diff --git a/cmd/webrtc/candidates.go b/cmd/webrtc/candidates.go
index 28b06908..1ce249ea 100644
--- a/cmd/webrtc/candidates.go
+++ b/cmd/webrtc/candidates.go
@@ -56,7 +56,7 @@ func GetCandidates() (candidates []string) {
 	return
 }
 
-func asyncCandidates(tr *api.Transport, cons *webrtc.Server) {
+func asyncCandidates(tr *api.Transport, cons *webrtc.Conn) {
 	tr.WithContext(func(ctx map[any]any) {
 		if candidates, ok := ctx["candidate"].([]string); ok {
 			// process candidates that receive before this moment
@@ -117,9 +117,9 @@ func candidateHandler(tr *api.Transport, msg *api.Message) error {
 		candidate := msg.String()
 		log.Trace().Str("candidate", candidate).Msg("[webrtc] remote")
 
-		if cons, ok := ctx["webrtc"].(*webrtc.Server); ok {
+		if cons, ok := ctx["webrtc"].(*webrtc.Conn); ok {
 			// if webrtc.Server already initialized - process candidate
-			cons.AddCandidate(candidate)
+			_ = cons.AddCandidate(candidate)
 		} else {
 			// or collect candidate and process it later
 			list, _ := ctx["candidate"].([]string)
diff --git a/cmd/webrtc/client.go b/cmd/webrtc/client.go
new file mode 100644
index 00000000..6f1b7e39
--- /dev/null
+++ b/cmd/webrtc/client.go
@@ -0,0 +1,154 @@
+package webrtc
+
+import (
+	"errors"
+	"github.com/AlexxIT/go2rtc/cmd/api"
+	"github.com/AlexxIT/go2rtc/pkg/streamer"
+	"github.com/AlexxIT/go2rtc/pkg/webrtc"
+	"github.com/gorilla/websocket"
+	pion "github.com/pion/webrtc/v3"
+	"io"
+	"net/http"
+	"strings"
+	"time"
+)
+
+func streamsHandler(url string) (streamer.Producer, error) {
+	url = url[7:]
+	if i := strings.Index(url, "://"); i > 0 {
+		switch url[:i] {
+		case "ws", "wss":
+			return asyncClient(url)
+		case "http", "https":
+			return syncClient(url)
+		}
+	}
+	return nil, errors.New("unsupported url: " + url)
+}
+
+func asyncClient(url string) (streamer.Producer, error) {
+	// 1. Connect to signalign server
+	ws, _, err := websocket.DefaultDialer.Dial(url, nil)
+	if err != nil {
+		return nil, err
+	}
+	defer func() {
+		if err != nil {
+			_ = ws.Close()
+		}
+	}()
+
+	// 2. Create PeerConnection
+	pc, err := newPeerConnection()
+	if err != nil {
+		log.Error().Err(err).Caller().Send()
+		return nil, err
+	}
+
+	prod := webrtc.NewConn(pc)
+	prod.Listen(func(msg any) {
+		switch msg := msg.(type) {
+		case pion.PeerConnectionState:
+			_ = ws.Close()
+
+		case *pion.ICECandidate:
+			if msg != nil {
+				s := msg.ToJSON().Candidate
+				log.Trace().Str("candidate", s).Msg("[webrtc] local")
+				_ = ws.WriteJSON(&api.Message{Type: "webrtc/candidate", Value: s})
+			}
+		}
+	})
+
+	// 3. Create offer
+	offer, err := prod.CreateOffer()
+	if err != nil {
+		return nil, err
+	}
+
+	// 4. Send offer
+	msg := &api.Message{Type: "webrtc/offer", Value: offer}
+	if err = ws.WriteJSON(msg); err != nil {
+		return nil, err
+	}
+
+	// 5. Get answer
+	if err = ws.ReadJSON(msg); err != nil {
+		return nil, err
+	}
+
+	if msg.Type != "webrtc/answer" {
+		return nil, errors.New("wrong answer: " + msg.Type)
+	}
+
+	answer := msg.String()
+	if err = prod.SetAnswer(answer); err != nil {
+		return nil, err
+	}
+
+	// 6. Continue to receiving candidates
+	go func() {
+		for {
+			// receive data from remote
