelineStrUnsafe := C.CString(pipelineStr)
 	defer C.free(unsafe.Pointer(pipelineStrUnsafe))
 
@@ -193,7 +180,6 @@ func CreatePipeline(pipelineStr string, clockRate float32) (*Pipeline, error) {
 	p := &Pipeline{
 		Pipeline:  gstPipeline,
 		Sample:    make(chan types.Sample),
-		ClockRate: clockRate,
 		Src:       pipelineStr,
 		id:        len(pipelines),
 	}
@@ -243,7 +229,7 @@ func goHandlePipelineBuffer(buffer unsafe.Pointer, bufferLen C.int, duration C.i
 	if ok {
 		pipeline.Sample <- types.Sample{
 			Data: C.GoBytes(buffer, bufferLen),
-			Samples: uint32(pipeline.ClockRate * (float32(duration) / 1e9)),
+			Duration: time.Duration(duration),
 		}
 	} else {
 		fmt.Printf("discarding buffer, no pipeline with id %d", int(pipelineID))
diff --git a/internal/capture/manager.go b/internal/capture/manager.go
index 46770ac2..2116b394 100644
--- a/internal/capture/manager.go
+++ b/internal/capture/manager.go
@@ -8,6 +8,7 @@ import (
 	"github.com/rs/zerolog/log"
 
 	"demodesk/neko/internal/types"
+	"demodesk/neko/internal/types/codec"
 	"demodesk/neko/internal/config"
 	"demodesk/neko/internal/capture/gst"
 )
@@ -125,11 +126,11 @@ func (manager *CaptureManagerCtx) Screencast() types.ScreencastManager {
 	return manager.screencast
 }
 
-func (manager *CaptureManagerCtx) VideoCodec() string {
+func (manager *CaptureManagerCtx) VideoCodec() codec.RTP {
 	return manager.config.VideoCodec
 }
 
-func (manager *CaptureManagerCtx) AudioCodec() string {
+func (manager *CaptureManagerCtx) AudioCodec() codec.RTP {
 	return manager.config.AudioCodec
 }
 
@@ -178,7 +179,7 @@ func (manager *CaptureManagerCtx) createVideoPipeline() {
 	var err error
 
 	manager.logger.Info().
-		Str("video_codec", manager.config.VideoCodec).
+		Str("video_codec", manager.config.VideoCodec.Name).
 		Str("video_display", manager.config.Display).
 		Str("video_params", manager.config.VideoParams).
 		Msgf("creating video pipeline")
@@ -212,7 +213,7 @@ func (manager *CaptureManagerCtx) createAudioPipeline() {
 	var err error
 
 	manager.logger.Info().
-		Str("audio_codec", manager.config.AudioCodec).
+		Str("audio_codec", manager.config.AudioCodec.Name).
 		Str("audio_display", manager.config.Device).
 		Str("audio_params", manager.config.AudioParams).
 		Msgf("creating audio pipeline")
diff --git a/internal/config/capture.go b/internal/config/capture.go
index 397367b4..fb072d02 100644
--- a/internal/config/capture.go
+++ b/internal/config/capture.go
@@ -1,17 +1,18 @@
 package config
 
 import (
-	"github.com/pion/webrtc/v2"
 	"github.com/spf13/cobra"
 	"github.com/spf13/viper"
+
+	"demodesk/neko/internal/types/codec"
 )
 
 type Capture struct {
 	Display      string
 	Device       string
-	AudioCodec   string
+	AudioCodec   codec.RTP
 	AudioParams  string
-	VideoCodec   string
+	VideoCodec   codec.RTP
 	VideoParams  string
 
 	BroadcastPipeline string
@@ -110,24 +111,30 @@ func (Capture) Init(cmd *cobra.Command) error {
 }
 
 func (s *Capture) Set() {
-	videoCodec := webrtc.VP8
+	var videoCodec codec.RTP
 	if viper.GetBool("vp8") {
-		videoCodec = webrtc.VP8
+		videoCodec = codec.New(codec.VP8)
 	} else if viper.GetBool("vp9") {
-		videoCodec = webrtc.VP9
+		videoCodec = codec.New(codec.VP9)
 	} else if viper.GetBool("h264") {
-		videoCodec = webrtc.H264
+		videoCodec = codec.New(codec.H264)
+	} else {
+		// default
+		videoCodec = codec.New(codec.VP8)
 	}
 
