api: match WebRTC api more closely

errors.go

@@ -0,0 +1,92 @@
+package webrtc
+
+import (
+	"errors"
+	"fmt"
+)
+
+// Types of InvalidStateErrors
+var (
+	ErrConnectionClosed = errors.New("connection closed")
+)
+
+// InvalidStateError indicates the object is in an invalid state.
+type InvalidStateError struct {
+	Err error
+}
+
+func (e *InvalidStateError) Error() string {
+	return fmt.Sprintf("invalid state error: %v", e.Err)
+}
+
+// Types of UnknownErrors
+var (
+	ErrNoConfig = errors.New("no configuration provided")
+)
+
+// UnknownError indicates the operation failed for an unknown transient reason
+type UnknownError struct {
+	Err error
+}
+
+func (e *UnknownError) Error() string {
+	return fmt.Sprintf("unknown error: %v", e.Err)
+}
+
+// Types of InvalidAccessErrors
+var (
+	ErrCertificateExpired = errors.New("certificate expired")
+	ErrNoTurnCred         = errors.New("turn server credentials required")
+	ErrTurnCred           = errors.New("invalid turn server credentials")
+	ErrExistingTrack      = errors.New("track aready exists")
+)
+
+// InvalidAccessError indicates the object does not support the operation or argument.
+type InvalidAccessError struct {
+	Err error
+}
+
+func (e *InvalidAccessError) Error() string {
+	return fmt.Sprintf("invalid access error: %v", e.Err)
+}
+
+// Types of NotSupportedErrors
+var ()
+
+// NotSupportedError indicates the operation is not supported.
+type NotSupportedError struct {
+	Err error
+}
+
+func (e *NotSupportedError) Error() string {
+	return fmt.Sprintf("not supported error: %v", e.Err)
+}
+
+// Types of InvalidModificationErrors
+var (
+	ErrModPeerIdentity         = errors.New("peer identity cannot be modified")
+	ErrModCertificates         = errors.New("certificates cannot be modified")
+	ErrModRtcpMuxPolicy        = errors.New("rtcp mux policy cannot be modified")
+	ErrModIceCandidatePoolSize = errors.New("ice candidate pool size cannot be modified")
+)
+
+// InvalidModificationError indicates the object can not be modified in this way.
+type InvalidModificationError struct {
+	Err error
+}
+
+func (e *InvalidModificationError) Error() string {
+	return fmt.Sprintf("invalid modification error: %v", e.Err)
+}
+
+// Types of SyntaxErrors
+var ()
+
+// SyntaxError indicates the string did not match the expected pattern.
+type SyntaxError struct {
+	Err error
+}
+
+func (e *SyntaxError) Error() string {
+	return fmt.Sprintf("syntax error: %v", e.Err)
+}

examples/gstreamer-receive/gst/gst.go

@@ -23,9 +23,9 @@ type Pipeline struct {
 }
 
 // CreatePipeline creates a GStreamer Pipeline
-func CreatePipeline(codec webrtc.TrackType) *Pipeline {
+func CreatePipeline(codecName string) *Pipeline {
 	pipelineStr := "appsrc format=time is-live=true do-timestamp=true name=src ! application/x-rtp"
-	switch codec {
+	switch codecName {
 	case webrtc.VP8:
 		pipelineStr += ", encoding-name=VP8-DRAFT-IETF-01 ! rtpvp8depay ! decodebin ! autovideosink"
 	case webrtc.Opus:
@@ -35,7 +35,7 @@ func CreatePipeline(codec webrtc.TrackType) *Pipeline {
 	case webrtc.H264:
 		pipelineStr += " ! rtph264depay ! decodebin ! autovideosink"
 	default:
-		panic("Unhandled codec " + codec.String())
+		panic("Unhandled codec " + codecName)
 	}
 
 	pipelineStrUnsafe := C.CString(pipelineStr)

examples/gstreamer-receive/main.go

@@ -2,6 +2,7 @@ package main
 
 import (
 	"fmt"
+	"io"
 	"os"
 
 	"bufio"
@@ -10,13 +11,12 @@ import (
 	"github.com/pions/webrtc"
 	"github.com/pions/webrtc/examples/gstreamer-receive/gst"
 	"github.com/pions/webrtc/pkg/ice"
-	"github.com/pions/webrtc/pkg/rtp"
 )
 
 func main() {
 	reader := bufio.NewReader(os.Stdin)
 	rawSd, err := reader.ReadString('\n')
-	if err != nil {
+	if err != nil && err != io.EOF {
 		panic(err)
 	}
 
@@ -28,20 +28,25 @@ func main() {
 
 	/* Everything below is the pion-WebRTC API, thanks for using it! */
 
+	// Setup the codecs you want to use.
+	// We'll use the default ones but you can also define your own
+	webrtc.RegisterDefaultCodecs()
+
 	// Create a new RTCPeerConnection
-	peerConnection, err := webrtc.New(&webrtc.RTCConfiguration{})
+	peerConnection, err := webrtc.New(webrtc.RTCConfiguration{})
 	if err != nil {
 		panic(err)
 	}
 
 	// Set a handler for when a new remote track starts, this handler creates a gstreamer pipeline
 	// for the given codec
-	peerConnection.Ontrack = func(mediaType webrtc.TrackType, packets <-chan *rtp.Packet) {
-		fmt.Printf("Track has started, of type %s \n", mediaType.String())
-		pipeline := gst.CreatePipeline(mediaType)
+	peerConnection.Ontrack = func(track *webrtc.RTCTrack) {
+		codec := track.Codec
+		fmt.Printf("Track has started, of type %d: %s \n", track.PayloadType, codec.Name)
+		pipeline := gst.CreatePipeline(codec.Name)
 		pipeline.Start()
 		for {
-			p := <-packets
+			p := <-track.Packets
 			pipeline.Push(p.Raw)
 		}
 	}
@@ -53,17 +58,21 @@ func main() {
 	}
 