+			msg := new(api.Message)
+			if err = ws.ReadJSON(msg); err != nil {
+				if cerr, ok := err.(*websocket.CloseError); ok {
+					log.Trace().Err(err).Caller().Msgf("[webrtc] ws code=%d", cerr)
+				}
+				break
+			}
+
+			switch msg.Type {
+			case "webrtc/candidate":
+				if msg.Value != nil {
+					_ = prod.AddCandidate(msg.String())
+				}
+			}
+		}
+
+		_ = ws.Close()
+	}()
+
+	return prod, nil
+}
+
+// syncClient - support WebRTC-HTTP Egress Protocol (WHEP)
+func syncClient(url string) (streamer.Producer, error) {
+	// 2. Create PeerConnection
+	pc, err := newPeerConnection()
+	if err != nil {
+		log.Error().Err(err).Caller().Send()
+		return nil, err
+	}
+
+	prod := webrtc.NewConn(pc)
+
+	// 3. Create offer
+	offer, err := prod.CreateCompleteOffer()
+	if err != nil {
+		return nil, err
+	}
+
+	req, err := http.NewRequest("POST", url, strings.NewReader(offer))
+	req.Header.Set("Content-Type", MimeSDP)
+	if err != nil {
+		return nil, err
+	}
+
+	client := http.Client{Timeout: time.Second * 5000}
+	res, err := client.Do(req)
+	if err != nil {
+		return nil, err
+	}
+
+	answer, err := io.ReadAll(res.Body)
+	if err != nil {
+		return nil, err
+	}
+
+	if err = prod.SetAnswer(string(answer)); err != nil {
+		return nil, err
+	}
+
+	return prod, nil
+}
diff --git a/cmd/webrtc/server.go b/cmd/webrtc/server.go
new file mode 100644
index 00000000..0f56ca4f
--- /dev/null
+++ b/cmd/webrtc/server.go
@@ -0,0 +1,196 @@
+package webrtc
+
+import (
+	"encoding/json"
+	"github.com/AlexxIT/go2rtc/cmd/streams"
+	"github.com/AlexxIT/go2rtc/pkg/webrtc"
+	pion "github.com/pion/webrtc/v3"
+	"io"
+	"net/http"
+	"strconv"
+	"strings"
+	"time"
+)
+
+const MimeSDP = "application/sdp"
+
+var sessions = map[string]*webrtc.Conn{}
+
+func syncHandler(w http.ResponseWriter, r *http.Request) {
+	switch r.Method {
+	case "POST":
+		query := r.URL.Query()
+		if query.Get("src") != "" {
+			// WHEP or JSON SDP or raw SDP exchange
+			outputWebRTC(w, r)
+		} else if query.Get("dst") != "" {
+			// WHIP SDP exchange
+			inputWebRTC(w, r)
+		} else {
+			http.Error(w, "", http.StatusBadRequest)
+		}
+
+	case "PATCH":
+		// TODO: WHEP/WHIP
+		http.Error(w, "", http.StatusMethodNotAllowed)
+
+	case "DELETE":
+		if id := r.URL.Query().Get("id"); id != "" {
+			if conn, ok := sessions[id]; ok {
+				delete(sessions, id)
+				_ = conn.Close()
+			} else {
+				http.Error(w, "", http.StatusNotFound)
+			}
+		} else {
+			http.Error(w, "", http.StatusBadRequest)
+		}
+
+	default:
+		http.Error(w, "", http.StatusMethodNotAllowed)
+	}
+}
+
+// outputWebRTC support API depending on Content-Type:
+// 1. application/json - receive {"type":"offer","sdp":"v=0\r\n..."} and response {"type":"answer","sdp":"v=0\r\n..."}
+// 2. application/sdp - receive/response SDP via WebRTC-HTTP Egress Protocol (WHEP)
+// 3. other - receive/response raw SDP
+func outputWebRTC(w http.ResponseWriter, r *http.Request) {
+	url := r.URL.Query().Get("src")
+	stream := streams.Get(url)
+	if stream == nil {
+		return
+	}
+
+	mediaType := r.Header.Get("Content-Type")
+	if mediaType != "" {
+		mediaType, _, _ = strings.Cut(mediaType, ";")
+		mediaType = strings.ToLower(strings.TrimSpace(mediaType))
+	}
+