-	audioCodec := webrtc.Opus
+	var audioCodec codec.RTP
 	if viper.GetBool("opus") {
-		audioCodec = webrtc.Opus
+		audioCodec = codec.New(codec.Opus)
 	} else if viper.GetBool("g722") {
-		audioCodec = webrtc.G722
+		audioCodec = codec.New(codec.G722)
 	} else if viper.GetBool("pcmu") {
-		audioCodec = webrtc.PCMU
+		audioCodec = codec.New(codec.PCMU)
 	} else if viper.GetBool("pcma") {
-		audioCodec = webrtc.PCMA
+		audioCodec = codec.New(codec.PCMA)
+	} else {
+		// default
+		audioCodec = codec.New(codec.Opus)
 	}
 
 	s.Device = viper.GetString("device")
diff --git a/internal/config/webrtc.go b/internal/config/webrtc.go
index 13cbf185..faa23768 100644
--- a/internal/config/webrtc.go
+++ b/internal/config/webrtc.go
@@ -6,6 +6,7 @@ import (
 
 	"github.com/spf13/cobra"
 	"github.com/spf13/viper"
+
 	"demodesk/neko/internal/utils"
 )
 
diff --git a/internal/types/capture.go b/internal/types/capture.go
index 6481dfe4..7af5d01f 100644
--- a/internal/types/capture.go
+++ b/internal/types/capture.go
@@ -1,9 +1,12 @@
 package types
 
-type Sample struct {
-	Data    []byte
-	Samples uint32
-}
+import (
+	"github.com/pion/webrtc/v3/pkg/media"
+
+	"demodesk/neko/internal/types/codec"
+)
+
+type Sample media.Sample
 
 type BroadcastManager interface {
 	Start(url string) error
@@ -25,8 +28,8 @@ type CaptureManager interface {
 	Broadcast() BroadcastManager
 	Screencast() ScreencastManager
 
-	VideoCodec() string
-	AudioCodec() string
+	VideoCodec() codec.RTP
+	AudioCodec() codec.RTP
 
 	OnVideoFrame(listener func(sample Sample))
 	OnAudioFrame(listener func(sample Sample))
diff --git a/internal/types/codec/codecs.go b/internal/types/codec/codecs.go
new file mode 100644
index 00000000..4ffc4db7
--- /dev/null
+++ b/internal/types/codec/codecs.go
@@ -0,0 +1,113 @@
+package codec
+
+import "github.com/pion/webrtc/v3"
+
+const (
+	VP8 = "vp8"
+	VP9 = "vp9"
+	H264 = "h264"
+	Opus = "opus"
+	G722 = "g722"
+	PCMU = "pcmu"
+	PCMA = "pcma"
+)
+
+type RTP struct {
+	Name        string
+	PayloadType webrtc.PayloadType
+	Type        webrtc.RTPCodecType
+	Capability  webrtc.RTPCodecCapability
+}
+
+func New(codecType string) RTP {
+	codec := RTP{}
+
+	switch codecType {
+	case "vp8":
+		codec.Name = "vp8"
+		codec.PayloadType = 96
+		codec.Type = webrtc.RTPCodecTypeVideo
+		codec.Capability = webrtc.RTPCodecCapability{
+			MimeType: webrtc.MimeTypeVP8,
+			ClockRate: 90000,
+			Channels: 0,
+			SDPFmtpLine: "",
+			RTCPFeedback: nil,
+		}
+	case "vp9":
+		codec.Name = "vp9"
+		codec.PayloadType = 98
+		codec.Type = webrtc.RTPCodecTypeVideo
+		codec.Capability = webrtc.RTPCodecCapability{
+			MimeType: webrtc.MimeTypeVP9,
+			ClockRate: 90000,
+			Channels: 0,
+			SDPFmtpLine: "profile-id=0",
+			RTCPFeedback: nil,
+		}
+	case "h264":
+		codec.Name = "h264"
+		codec.PayloadType = 102
+		codec.Type = webrtc.RTPCodecTypeVideo
+		codec.Capability = webrtc.RTPCodecCapability{
+			MimeType: webrtc.MimeTypeH264,
+			ClockRate: 90000,
+			Channels: 0,
+			SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f",
+			RTCPFeedback: nil,
+		}
+	case "opus":
+		codec.Name = "opus"
+		codec.PayloadType = 111
+		codec.Type = webrtc.RTPCodecTypeAudio
+		codec.Capability = webrtc.RTPCodecCapability{
+			MimeType: webrtc.MimeTypeOpus,
+			ClockRate: 48000,
+			Channels: 2,
+			SDPFmtpLine: "",
+			RTCPFeedback: nil,
+		}
+	case "g722":
+		codec.Name = "g722"
+		codec.PayloadType = 9
+		codec.Type = webrtc.RTPCodecTypeAudio
+		codec.Capability = webrtc.RTPCodecCapability{
+			MimeType: webrtc.MimeTypeG722,
+			ClockRate: 8000,
+			Channels: 0,
+			SDPFmtpLine: "",
+			RTCPFeedback: nil,
+		}
+	case "pcmu":
+		codec.Name = "pcmu"
+		codec.PayloadType = 0
+		codec.Type = webrtc.RTPCodecTypeAudio
+		codec.Capability = webrtc.RTPCodecCapability{
+			MimeType: webrtc.MimeTypePCMU,
+			ClockRate: 8000,
+			Channels: 0,
+			SDPFmtpLine: "",
+			RTCPFeedback: nil,
+		}
+	case "pcma":
+		codec.Name = "pcma"
+		codec.PayloadType = 8
+		codec.Type = webrtc.RTPCodecTypeAudio
+		codec.Capability = webrtc.RTPCodecCapability{
+			MimeType: webrtc.MimeTypePCMA,
+			ClockRate: 8000,
+			Channels: 0,
+			SDPFmtpLine: "",
+			RTCPFeedback: nil,
+		}
+	}
+
+	return codec
+}
+
+func (codec *RTP) Register(engine *webrtc.MediaEngine) error {
+	return engine.RegisterCodec(webrtc.RTPCodecParameters{
+		RTPCodecCapability: codec.Capability,
+		PayloadType:        codec.PayloadType,
+	}, codec.Type)
+}
diff --git a/internal/webrtc/manager.go b/internal/webrtc/manager.go
index e0346383..fdfd491a 100644
--- a/internal/webrtc/manager.go
+++ b/internal/webrtc/manager.go
@@ -3,15 +3,15 @@ package webrtc
 import (
 	"fmt"
 	"io"
-	"math/rand"
 	"strings"
 