 	// Set the remote SessionDescription
-	if err := peerConnection.SetRemoteDescription(string(sd)); err != nil {
+	offer := webrtc.RTCSessionDescription{
+		Typ: webrtc.RTCSdpTypeOffer,
+		Sdp: string(sd),
+	}
+	if err := peerConnection.SetRemoteDescription(offer); err != nil {
 		panic(err)
 	}
 
 	// Sets the LocalDescription, and starts our UDP listeners
-	if err := peerConnection.CreateAnswer(); err != nil {
+	answer, err := peerConnection.CreateAnswer(nil)
+	if err != nil {
 		panic(err)
 	}
 
 	// Get the LocalDescription and take it to base64 so we can paste in browser
-	localDescriptionStr := peerConnection.LocalDescription.Marshal()
-	fmt.Println(base64.StdEncoding.EncodeToString([]byte(localDescriptionStr)))
+	fmt.Println(base64.StdEncoding.EncodeToString([]byte(answer.Sdp)))
 	select {}
 }

examples/gstreamer-send/gst/gst.go

@@ -24,16 +24,16 @@ type Pipeline struct {
 	Pipeline  *C.GstElement
 	in        chan<- webrtc.RTCSample
 	id        int
-	codec    webrtc.TrackType
+	codecName string
 }
 
 var pipelines = make(map[int]*Pipeline)
 var pipelinesLock sync.Mutex
 
 // CreatePipeline creates a GStreamer Pipeline
-func CreatePipeline(codec webrtc.TrackType, in chan<- webrtc.RTCSample) *Pipeline {
+func CreatePipeline(codecName string, in chan<- webrtc.RTCSample) *Pipeline {
 	pipelineStr := "appsink name=appsink"
-	switch codec {
+	switch codecName {
 	case webrtc.VP8:
 		pipelineStr = "videotestsrc ! vp8enc ! " + pipelineStr
 	case webrtc.VP9:
@@ -43,7 +43,7 @@ func CreatePipeline(codec webrtc.TrackType, in chan<- webrtc.RTCSample) *Pipelin
 	case webrtc.Opus:
 		pipelineStr = "audiotestsrc ! opusenc ! " + pipelineStr
 	default:
-		panic("Unhandled codec " + codec.String())
+		panic("Unhandled codec " + codecName)
 	}
 
 	pipelineStrUnsafe := C.CString(pipelineStr)
@@ -56,7 +56,7 @@ func CreatePipeline(codec webrtc.TrackType, in chan<- webrtc.RTCSample) *Pipelin
 		Pipeline:  C.gstreamer_send_create_pipeline(pipelineStrUnsafe),
 		in:        in,
 		id:        len(pipelines),
-		codec:    codec,
+		codecName: codecName,
 	}
 
 	pipelines[pipeline.id] = pipeline
@@ -85,7 +85,7 @@ func goHandlePipelineBuffer(buffer unsafe.Pointer, bufferLen C.int, duration C.i
 
 	if pipeline, ok := pipelines[int(pipelineID)]; ok {
 		var samples uint32
-		if pipeline.codec == webrtc.Opus {
+		if pipeline.codecName == webrtc.Opus {
 			samples = uint32(audioClockRate * (float32(duration) / 1000000000))
 		} else {
 			samples = uint32(videoClockRate * (float32(duration) / 1000000000))

examples/gstreamer-send/main.go

@@ -2,6 +2,7 @@ package main
 
 import (
 	"fmt"
+	"io"
 	"os"
 
 	"bufio"
@@ -15,7 +16,7 @@ import (
 func main() {
 	reader := bufio.NewReader(os.Stdin)
 	rawSd, err := reader.ReadString('\n')
-	if err != nil {
+	if err != nil && err != io.EOF {
 		panic(err)
 	}
 
@@ -27,46 +28,62 @@ func main() {
 
 	/* Everything below is the pion-WebRTC API, thanks for using it! */
 
+	// Setup the codecs you want to use.
+	// We'll use the default ones but you can also define your own
+	webrtc.RegisterDefaultCodecs()
+
 	// Create a new RTCPeerConnection
-	peerConnection, err := webrtc.New(&webrtc.RTCConfiguration{})
+	peerConnection, err := webrtc.New(webrtc.RTCConfiguration{})
 	if err != nil {
 		panic(err)
 	}
 
-	// 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.VP8, 90000)
-	if err != nil {
-		panic(err)
-	}
-
-	// Set the remote SessionDescription
-	if err := peerConnection.SetRemoteDescription(string(sd)); err != nil {
-		panic(err)
-	}
-
-	// Sets the LocalDescription, and starts our UDP listeners
-	if err := peerConnection.CreateAnswer(); err != nil {
-		panic(err)
-	}
-
 	// Set the handler for ICE connection state
 	// This will notify you when the peer has connected/disconnected
 	peerConnection.OnICEConnectionStateChange = func(connectionState ice.ConnectionState) {
 		fmt.Printf("Connection State has changed %s \n", connectionState.String())
 	}
 