+	var offer string
+
+	switch mediaType {
+	case "application/json":
+		var desc pion.SessionDescription
+		if err := json.NewDecoder(r.Body).Decode(&desc); err != nil {
+			log.Error().Err(err).Caller().Send()
+			http.Error(w, err.Error(), http.StatusBadRequest)
+			return
+		}
+		offer = desc.SDP
+
+	default:
+		body, err := io.ReadAll(r.Body)
+		if err != nil {
+			log.Error().Err(err).Caller().Send()
+			http.Error(w, err.Error(), http.StatusInternalServerError)
+			return
+		}
+		offer = string(body)
+	}
+
+	answer, err := ExchangeSDP(stream, offer, r.UserAgent())
+	if err != nil {
+		log.Error().Err(err).Caller().Send()
+		http.Error(w, err.Error(), http.StatusInternalServerError)
+		return
+	}
+
+	switch mediaType {
+	case "application/json":
+		w.Header().Set("Content-Type", mediaType)
+
+		v := pion.SessionDescription{
+			Type: pion.SDPTypeAnswer, SDP: answer,
+		}
+		err = json.NewEncoder(w).Encode(v)
+
+	case MimeSDP:
+		w.Header().Set("Content-Type", mediaType)
+		w.WriteHeader(http.StatusCreated)
+
+		_, err = w.Write([]byte(answer))
+
+	default:
+		_, err = w.Write([]byte(answer))
+	}
+
+	if err != nil {
+		log.Error().Err(err).Caller().Send()
+		http.Error(w, err.Error(), http.StatusInternalServerError)
+	}
+}
+
+func inputWebRTC(w http.ResponseWriter, r *http.Request) {
+	dst := r.URL.Query().Get("dst")
+	stream := streams.Get(dst)
+	if stream == nil {
+		stream = streams.New(dst, nil)
+	}
+
+	// 1. Get offer
+	offer, err := io.ReadAll(r.Body)
+	if err != nil {
+		log.Error().Err(err).Caller().Send()
+		http.Error(w, err.Error(), http.StatusInternalServerError)
+		return
+	}
+
+	log.Trace().Msgf("[webrtc] WHIP offer\n%s", offer)
+
+	pc, err := newPeerConnection()
+	if err != nil {
+		log.Error().Err(err).Caller().Send()
+		http.Error(w, err.Error(), http.StatusInternalServerError)
+		return
+	}
+
+	// create new webrtc instance
+	conn := webrtc.NewConn(pc)
+	conn.UserAgent = r.UserAgent()
+
+	if err = conn.SetOffer(string(offer)); err != nil {
+		log.Warn().Err(err).Caller().Send()
+		http.Error(w, err.Error(), http.StatusInternalServerError)
+		return
+	}
+
+	answer, err := conn.GetCompleteAnswer()
+	if err == nil {
+		answer, err = syncCanditates(answer)
+	}
+	if err != nil {
+		log.Warn().Err(err).Caller().Send()
+		http.Error(w, err.Error(), http.StatusInternalServerError)
+		return
+	}
+
+	log.Trace().Msgf("[webrtc] WHIP answer\n%s", answer)
+
+	id := strconv.FormatInt(time.Now().UnixNano(), 36)
+	sessions[id] = conn
+
+	conn.Listen(func(msg interface{}) {
+		switch msg := msg.(type) {
+		case pion.PeerConnectionState:
+			if msg == pion.PeerConnectionStateClosed {
+				stream.RemoveProducer(conn)
+				if _, ok := sessions[id]; ok {
+					delete(sessions, id)
+				}
+			}
+		}
+	})
+
+	stream.AddProducer(conn)
+
+	w.Header().Set("Content-Type", MimeSDP)
+	w.Header().Set("Location", "webrtc?id="+id)
+	w.WriteHeader(http.StatusCreated)
+
+	if _, err = w.Write([]byte(answer)); err != nil {
+		log.Warn().Err(err).Caller().Send()
+		return
+	}
+}
diff --git a/cmd/webrtc/webrtc.go b/cmd/webrtc/webrtc.go
index 8dd340fb..dd41a2a9 100644
--- a/cmd/webrtc/webrtc.go
+++ b/cmd/webrtc/webrtc.go
@@ -1,7 +1,6 @@
 package webrtc
 