-	"github.com/pion/webrtc/v2"
-	"github.com/pion/webrtc/v2/pkg/media"
+	"github.com/pion/webrtc/v3"
+	"github.com/pion/webrtc/v3/pkg/media"
 	"github.com/rs/zerolog"
 	"github.com/rs/zerolog/log"
 
 	"demodesk/neko/internal/types"
+	"demodesk/neko/internal/types/codec"
 	"demodesk/neko/internal/config"
 )
 
@@ -26,10 +26,10 @@ func New(desktop types.DesktopManager, capture types.CaptureManager, config *con
 
 type WebRTCManagerCtx struct {
 	logger     zerolog.Logger
-	videoTrack *webrtc.Track
-	audioTrack *webrtc.Track
-	videoCodec *webrtc.RTPCodec
-	audioCodec *webrtc.RTPCodec
+	videoTrack *webrtc.TrackLocalStaticSample
+	audioTrack *webrtc.TrackLocalStaticSample
+	videoCodec codec.RTP
+	audioCodec codec.RTP
 	desktop    types.DesktopManager
 	capture    types.CaptureManager
 	config     *config.WebRTC
@@ -38,7 +38,9 @@ type WebRTCManagerCtx struct {
 func (manager *WebRTCManagerCtx) Start() {
 	var err error
 
-	manager.audioTrack, manager.audioCodec, err = manager.createTrack(manager.capture.AudioCodec())
+	// create audio track
+	manager.audioCodec = manager.capture.AudioCodec()
+	manager.audioTrack, err = webrtc.NewTrackLocalStaticSample(manager.audioCodec.Capability, "audio", "stream")
 	if err != nil {
 		manager.logger.Panic().Err(err).Msg("unable to create audio track")
 	}
@@ -49,7 +51,9 @@ func (manager *WebRTCManagerCtx) Start() {
 		}
 	})
 
-	manager.videoTrack, manager.videoCodec, err = manager.createTrack(manager.capture.VideoCodec())
+	// create video track
+	manager.videoCodec = manager.capture.VideoCodec()
+	manager.videoTrack, err = webrtc.NewTrackLocalStaticSample(manager.videoCodec.Capability, "video", "stream")
 	if err != nil {
 		manager.logger.Panic().Err(err).Msg("unable to create video track")
 	}
@@ -93,6 +97,7 @@ func (manager *WebRTCManagerCtx) CreatePeer(session types.Session) (string, bool
 		configuration = &webrtc.Configuration{
 			SDPSemantics: webrtc.SDPSemanticsUnifiedPlanWithFallback,
 		}
+
 		settings.SetLite(true)
 	}
 
@@ -104,10 +109,15 @@ func (manager *WebRTCManagerCtx) CreatePeer(session types.Session) (string, bool
 	settings.SetNAT1To1IPs(manager.config.NAT1To1IPs, webrtc.ICECandidateTypeHost)
 
 	// Create MediaEngine based off sdp
-	engine := webrtc.MediaEngine{}
+	engine := &webrtc.MediaEngine{}
 