+	// Create a audio track
+	opusTrack, err := peerConnection.NewRTCTrack(webrtc.PayloadTypeOpus, "audio", "pions1")
+	if err != nil {
+		panic(err)
+	}
+	_, err = peerConnection.AddTrack(opusTrack)
+	if err != nil {
+		panic(err)
+	}
+
+	// Create a video track
+	vp8Track, err := peerConnection.NewRTCTrack(webrtc.PayloadTypeVP8, "video", "pions2")
+	if err != nil {
+		panic(err)
+	}
+	_, err = peerConnection.AddTrack(vp8Track)
+	if err != nil {
+		panic(err)
+	}
+
+	// Set the remote SessionDescription
+	offer := webrtc.RTCSessionDescription{
+		Typ: webrtc.RTCSdpTypeOffer,
+		Sdp: string(sd),
+	}
+	if err := peerConnection.SetRemoteDescription(offer); err != nil {
+		panic(err)
+	}
+
+	// Sets the LocalDescription, and starts our UDP listeners
+	answer, err := peerConnection.CreateAnswer(nil)
+	if err != nil {
+		panic(err)
+	}
+
 	// Get the LocalDescription and take it to base64 so we can paste in browser
-	localDescriptionStr := peerConnection.LocalDescription.Marshal()
-	fmt.Println(base64.StdEncoding.EncodeToString([]byte(localDescriptionStr)))
+	fmt.Println(base64.StdEncoding.EncodeToString([]byte(answer.Sdp)))
 
 	// Start pushing buffers on these tracks
-	gst.CreatePipeline(webrtc.Opus, opusIn).Start()
-	gst.CreatePipeline(webrtc.VP8, vp8In).Start()
+	gst.CreatePipeline(webrtc.Opus, opusTrack.Samples).Start()
+	gst.CreatePipeline(webrtc.VP8, vp8Track.Samples).Start()
 	select {}
 }

examples/save-to-disk/main.go

@@ -4,17 +4,17 @@ import (
 	"bufio"
 	"encoding/base64"
 	"fmt"
+	"io"
 	"os"
 
 	"github.com/pions/webrtc"
 	"github.com/pions/webrtc/pkg/ice"
-	"github.com/pions/webrtc/pkg/rtp"
 )
 
 func main() {
 	reader := bufio.NewReader(os.Stdin)
 	rawSd, err := reader.ReadString('\n')
-	if err != nil {
+	if err != nil && err != io.EOF {
 		panic(err)
 	}
 
@@ -26,8 +26,13 @@ func main() {
 
 	/* Everything below is the pion-WebRTC API, thanks for using it! */
 
+	// Setup the codecs you want to use.
+	// We'll use a VP8 codec but you can also define your own
+	webrtc.RegisterCodec(webrtc.NewRTCRtpOpusCodec(webrtc.PayloadTypeOpus, 48000, 2))
+	webrtc.RegisterCodec(webrtc.NewRTCRtpVP8Codec(webrtc.PayloadTypeVP8, 90000))
+
 	// Create a new RTCPeerConnection
-	peerConnection, err := webrtc.New(&webrtc.RTCConfiguration{})
+	peerConnection, err := webrtc.New(webrtc.RTCConfiguration{})
 	if err != nil {
 		panic(err)
 	}
@@ -35,15 +40,15 @@ func main() {
 	// Set a handler for when a new remote track starts, this handler saves buffers to disk as
 	// an ivf file, since we could have multiple video tracks we provide a counter.
 	// In your application this is where you would handle/process video
-	peerConnection.Ontrack = func(mediaType webrtc.TrackType, packets <-chan *rtp.Packet) {
-		if mediaType == webrtc.VP8 {
+	peerConnection.Ontrack = func(track *webrtc.RTCTrack) {
+		if track.Codec.Name == webrtc.VP8 {
 			fmt.Println("Got VP8 track, saving to disk as output.ivf")
 			i, err := newIVFWriter("output.ivf")
 			if err != nil {
 				panic(err)
 			}
 			for {
-				i.addPacket(<-packets)
+				i.addPacket(<-track.Packets)
 			}
 		}
 	}
@@ -55,17 +60,21 @@ func main() {
 	}
 
 	// Set the remote SessionDescription
-	if err := peerConnection.SetRemoteDescription(string(sd)); err != nil {
+	offer := webrtc.RTCSessionDescription{
+		Typ: webrtc.RTCSdpTypeOffer,
+		Sdp: string(sd),
+	}
+	if err := peerConnection.SetRemoteDescription(offer); err != nil {
 		panic(err)
 	}
 
 	// Sets the LocalDescription, and starts our UDP listeners
-	if err := peerConnection.CreateAnswer(); err != nil {
+	answer, err := peerConnection.CreateAnswer(nil)
+	if err != nil {
 		panic(err)
 	}
 
 	// Get the LocalDescription and take it to base64 so we can paste in browser
-	localDescriptionStr := peerConnection.LocalDescription.Marshal()
-	fmt.Println(base64.StdEncoding.EncodeToString([]byte(localDescriptionStr)))
+	fmt.Println(base64.StdEncoding.EncodeToString([]byte(answer.Sdp)))
 	select {}
 }

examples/stun/main.go

@@ -4,17 +4,17 @@ import (
 	"bufio"
 	"encoding/base64"
 	"fmt"
+	"io"
 	"os"
 
 	"github.com/pions/webrtc"
 	"github.com/pions/webrtc/pkg/ice"
-	"github.com/pions/webrtc/pkg/rtp"
 )
 
 func main() {
 	reader := bufio.NewReader(os.Stdin)
 	rawSd, err := reader.ReadString('\n')
-	if err != nil {
+	if err != nil && err != io.EOF {
 		panic(err)
 	}
 