 import (
-	"encoding/json"
 	"errors"
 	"github.com/AlexxIT/go2rtc/cmd/api"
 	"github.com/AlexxIT/go2rtc/cmd/app"
@@ -9,10 +8,7 @@ import (
 	"github.com/AlexxIT/go2rtc/pkg/webrtc"
 	pion "github.com/pion/webrtc/v3"
 	"github.com/rs/zerolog"
-	"io"
-	"mime"
 	"net"
-	"net/http"
 )
 
 func Init() {
@@ -62,11 +58,16 @@ func Init() {
 		AddCandidate(candidate)
 	}
 
+	// async WebRTC server (two API versions)
 	api.HandleWS("webrtc", asyncHandler)
 	api.HandleWS("webrtc/offer", asyncHandler)
 	api.HandleWS("webrtc/candidate", candidateHandler)
 
+	// sync WebRTC server (two API versions)
 	api.HandleFunc("api/webrtc", syncHandler)
+
+	// WebRTC client
+	streams.HandleFunc("webrtc", streamsHandler)
 }
 
 var Port string
@@ -90,7 +91,7 @@ func asyncHandler(tr *api.Transport, msg *api.Message) error {
 		return err
 	}
 
-	cons := webrtc.NewServer(pc)
+	cons := webrtc.NewConn(pc)
 	cons.UserAgent = tr.Request.UserAgent()
 	cons.Listen(func(msg any) {
 		switch msg := msg.(type) {
@@ -153,65 +154,6 @@ func asyncHandler(tr *api.Transport, msg *api.Message) error {
 	return nil
 }
 
-func syncHandler(w http.ResponseWriter, r *http.Request) {
-	url := r.URL.Query().Get("src")
-	stream := streams.Get(url)
-	if stream == nil {
-		return
-	}
-
-	ct := r.Header.Get("Content-Type")
-	if ct != "" {
-		ct, _, _ = mime.ParseMediaType(ct)
-	}
-
-	// V2 - json/object exchange, V1 - raw SDP exchange
-	apiV2 := ct == "application/json"
-
-	var offer string
-	if apiV2 {
-		var desc pion.SessionDescription
-		if err := json.NewDecoder(r.Body).Decode(&desc); err != nil {
-			log.Error().Err(err).Caller().Send()
-			http.Error(w, err.Error(), http.StatusBadRequest)
-			return
-		}
-		offer = desc.SDP
-	} else {
-		body, err := io.ReadAll(r.Body)
-		if err != nil {
-			log.Error().Err(err).Caller().Send()
-			http.Error(w, err.Error(), http.StatusInternalServerError)
-			return
-		}
-		offer = string(body)
-	}
-
-	answer, err := ExchangeSDP(stream, offer, r.UserAgent())
-	if err != nil {
-		log.Error().Err(err).Caller().Send()
-		http.Error(w, err.Error(), http.StatusInternalServerError)
-		return
-	}
-
-	// send SDP to client
-	if apiV2 {
-		w.Header().Set("Content-Type", ct)
-
-		v := pion.SessionDescription{
-			Type: pion.SDPTypeAnswer, SDP: answer,
-		}
-		if err = json.NewEncoder(w).Encode(v); err != nil {
-			log.Error().Err(err).Caller().Send()
-			http.Error(w, err.Error(), http.StatusInternalServerError)
-		}
-	} else {
-		if _, err = w.Write([]byte(answer)); err != nil {
-			log.Error().Err(err).Caller().Send()
-		}
-	}
-}
-
 func ExchangeSDP(stream *streams.Stream, offer string, userAgent string) (answer string, err error) {
 	pc, err := newPeerConnection()
 	if err != nil {
@@ -220,7 +162,7 @@ func ExchangeSDP(stream *streams.Stream, offer string, userAgent string) (answer
 	}
 