-	engine.RegisterCodec(manager.audioCodec)
-	engine.RegisterCodec(manager.videoCodec)
+	if err := manager.videoCodec.Register(engine); err != nil {
+		return "", manager.config.ICELite, manager.config.ICEServers, err
+	}
+
+	if err := manager.audioCodec.Register(engine); err != nil {
+		return "", manager.config.ICELite, manager.config.ICEServers, err
+	}
 
 	// Create API with MediaEngine and SettingEngine
 	api := webrtc.NewAPI(webrtc.WithMediaEngine(engine), webrtc.WithSettingEngine(settings))
@@ -118,6 +128,11 @@ func (manager *WebRTCManagerCtx) CreatePeer(session types.Session) (string, bool
 		return "", manager.config.ICELite, manager.config.ICEServers, err
 	}
 
+	_, err = connection.CreateDataChannel("data", nil)
+	if err != nil {
+		return "", manager.config.ICELite, manager.config.ICEServers, err
+	}
+
 	if _, err = connection.AddTransceiverFromTrack(manager.videoTrack, webrtc.RtpTransceiverInit{
 		Direction: webrtc.RTPTransceiverDirectionSendonly,
 	}); err != nil {
@@ -130,27 +145,32 @@ func (manager *WebRTCManagerCtx) CreatePeer(session types.Session) (string, bool
 		return "", manager.config.ICELite, manager.config.ICEServers, err
 	}
 
-	description, err := connection.CreateOffer(nil)
+	offer, err := connection.CreateOffer(nil)
 	if err != nil {
 		return "", manager.config.ICELite, manager.config.ICEServers, err
 	}
 
-	connection.OnDataChannel(func(d *webrtc.DataChannel) {
-		d.OnMessage(func(msg webrtc.DataChannelMessage) {
+	connection.OnDataChannel(func(channel *webrtc.DataChannel) {
+		channel.OnMessage(func(message webrtc.DataChannelMessage) {
 			if !session.IsHost() {
 				return
 			}
 
-			if err = manager.handle(msg); err != nil {
+			if err = manager.handle(message); err != nil {
 				manager.logger.Warn().Err(err).Msg("data handle failed")
 			}
 		})
 	})
 
-	if err := connection.SetLocalDescription(description); err != nil {
+	// TODO: Refactor, send request to client.
+	gatherComplete := webrtc.GatheringCompletePromise(connection)
+
+	if err := connection.SetLocalDescription(offer); err != nil {
 		return "", manager.config.ICELite, manager.config.ICEServers, err
 	}
 
+	<-gatherComplete
+
 	connection.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
 		switch state {
 		case webrtc.PeerConnectionStateConnected:
@@ -170,40 +190,11 @@ func (manager *WebRTCManagerCtx) CreatePeer(session types.Session) (string, bool
 
 	session.SetWebRTCPeer(&WebRTCPeerCtx{
 		api:           api,
-		engine:        &engine,
+		engine:        engine,
 		settings:      &settings,
 		connection:    connection,
 		configuration: configuration,
 	})
 
-	return description.SDP, manager.config.ICELite, manager.config.ICEServers, nil
-}
-
-func (m *WebRTCManagerCtx) createTrack(codecName string) (*webrtc.Track, *webrtc.RTPCodec, error) {
-	var codec *webrtc.RTPCodec
-	switch codecName {
-	case webrtc.VP8:
-		codec = webrtc.NewRTPVP8Codec(webrtc.DefaultPayloadTypeVP8, 90000)
-	case webrtc.VP9:
-		codec = webrtc.NewRTPVP9Codec(webrtc.DefaultPayloadTypeVP9, 90000)
-	case webrtc.H264:
-		codec = webrtc.NewRTPH264Codec(webrtc.DefaultPayloadTypeH264, 90000)
-	case webrtc.Opus:
-		codec = webrtc.NewRTPOpusCodec(webrtc.DefaultPayloadTypeOpus, 48000)
-	case webrtc.G722:
-		codec = webrtc.NewRTPG722Codec(webrtc.DefaultPayloadTypeG722, 8000)
-	case webrtc.PCMU:
-		codec = webrtc.NewRTPPCMUCodec(webrtc.DefaultPayloadTypePCMU, 8000)
-	case webrtc.PCMA:
-		codec = webrtc.NewRTPPCMACodec(webrtc.DefaultPayloadTypePCMA, 8000)
-	default:
-		return nil, nil, fmt.Errorf("unknown codec %s", codecName)
-	}
-
-	track, err := webrtc.NewTrack(codec.PayloadType, rand.Uint32(), "stream", "stream", codec)
-	if err != nil {
-		return nil, nil, err
-	}
-
-	return track, codec, nil
+	return connection.LocalDescription().SDP, manager.config.ICELite, manager.config.ICEServers, nil
 }