@@ -24,7 +24,14 @@ func main() {
 		panic(err)
 	}
 
-	peerConnection, err := webrtc.New(&webrtc.RTCConfiguration{
+	/* Everything below is the pion-WebRTC API, thanks for using it! */
+
+	// Setup the codecs you want to use.
+	// We'll use the default ones but you can also define your own
+	webrtc.RegisterDefaultCodecs()
+
+	// Create a new RTCPeerConnection, providing ICE servers
+	peerConnection, err := webrtc.New(webrtc.RTCConfiguration{
 		ICEServers: []webrtc.RTCICEServer{
 			{
 				URLs: []string{"stun:stun.l.google.com:19302"},
@@ -35,23 +42,30 @@ func main() {
 		panic(err)
 	}
 
-	peerConnection.Ontrack = func(mediaType webrtc.TrackType, packets <-chan *rtp.Packet) {
-		fmt.Printf("Got a %s track\n", mediaType)
+	peerConnection.Ontrack = func(track *webrtc.RTCTrack) {
+		fmt.Printf("Got a %s track\n", track.Codec.Name)
 	}
 
 	peerConnection.OnICEConnectionStateChange = func(connectionState ice.ConnectionState) {
 		fmt.Printf("Connection State has changed %s \n", connectionState.String())
 	}
 
-	if err := peerConnection.SetRemoteDescription(string(sd)); err != nil {
+	// Set the remote SessionDescription
+	offer := webrtc.RTCSessionDescription{
+		Typ: webrtc.RTCSdpTypeOffer,
+		Sdp: string(sd),
+	}
+	if err := peerConnection.SetRemoteDescription(offer); err != nil {
 		panic(err)
 	}
 
-	if err := peerConnection.CreateAnswer(); err != nil {
+	// Sets the LocalDescription, and starts our UDP listeners
+	answer, err := peerConnection.CreateAnswer(nil)
+	if err != nil {
 		panic(err)
 	}
 
-	localDescriptionStr := peerConnection.LocalDescription.Marshal()
-	fmt.Println(base64.StdEncoding.EncodeToString([]byte(localDescriptionStr)))
+	// Get the LocalDescription and take it to base64 so we can paste in browser
+	fmt.Println(base64.StdEncoding.EncodeToString([]byte(answer.Sdp)))
 	select {}
 }

internal/network/port-receive.go

@@ -94,7 +94,7 @@ func (p *Port) handleICE(in *incomingPacket, remoteKey []byte, iceTimer *time.Ti
 		); err != nil {
 			fmt.Println(err)
 		} else {
-			p.ICEState = ice.Completed
+			p.ICEState = ice.ConnectionStateCompleted
 			iceTimer.Reset(iceTimeout)
 			iceNotifier(p)
 		}
@@ -131,7 +131,7 @@ func (p *Port) networkLoop(remoteKey []byte, tlscfg *dtls.TLSCfg, b BufferTransp
 	for {
 		select {
 		case <-iceTimer.C:
-			p.ICEState = ice.Failed
+			p.ICEState = ice.ConnectionStateFailed
 			iceNotifier(p)
 		case in, inValid := <-incomingPackets:
 			if !inValid {