 	// create new webrtc instance
-	conn := webrtc.NewServer(pc)
+	conn := webrtc.NewConn(pc)
 	conn.UserAgent = userAgent
 	conn.Listen(func(msg interface{}) {
 		switch msg := msg.(type) {
diff --git a/pkg/webrtc/client.go b/pkg/webrtc/client.go
new file mode 100644
index 00000000..946ae3f6
--- /dev/null
+++ b/pkg/webrtc/client.go
@@ -0,0 +1,34 @@
+package webrtc
+
+import "github.com/pion/webrtc/v3"
+
+func (c *Conn) CreateOffer() (string, error) {
+	init := webrtc.RTPTransceiverInit{Direction: webrtc.RTPTransceiverDirectionRecvonly}
+	_, _ = c.pc.AddTransceiverFromKind(webrtc.RTPCodecTypeVideo, init)
+	_, _ = c.pc.AddTransceiverFromKind(webrtc.RTPCodecTypeAudio, init)
+
+	desc, err := c.pc.CreateOffer(nil)
+	if err != nil {
+		return "", err
+	}
+
+	if err = c.pc.SetLocalDescription(desc); err != nil {
+		return "", err
+	}
+
+	return desc.SDP, nil
+}
+
+func (c *Conn) CreateCompleteOffer() (string, error) {
+	if _, err := c.CreateOffer(); err != nil {
+		return "", err
+	}
+
+	<-webrtc.GatheringCompletePromise(c.pc)
+	return c.pc.LocalDescription().SDP, nil
+}
+
+func (c *Conn) SetAnswer(answer string) (err error) {
+	desc := webrtc.SessionDescription{SDP: answer, Type: webrtc.SDPTypeAnswer}
+	return c.pc.SetRemoteDescription(desc)
+}
diff --git a/pkg/webrtc/conn.go b/pkg/webrtc/conn.go
new file mode 100644
index 00000000..4473f2f9
--- /dev/null
+++ b/pkg/webrtc/conn.go
@@ -0,0 +1,148 @@
+package webrtc
+
+import (
+	"github.com/AlexxIT/go2rtc/pkg/streamer"
+	"github.com/pion/webrtc/v3"
+)
+
+type Conn struct {
+	streamer.Element
+
+	UserAgent string
+
+	pc *webrtc.PeerConnection
+
+	medias []*streamer.Media
+	tracks []*streamer.Track
+
+	receive int
+	send    int
+
+	offer string
+}
+
+func NewConn(pc *webrtc.PeerConnection) *Conn {
+	c := &Conn{pc: pc}
+
+	pc.OnICECandidate(func(candidate *webrtc.ICECandidate) {
+		c.Fire(candidate)
+	})
+
+	pc.OnDataChannel(func(channel *webrtc.DataChannel) {
+		c.Fire(channel)
+	})
+
+	pc.OnTrack(func(remote *webrtc.TrackRemote, _ *webrtc.RTPReceiver) {
+		track := c.getTrack(remote)
+		if track == nil {
+			println("ERROR: webrtc: can't find track")
+			return
+		}
+
+		for {
+			packet, _, err := remote.ReadRTP()
+			if err != nil {
+				return
+			}
+			if len(packet.Payload) == 0 {
+				continue
+			}
+			c.receive += len(packet.Payload)
+			_ = track.WriteRTP(packet)
+		}
+	})
+
+	// OK connection:
+	// 15:01:46 ICE connection state changed: checking
+	// 15:01:46 peer connection state changed: connected
+	// 15:01:54 peer connection state changed: disconnected
+	// 15:02:20 peer connection state changed: failed
+	//
+	// Fail connection:
+	// 14:53:08 ICE connection state changed: checking
+	// 14:53:39 peer connection state changed: failed
+	pc.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
+		c.Fire(state)
+
+		// TODO: rewrite?
+		switch state {
+		case webrtc.PeerConnectionStateDisconnected:
+			// disconnect event comes earlier, than failed
+			// but it comes only for success connections
+			_ = pc.Close()
+		case webrtc.PeerConnectionStateFailed:
+			_ = pc.Close()
+		}
+	})
+
+	return c
+}
+
+func (c *Conn) Close() error {
+	return c.pc.Close()
+}
+
+func (c *Conn) AddCandidate(candidate string) error {
+	// pion uses only candidate value from json/object candidate struct
+	return c.pc.AddICECandidate(webrtc.ICECandidateInit{Candidate: candidate})
+}
+
+func (c *Conn) getTrack(remote *webrtc.TrackRemote) *streamer.Track {
+	payloadType := uint8(remote.PayloadType())
+
+	// search existing track (two way audio)
+	for _, track := range c.tracks {
+		if track.Codec.PayloadType == payloadType {
+			return track
+		}
+	}
+
+	// create new track (incoming WebRTC WHIP)
+	for _, media := range c.medias {
+		for _, codec := range media.Codecs {
+			if codec.PayloadType == payloadType {
+				track := streamer.NewTrack(codec, media.Direction)
+				c.tracks = append(c.tracks, track)
+				return track
+			}
+		}
+	}
+
+	return nil
+}
+
+func (c *Conn) remote() string {
+	if c.pc == nil {
+		return ""
+	}
+
+	for _, trans := range c.pc.GetTransceivers() {
+		if trans == nil {
+			continue
+		}
+
+		receiver := trans.Receiver()
+		if receiver == nil {
+			continue
+		}
+
+		transport := receiver.Transport()
+		if transport == nil {
+			continue
+		}
+
+		iceTransport := transport.ICETransport()
+		if iceTransport == nil {
+			continue
+		}
+
+		pair, _ := iceTransport.GetSelectedCandidatePair()
+		if pair == nil || pair.Remote == nil {
+			continue
+		}
+
+		return pair.Remote.String()
+	}
+
+	return ""
+}
diff --git a/pkg/webrtc/consumer.go b/pkg/webrtc/consumer.go
index f013274d..828a03f1 100644
--- a/pkg/webrtc/consumer.go
+++ b/pkg/webrtc/consumer.go
@@ -9,11 +9,11 @@ import (
 	"github.com/pion/webrtc/v3"
 )
 