internal/sdp/jsep.go

@@ -0,0 +1,135 @@
+package sdp
+
+import (
+	"fmt"
+	"time"
+)
+
+// Constants for SDP attributes used in JSEP
+const (
+	AttrKeyIdentity        = "identity"
+	AttrKeyGroup           = "group"
+	AttrKeySsrc            = "ssrc"
+	AttrKeySsrcGroup       = "ssrc-group"
+	AttrKeyMsidSemantic    = "msid-semantic"
+	AttrKeyConnectionSetup = "setup"
+	AttrKeyMID             = "mid"
+	AttrKeyICELite         = "ice-lite"
+	AttrKeyRtcpMux         = "rtcp-mux"
+	AttrKeyRtcpRsize       = "rtcp-rsize"
+)
+
+// Constants for semantic tokens used in JSEP
+const (
+	SemanticTokenLipSynchronization     = "LS"
+	SemanticTokenFlowIdentification     = "FID"
+	SemanticTokenForwardErrorCorrection = "FEC"
+	SemanticTokenWebRTCMediaStreams     = "WMS"
+)
+
+// API to match draft-ietf-rtcweb-jsep
+// Move to webrtc or its own package?
+
+// NewJSEPSessionDescription creates a new SessionDescription with
+// some settings that are required by the JSEP spec.
+func NewJSEPSessionDescription(fingerprint string, identity bool) *SessionDescription {
+	d := &SessionDescription{
+		ProtocolVersion: 0,
+		Origin: fmt.Sprintf(
+			"- %d %d IN IP4 0.0.0.0",
+			newSessionID(),
+			time.Now().Unix(),
+		),
+		SessionName: "-",
+		Timing:      []string{"0 0"},
+		Attributes: []string{
+			//	"ice-options:trickle", // TODO: implement trickle ICE
+			"fingerprint:sha-256 " + fingerprint,
+		},
+	}
+
+	if identity {
+		d.WithPropertyAttribute(AttrKeyIdentity)
+	}
+
+	return d
+}
+
+// WithPropertyAttribute adds a property attribute 'a=key' to the session description
+func (d *SessionDescription) WithPropertyAttribute(key string) *SessionDescription {
+	d.Attributes = append(d.Attributes, key)
+	return d
+}
+
+// WithValueAttribute adds a value attribute 'a=key:value' to the session description
+func (d *SessionDescription) WithValueAttribute(key, value string) *SessionDescription {
+	d.Attributes = append(d.Attributes, fmt.Sprintf("%s:%s", key, value))
+	return d
+}
+
+// WithMedia adds a media description to the session description
+func (d *SessionDescription) WithMedia(md *MediaDescription) *SessionDescription {
+	d.MediaDescriptions = append(d.MediaDescriptions, md)
+	return d
+}
+
+// NewJSEPMediaDescription creates a new MediaDescription with
+// some settings that are required by the JSEP spec.
+func NewJSEPMediaDescription(typ string, codecPrefs []string) *MediaDescription {
+	// TODO: handle codecPrefs
+	d := &MediaDescription{
+		MediaName:      fmt.Sprintf("%s 9 UDP/TLS/RTP/SAVPF", typ), // TODO: other transports?
+		ConnectionData: "IN IP4 0.0.0.0",
+		Attributes:     []string{},
+	}
+	return d
+}
+
+// WithPropertyAttribute adds a property attribute 'a=key' to the media description
+func (d *MediaDescription) WithPropertyAttribute(key string) *MediaDescription {
+	d.Attributes = append(d.Attributes, key)
+	return d
+}
+
+// WithValueAttribute adds a value attribute 'a=key:value' to the media description
+func (d *MediaDescription) WithValueAttribute(key, value string) *MediaDescription {
+	d.Attributes = append(d.Attributes, fmt.Sprintf("%s:%s", key, value))
+	return d
+}
+
+// WithICECredentials adds ICE credentials to the media description
+func (d *MediaDescription) WithICECredentials(username, password string) *MediaDescription {
+	return d.
+		WithValueAttribute("ice-ufrag", username).
+		WithValueAttribute("ice-pwd", password)
+}
+
+// WithCodec adds codec information to the media description
+func (d *MediaDescription) WithCodec(payloadType uint8, name string, clockrate uint32, channels uint16, fmtp string) *MediaDescription {
+	d.MediaName = fmt.Sprintf("%s %d", d.MediaName, payloadType)
+	rtpmap := fmt.Sprintf("%d %s/%d", payloadType, name, clockrate)
+	if channels > 0 {
+		rtpmap = rtpmap + fmt.Sprintf("/%d", channels)
+	}
+	d.WithValueAttribute("rtpmap", rtpmap)
+	if fmtp != "" {
+		d.WithValueAttribute("fmtp", fmt.Sprintf("%d %s", payloadType, fmtp))
+	}
+	return d
+}
+
+// WithMediaSource adds media source information to the media description
+func (d *MediaDescription) WithMediaSource(ssrc uint32, cname, streamLabel, label string) *MediaDescription {
+	return d.
+		WithValueAttribute("ssrc", fmt.Sprintf("%d cname:%s", ssrc, cname)). // Deprecated but not pased out?
+		WithValueAttribute("ssrc", fmt.Sprintf("%d msid:%s %s", ssrc, streamLabel, label)).
+		WithValueAttribute("ssrc", fmt.Sprintf("%d mslabel:%s", ssrc, streamLabel)). // Deprecated but not pased out?
+		WithValueAttribute("ssrc", fmt.Sprintf("%d label:%s", ssrc, label))          // Deprecated but not pased out?
+}
+
+// WithCandidate adds an ICE candidate to the media description
+func (d *MediaDescription) WithCandidate(id int, transport string, basePriority uint16, ip string, port int, typ string) *MediaDescription {
+	return d.
+		WithValueAttribute("candidate",
+			fmt.Sprintf("%scandidate %d %s %d %s %d typ %s", transport, id, transport, basePriority, ip, port, typ))
+}

internal/sdp/util.go

@@ -1,6 +1,7 @@
 package sdp
 
 import (
+	"errors"
 	"fmt"
 	"math/rand"
 	"strconv"
@@ -22,102 +23,99 @@ type SessionBuilder struct {
 	Tracks []*SessionBuilderTrack
 }
 
-// BaseSessionDescription generates a default SDP response that is ice-lite, initiates the DTLS session and
-// supports VP8, VP9, H264 and Opus
-func BaseSessionDescription(b *SessionBuilder) *SessionDescription {
-	addMediaCandidates := func(m *MediaDescription) *MediaDescription {
-		m.Attributes = append(m.Attributes, b.Candidates...)
-		m.Attributes = append(m.Attributes, "end-of-candidates")
-		return m
-	}
+// ConnectionRole indicates which of the end points should initiate the connection establishment
+type ConnectionRole int
 
-	audioMediaDescription := &MediaDescription{
-		MediaName:      "audio 9 RTP/SAVPF 111",
-		ConnectionData: "IN IP4 127.0.0.1",
-		Attributes: []string{
-			"setup:active",
-			"mid:audio",
-			"sendrecv",
-			"ice-ufrag:" + b.IceUsername,
-			"ice-pwd:" + b.IcePassword,
-			"ice-lite",
-			"fingerprint:sha-256 " + b.Fingerprint,
-			"rtcp-mux",
-			"rtcp-rsize",
-			"rtpmap:111 opus/48000/2",
-			"fmtp:111 minptime=10;useinbandfec=1",
-		},
-	}
+const (
 
-	videoMediaDescription := &MediaDescription{
-		MediaName:      "video 9 RTP/SAVPF 96 98 100",
-		ConnectionData: "IN IP4 127.0.0.1",
-		Attributes: []string{
-			"setup:active",
-			"mid:video",
-			"sendrecv",
-			"ice-ufrag:" + b.IceUsername,
-			"ice-pwd:" + b.IcePassword,
-			"ice-lite",
-			"fingerprint:sha-256 " + b.Fingerprint,
-			"rtcp-mux",
-			"rtcp-rsize",
-			"rtpmap:96 VP8/90000",
-			"rtpmap:98 VP9/90000",
-			"rtpmap:100 H264/90000",
-			"fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f",
-		},
-	}
+	// ConnectionRoleActive indicates the endpoint will initiate an outgoing connection.
+	ConnectionRoleActive ConnectionRole = iota + 1
 