-func (c *Server) GetMedias() []*streamer.Media {
+func (c *Conn) GetMedias() []*streamer.Media {
 	return c.medias
 }
 
-func (c *Server) AddTrack(media *streamer.Media, track *streamer.Track) *streamer.Track {
+func (c *Conn) AddTrack(media *streamer.Media, track *streamer.Track) *streamer.Track {
 	switch track.Direction {
 	// send our track to WebRTC consumer
 	case streamer.DirectionSendonly:
@@ -41,7 +41,14 @@ func (c *Server) AddTrack(media *streamer.Media, track *streamer.Track) *streame
 			return nil
 		}
 
-		if _, err = c.conn.AddTrack(trackLocal); err != nil {
+		init := webrtc.RTPTransceiverInit{Direction: webrtc.RTPTransceiverDirectionSendonly}
+		tr, err := c.pc.AddTransceiverFromTrack(trackLocal, init)
+		if err != nil {
+			return nil
+		}
+
+		codecs := []webrtc.RTPCodecParameters{{RTPCodecCapability: caps}}
+		if err = tr.SetCodecPreferences(codecs); err != nil {
 			return nil
 		}
 
@@ -78,37 +85,39 @@ func (c *Server) AddTrack(media *streamer.Media, track *streamer.Track) *streame
 
 	// receive track from WebRTC consumer (microphone, backchannel, two way audio)
 	case streamer.DirectionRecvonly:
-		for _, tr := range c.conn.GetTransceivers() {
-			if tr.Mid() != media.MID {
-				continue
-			}
-
-			codec := track.Codec
-			caps := webrtc.RTPCodecCapability{
-				MimeType:  MimeType(codec),
-				ClockRate: codec.ClockRate,
-				Channels:  codec.Channels,
-			}
-			codecs := []webrtc.RTPCodecParameters{
-				{RTPCodecCapability: caps},
-			}
-			if err := tr.SetCodecPreferences(codecs); err != nil {
-				return nil
-			}
-
-			c.tracks = append(c.tracks, track)
-			return track
+		caps := webrtc.RTPCodecCapability{
+			MimeType:  MimeType(track.Codec),
+			ClockRate: track.Codec.ClockRate,
+			Channels:  track.Codec.Channels,
 		}
+
+		init := webrtc.RTPTransceiverInit{Direction: webrtc.RTPTransceiverDirectionRecvonly}
+		tr, err := c.pc.AddTransceiverFromKind(webrtc.RTPCodecTypeAudio, init)
+		if err != nil {
+			return nil
+		}
+
+		codecs := []webrtc.RTPCodecParameters{
+			{RTPCodecCapability: caps, PayloadType: webrtc.PayloadType(track.Codec.PayloadType)},
+		}
+		if err = tr.SetCodecPreferences(codecs); err != nil {
+			return nil
+		}
+
+		c.tracks = append(c.tracks, track)
+		return track
 	}
 
 	panic("wrong direction")
 }
 
-func (c *Server) MarshalJSON() ([]byte, error) {
+func (c *Conn) MarshalJSON() ([]byte, error) {
 	info := &streamer.Info{
-		Type:       "WebRTC client",
+		Type:       "WebRTC",
 		RemoteAddr: c.remote(),
 		UserAgent:  c.UserAgent,
+		Medias:     c.medias,
+		Tracks:     c.tracks,
 		Recv:       uint32(c.receive),
 		Send:       uint32(c.send),
 	}
diff --git a/pkg/webrtc/producer.go b/pkg/webrtc/producer.go
new file mode 100644
index 00000000..c0d0574d
--- /dev/null
+++ b/pkg/webrtc/producer.go
@@ -0,0 +1,20 @@
+package webrtc
+
+import "github.com/AlexxIT/go2rtc/pkg/streamer"
+
+func (c *Conn) GetTrack(media *streamer.Media, codec *streamer.Codec) *streamer.Track {
+	for _, track := range c.tracks {
+		if track.Codec == codec {
+			return track
+		}
+	}
+	return nil
+}
+
+func (c *Conn) Start() error {
+	return nil
+}
+
+func (c *Conn) Stop() error {
+	return c.pc.Close()
+}
diff --git a/pkg/webrtc/server.go b/pkg/webrtc/server.go
index 098c1f53..e1586e44 100644
--- a/pkg/webrtc/server.go
+++ b/pkg/webrtc/server.go
@@ -6,96 +6,11 @@ import (
 	"github.com/pion/webrtc/v3"
 )
 
-type Server struct {
-	streamer.Element
+func (c *Conn) SetOffer(offer string) (err error) {
+	c.offer = offer
 
-	UserAgent string
-
-	conn *webrtc.PeerConnection
-
-	medias []*streamer.Media
-	tracks []*streamer.Track
-
-	receive int
-	send    int
-}
-
-func NewServer(conn *webrtc.PeerConnection) *Server {
-	c := &Server{conn: conn}
-
-	conn.OnICECandidate(func(candidate *webrtc.ICECandidate) {
-		c.Fire(candidate)
-	})
-
-	conn.OnTrack(func(remote *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) {
-		for _, track := range c.tracks {
-			if track.Direction != streamer.DirectionRecvonly {
-				continue
-			}
-			if track.Codec.PayloadType != uint8(remote.PayloadType()) {
-				continue
-			}
-
-			for {
-				packet, _, err := remote.ReadRTP()
-				if err != nil {
-					return
-				}
-				if len(packet.Payload) == 0 {
-					continue
-				}
-				c.receive += len(packet.Payload)
-				_ = track.WriteRTP(packet)
-			}
-		}
-
-		//fmt.Printf("TODO: webrtc ontrack %+v\n", remote)
-	})
-
-	conn.OnDataChannel(func(channel *webrtc.DataChannel) {
-		c.Fire(channel)
-	})
-
-	// OK connection:
-	// 15:01:46 ICE connection state changed: checking
-	// 15:01:46 peer connection state changed: connected
-	// 15:01:54 peer connection state changed: disconnected
-	// 15:02:20 peer connection state changed: failed
-	//
-	// Fail connection:
-	// 14:53:08 ICE connection state changed: checking
-	// 14:53:39 peer connection state changed: failed
-	conn.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
-		c.Fire(state)
-
-		// TODO: remove
-		switch state {
-		case webrtc.PeerConnectionStateConnected:
-			c.Fire(streamer.StatePlaying) // TODO: remove
-		case webrtc.PeerConnectionStateDisconnected:
-			c.Fire(streamer.StateNull) // TODO: remove
-			// disconnect event comes earlier, than failed
-			// but it comes only for success connections
-			_ = conn.Close()
-		case webrtc.PeerConnectionStateFailed:
-			_ = conn.Close()
-		}
-	})
-
-	return c
-}
-
-func (c *Server) SetOffer(offer string) (err error) {
-	desc := webrtc.SessionDescription{
-		Type: webrtc.SDPTypeOffer, SDP: offer,
-	}
-	if err = c.conn.SetRemoteDescription(desc); err != nil {
-		return
-	}
-
-	rawSDP := []byte(c.conn.RemoteDescription().SDP)
 	sd := &sdp.SessionDescription{}
-	if err = sd.Unmarshal(rawSDP); err != nil {
+	if err = sd.Unmarshal([]byte(offer)); err != nil {
 		return
 	}
 
@@ -117,85 +32,39 @@ func (c *Server) SetOffer(offer string) (err error) {
 	return
 }
 
-func (c *Server) GetAnswer() (answer string, err error) {
-	for _, tr := range c.conn.GetTransceivers() {
-		if tr.Direction() != webrtc.RTPTransceiverDirectionSendonly {
-			continue
-		}
+func (c *Conn) GetAnswer() (answer string, err error) {
+	// we need to process remote offer after we create transeivers
+	desc := webrtc.SessionDescription{Type: webrtc.SDPTypeOffer, SDP: c.offer}
+	if err = c.pc.SetRemoteDescription(desc); err != nil {
+		return "", err
+	}
 