-	mediaStreamsAttribute := "msid-semantic: WMS"
-	for i, track := range b.Tracks {
-		var attributes *[]string
-		if track.IsAudio {
-			attributes = &audioMediaDescription.Attributes
-		} else {
-			attributes = &videoMediaDescription.Attributes
-		}
-		appendAttr := func(attr string) {
-			*attributes = append(*attributes, attr)
-		}
+	// ConnectionRolePassive indicates the endpoint will accept an incoming connection.
+	ConnectionRolePassive
 
-		appendAttr("ssrc:" + fmt.Sprint(track.SSRC) + " cname:pion" + strconv.Itoa(i))
-		appendAttr("ssrc:" + fmt.Sprint(track.SSRC) + " msid:pion" + strconv.Itoa(i) + " pion" + strconv.Itoa(i))
-		appendAttr("ssrc:" + fmt.Sprint(track.SSRC) + " mslabel:pion" + strconv.Itoa(i))
-		appendAttr("ssrc:" + fmt.Sprint(track.SSRC) + " label:pion" + strconv.Itoa(i))
+	// ConnectionRoleActpass indicates the endpoint is willing to accept an incoming connection or to initiate an outgoing connection.
+	ConnectionRoleActpass
 
-		mediaStreamsAttribute += " pion" + strconv.Itoa(i)
-	}
+	// ConnectionRoleHoldconn indicates the endpoint does not want the connection to be established for the time being.
+	ConnectionRoleHoldconn
+)
 
-	sessionID := strconv.FormatUint(uint64(rand.Uint32())<<32+uint64(rand.Uint32()), 10)
-	return &SessionDescription{
-		ProtocolVersion: 0,
-		Origin:          "pion-webrtc " + sessionID + " 2 IN IP4 0.0.0.0",
-		SessionName:     "-",
-		Timing:          []string{"0 0"},
-		Attributes: []string{
-			"group:BUNDLE audio video",
-			mediaStreamsAttribute,
-		},
-		MediaDescriptions: []*MediaDescription{
-			addMediaCandidates(audioMediaDescription),
-			addMediaCandidates(videoMediaDescription),
-		},
+func (t ConnectionRole) String() string {
+	switch t {
+	case ConnectionRoleActive:
+		return "active"
+	case ConnectionRolePassive:
+		return "passive"
+	case ConnectionRoleActpass:
+		return "actpass"
+	case ConnectionRoleHoldconn:
+		return "holdconn"
+	default:
+		return "Unknown"
 	}
 }
 
+func newSessionID() uint64 {
+	return uint64(rand.Uint32())<<32 + uint64(rand.Uint32())
+}
+
+// Codec represents a codec
+type Codec struct {
+	PayloadType        uint8
+	Name               string
+	ClockRate          uint32
+	EncodingParameters string
+	Fmtp               string
+}
+
+func (c Codec) String() string {
+	return fmt.Sprintf("%d %s/%d/%s", c.PayloadType, c.Name, c.ClockRate, c.EncodingParameters)
+}
+
 // GetCodecForPayloadType scans the SessionDescription for the given payloadType and returns the codec
-func GetCodecForPayloadType(payloadType uint8, sd *SessionDescription) (ok bool, codec string) {
+func (sd *SessionDescription) GetCodecForPayloadType(payloadType uint8) (Codec, error) {
+	codec := Codec{
+		PayloadType: payloadType,
+	}
+
+	found := false
+	payloadTypeString := strconv.Itoa(int(payloadType))
+	rtpmapPrefix := "rtpmap:" + payloadTypeString
+	fmtpPrefix := "fmtp:" + payloadTypeString
+
 	for _, m := range sd.MediaDescriptions {
 		for _, a := range m.Attributes {
-			if strings.Contains(a, "rtpmap:"+strconv.Itoa(int(payloadType))) {
+			if strings.HasPrefix(a, rtpmapPrefix) {
+				found = true
+				// a=rtpmap:<payload type> <encoding name>/<clock rate> [/<encoding parameters>]
 				split := strings.Split(a, " ")
 				if len(split) == 2 {
 					split := strings.Split(split[1], "/")
-					return true, split[0]
+					codec.Name = split[0]
+					parts := len(split)
+					if parts > 1 {
+						rate, err := strconv.Atoi(split[1])
+						if err != nil {
+							return codec, err
+						}
+						codec.ClockRate = uint32(rate)
+					}
+					if parts > 2 {
+						codec.EncodingParameters = split[2]
+					}
+				}
+			} else if strings.HasPrefix(a, fmtpPrefix) {
+				// a=fmtp:<format> <format specific parameters>
+				split := strings.Split(a, " ")
+				if len(split) == 2 {
+					codec.Fmtp = split[1]
 				}
 			}
 		}
+		if found {
+			return codec, nil
 		}
-	return false, ""
+	}
+	return codec, errors.New("payload type not found")
 }