-		// disable transceivers if we don't have track
-		// make direction=inactive
-		// don't really necessary, but anyway
-		if tr.Sender() == nil {
+	// disable transceivers if we don't have track
+	// make direction=inactive
+	// don't really necessary, but anyway
+	for _, tr := range c.pc.GetTransceivers() {
+		if tr.Direction() == webrtc.RTPTransceiverDirectionSendonly && tr.Sender() == nil {
 			if err = tr.Stop(); err != nil {
 				return
 			}
 		}
 	}
 
-	var sdAnswer webrtc.SessionDescription
-	sdAnswer, err = c.conn.CreateAnswer(nil)
-	if err != nil {
+	if desc, err = c.pc.CreateAnswer(nil); err != nil {
+		return
+	}
+	if err = c.pc.SetLocalDescription(desc); err != nil {
 		return
 	}
 
-	if err = c.conn.SetLocalDescription(sdAnswer); err != nil {
-		return
-	}
-
-	return sdAnswer.SDP, nil
+	return desc.SDP, nil
 }
 
-func (c *Server) GetCompleteAnswer() (answer string, err error) {
+func (c *Conn) GetCompleteAnswer() (answer string, err error) {
 	if _, err = c.GetAnswer(); err != nil {
 		return
 	}
 
-	<-webrtc.GatheringCompletePromise(c.conn)
-	return c.conn.LocalDescription().SDP, nil
-}
-
-func (c *Server) Close() error {
-	return c.conn.Close()
-}
-
-func (c *Server) AddCandidate(candidate string) {
-	// pion uses only candidate value from json/object candidate struct
-	_ = c.conn.AddICECandidate(webrtc.ICECandidateInit{Candidate: candidate})
-}
-
-func (c *Server) remote() string {
-	if c.conn == nil {
-		return ""
-	}
-
-	for _, trans := range c.conn.GetTransceivers() {
-		if trans == nil {
-			continue
-		}
-
-		receiver := trans.Receiver()
-		if receiver == nil {
-			continue
-		}
-
-		transport := receiver.Transport()
-		if transport == nil {
-			continue
-		}
-
-		iceTransport := transport.ICETransport()
-		if iceTransport == nil {
-			continue
-		}
-
-		pair, _ := iceTransport.GetSelectedCandidatePair()
-		if pair == nil || pair.Remote == nil {
-			continue
-		}
-
-		return pair.Remote.String()
-	}
-
-	return ""
+	<-webrtc.GatheringCompletePromise(c.pc)
+	return c.pc.LocalDescription().SDP, nil
 }
diff --git a/www/webrtc.html b/www/webrtc.html
index ca6bbab0..39954bdc 100644
--- a/www/webrtc.html
+++ b/www/webrtc.html
@@ -37,7 +37,6 @@
 
             pc.createOffer().then(offer => {
                 pc.setLocalDescription(offer).then(() => {
-                    console.log(offer.sdp);
                     const msg = {type: 'webrtc/offer', value: pc.localDescription.sdp};
                     ws.send(JSON.stringify(msg));
                 });
@@ -82,14 +81,10 @@
             video.srcObject = ev.streams[0];
         }
 
-        // Safari don't support "offerToReceiveVideo"
-        // so need to create transeivers manually
-        pc.addTransceiver('video', {direction: 'recvonly'});
-        pc.addTransceiver('audio', {direction: 'recvonly'});
-
         if (stream) {
             stream.getTracks().forEach(track => {
-                const sender = pc.addTrack(track, stream)
+                pc.addTransceiver('audio', {direction: 'sendonly'});
+                const sender = pc.addTrack(track, stream);
                 // track.stop();
                 // setTimeout(() => {
                 //     navigator.mediaDevices.getUserMedia({audio: true}).then(stream => {
@@ -100,6 +95,11 @@
                 // }, 10000);
             });
         }
+
+        // Safari don't support "offerToReceiveVideo"
+        // so need to create transeivers manually
+        pc.addTransceiver('video', {direction: 'recvonly'});
+        pc.addTransceiver('audio', {direction: 'recvonly'});
     }
 
     if (navigator.mediaDevices) {