media.go

@@ -0,0 +1,246 @@
+package webrtc
+
+import (
+	"math/rand"
+
+	"github.com/pions/webrtc/pkg/rtp"
+	"github.com/pkg/errors"
+)
+
+// RTCRtpReceiver allows an application to inspect the receipt of a RTCTrack
+type RTCRtpReceiver struct {
+	Track *RTCTrack
+	// receiverTrack *RTCTrack
+	// receiverTransport
+	// receiverRtcpTransport
+}
+
+// TODO: receiving side
+// func newRTCRtpReceiver(kind, id string) {
+//
+// }
+
+// RTCRtpSender allows an application to control how a given RTCTrack is encoded and transmitted to a remote peer
+type RTCRtpSender struct {
+	Track *RTCTrack
+	// senderTrack *RTCTrack
+	// senderTransport
+	// senderRtcpTransport
+}
+
+func newRTCRtpSender(track *RTCTrack) *RTCRtpSender {
+	s := &RTCRtpSender{
+		Track: track,
+	}
+	return s
+}
+
+// RTCRtpTransceiverDirection indicates the direction of the RTCRtpTransceiver
+type RTCRtpTransceiverDirection int
+
+const (
+
+	// RTCRtpTransceiverDirectionSendrecv indicates the RTCRtpSender will offer to send RTP and RTCRtpReceiver the will offer to receive RTP
+	RTCRtpTransceiverDirectionSendrecv RTCRtpTransceiverDirection = iota + 1
+
+	// RTCRtpTransceiverDirectionSendonly indicates the RTCRtpSender will offer to send RTP
+	RTCRtpTransceiverDirectionSendonly
+
+	// RTCRtpTransceiverDirectionRecvonly indicates the RTCRtpReceiver the will offer to receive RTP
+	RTCRtpTransceiverDirectionRecvonly
+
+	// RTCRtpTransceiverDirectionInactive indicates the RTCRtpSender won't offer to send RTP and RTCRtpReceiver the won't offer to receive RTP
+	RTCRtpTransceiverDirectionInactive
+)
+
+func (t RTCRtpTransceiverDirection) String() string {
+	switch t {
+	case RTCRtpTransceiverDirectionSendrecv:
+		return "sendrecv"
+	case RTCRtpTransceiverDirectionSendonly:
+		return "sendonly"
+	case RTCRtpTransceiverDirectionRecvonly:
+		return "recvonly"
+	case RTCRtpTransceiverDirectionInactive:
+		return "inactive"
+	default:
+		return "Unknown"
+	}
+}
+
+// RTCRtpTransceiver represents a combination of an RTCRtpSender and an RTCRtpReceiver that share a common mid.
+type RTCRtpTransceiver struct {
+	Mid       string
+	Sender    *RTCRtpSender
+	Receiver  *RTCRtpReceiver
+	Direction RTCRtpTransceiverDirection
+	// currentDirection RTCRtpTransceiverDirection
+	// firedDirection   RTCRtpTransceiverDirection
+	// receptive bool
+	stopped bool
+}
+
+func (t *RTCRtpTransceiver) setSendingTrack(track *RTCTrack) {
+	t.Sender.Track = track
+
+	switch t.Direction {
+	case RTCRtpTransceiverDirectionRecvonly:
+		t.Direction = RTCRtpTransceiverDirectionSendrecv
+	case RTCRtpTransceiverDirectionInactive:
+		t.Direction = RTCRtpTransceiverDirectionSendonly
+	default:
+		panic("Invalid state change in RTCRtpTransceiver.setSending")
+	}
+}
+
+func (r *RTCPeerConnection) newRTCRtpTransceiver(
+	receiver *RTCRtpReceiver,
+	sender *RTCRtpSender,
+	direction RTCRtpTransceiverDirection,
+) *RTCRtpTransceiver {
+
+	t := &RTCRtpTransceiver{
+		Receiver:  receiver,
+		Sender:    sender,
+		Direction: direction,
+	}
+	r.rtpTransceivers = append(r.rtpTransceivers, t)
+	return t
+}
+
+// Stop irreversibly stops the RTCRtpTransceiver
+func (t *RTCRtpTransceiver) Stop() error {
+	panic("TODO")
+}
+
+// RTCSample contains media, and the amount of samples in it
+type RTCSample struct {
+	Data    []byte
+	Samples uint32
+}
+
+// RTCTrack represents a track that is communicated
+type RTCTrack struct {
+	PayloadType uint8
+	Kind        RTCRtpCodecType
+	ID          string
+	Label       string
+	Ssrc        uint32
+	Codec       *RTCRtpCodec
+	Packets     <-chan *rtp.Packet
+	Samples     chan<- RTCSample
+}
+
+// NewRTCTrack is used to create a new RTCTrack
+func (r *RTCPeerConnection) NewRTCTrack(payloadType uint8, id, label string) (*RTCTrack, error) {
+	codec, err := rtcMediaEngine.getCodec(payloadType)
+	if err != nil {
+		return nil, err
+	}
+
+	if codec.Payloader == nil {
+		return nil, errors.New("codec payloader not set")
+	}
+
+	trackInput := make(chan RTCSample, 15) // Is the buffering needed?
+	ssrc := rand.Uint32()
+	go func() {
+		packetizer := rtp.NewPacketizer(
+			1400,
+			payloadType,
+			ssrc,
+			codec.Payloader,
+			rtp.NewRandomSequencer(),
+			codec.ClockRate,
+		)
+		for {
+			in := <-trackInput
+			packets := packetizer.Packetize(in.Data, in.Samples)
+			for _, p := range packets {
+				for _, port := range r.ports {
+					port.Send(p)
+				}
+			}
+		}
+	}()
+
+	t := &RTCTrack{
+		PayloadType: payloadType,
+		Kind:        codec.Type,
+		ID:          id,
+		Label:       label,
+		Ssrc:        ssrc,
+		Codec:       codec,
+		Samples:     trackInput,
+	}
+
+	return t, nil
+}
+
+// AddTrack adds a RTCTrack to the RTCPeerConnection
+func (r *RTCPeerConnection) AddTrack(track *RTCTrack) (*RTCRtpSender, error) {
+	if r.IsClosed {
+		return nil, &InvalidStateError{Err: ErrConnectionClosed}
+	}
+	for _, tranceiver := range r.rtpTransceivers {
+		if tranceiver.Sender.Track == nil {
+			continue
+		}
+		if track.ID == tranceiver.Sender.Track.ID {
+			return nil, &InvalidAccessError{Err: ErrExistingTrack}
+		}
+	}
+	var tranciever *RTCRtpTransceiver
+	for _, t := range r.rtpTransceivers {
+		if !t.stopped &&
+			// t.Sender == nil && // TODO: check that the sender has never sent
+			t.Sender.Track == nil &&
+			t.Receiver.Track != nil &&
+			t.Receiver.Track.Kind == track.Kind {
+			tranciever = t
+			break
+		}
+	}
+	if tranciever != nil {
+		tranciever.setSendingTrack(track)
+	} else {
+		var receiver *RTCRtpReceiver
+		sender := newRTCRtpSender(track)
+		tranciever = r.newRTCRtpTransceiver(
+			receiver,
+			sender,
+			RTCRtpTransceiverDirectionSendonly,
+		)
+	}
+
+	tranciever.Mid = track.Kind.String() // TODO: Mid generation
+
+	return tranciever.Sender, nil
+}
+
+// GetSenders returns the RTCRtpSender that are currently attached to this RTCPeerConnection
+func (r *RTCPeerConnection) GetSenders() []RTCRtpSender {
+	result := make([]RTCRtpSender, len(r.rtpTransceivers))
+	for i, tranceiver := range r.rtpTransceivers {
+		result[i] = *tranceiver.Sender
+	}
+	return result
+}
+
+// GetReceivers returns the RTCRtpReceivers that are currently attached to this RTCPeerConnection
+func (r *RTCPeerConnection) GetReceivers() []RTCRtpReceiver {
+	result := make([]RTCRtpReceiver, len(r.rtpTransceivers))
+	for i, tranceiver := range r.rtpTransceivers {
+		result[i] = *tranceiver.Receiver
+	}
+	return result
+}
+
+// GetTransceivers returns the RTCRtpTransceiver that are currently attached to this RTCPeerConnection
+func (r *RTCPeerConnection) GetTransceivers() []RTCRtpTransceiver {
+	result := make([]RTCRtpTransceiver, len(r.rtpTransceivers))
+	for i, tranceiver := range r.rtpTransceivers {
+		result[i] = *tranceiver
+	}
+	return result
+}

mediaengine.go

@@ -0,0 +1,210 @@
+package webrtc
+
+import (
+	"strconv"
+
+	"github.com/pions/webrtc/internal/sdp"
+	"github.com/pions/webrtc/pkg/rtp"
+	"github.com/pions/webrtc/pkg/rtp/codecs"
+	"github.com/pkg/errors"
+)
+
+// PayloadTypes for the default codecs
+const (
+	PayloadTypeOpus = 111
+	PayloadTypeVP8  = 96
+	PayloadTypeVP9  = 98
+	PayloadTypeH264 = 100
+)
+
+// Names for the default codecs
+const (
+	Opus = "opus"
+	VP8  = "VP8"
+	VP9  = "VP9"
+	H264 = "H264"
+)
+
+var rtcMediaEngine = &mediaEngine{}
+
+// RegisterDefaultCodecs is a helper that registers the default codecs supported by pions-webrtc
+func RegisterDefaultCodecs() {
+	RegisterCodec(NewRTCRtpOpusCodec(PayloadTypeOpus, 48000, 2))
+	RegisterCodec(NewRTCRtpVP8Codec(PayloadTypeVP8, 90000))
+	RegisterCodec(NewRTCRtpH264Codec(PayloadTypeH264, 90000))
+	RegisterCodec(NewRTCRtpVP9Codec(PayloadTypeVP9, 90000))
+}
+
+// RegisterCodec is used to register a codec
+func RegisterCodec(codec *RTCRtpCodec) {
+	rtcMediaEngine.RegisterCodec(codec)
+}
+
+type mediaEngine struct {
+	codecs []*RTCRtpCodec
+}
+
+func (m *mediaEngine) RegisterCodec(codec *RTCRtpCodec) uint8 {
+	// TODO: generate PayloadType if not set
+	m.codecs = append(m.codecs, codec)
+	return codec.PayloadType
+}
+
+func (m *mediaEngine) ClearCodecs() {
+	m.codecs = nil
+}
+
+func (m *mediaEngine) getCodec(payloadType uint8) (*RTCRtpCodec, error) {
+	for _, codec := range m.codecs {
+		if codec.PayloadType == payloadType {
+			return codec, nil
+		}
+	}
+	return nil, errors.New("Codec not found")
+}
+
+func (m *mediaEngine) getCodecSDP(sdpCodec sdp.Codec) (*RTCRtpCodec, error) {
+	for _, codec := range m.codecs {
+		if codec.Name == sdpCodec.Name &&
+			codec.ClockRate == sdpCodec.ClockRate &&
+			(sdpCodec.EncodingParameters == "" ||
+				strconv.Itoa(int(codec.Channels)) == sdpCodec.EncodingParameters) &&
+			codec.SdpFmtpLine == sdpCodec.Fmtp { // TODO: Protocol specific matching?
+			return codec, nil
+		}
+	}
+	return nil, errors.New("Codec not found")
+}
+
+func (m *mediaEngine) getCodecsByKind(kind RTCRtpCodecType) []*RTCRtpCodec {
+	var codecs []*RTCRtpCodec
+	for _, codec := range rtcMediaEngine.codecs {
+		if codec.Type == kind {
+			codecs = append(codecs, codec)
+		}
+	}
+	return codecs
+}
+
+// NewRTCRtpOpusCodec is a helper to create an Opus codec
+func NewRTCRtpOpusCodec(payloadType uint8, clockrate uint32, channels uint16) *RTCRtpCodec {
+	c := NewRTCRtpCodec(RTCRtpCodecTypeAudio,
+		Opus,
+		clockrate,
+		channels,
+		"minptime=10;useinbandfec=1",
+		payloadType,
+		&codecs.OpusPayloader{})
+	return c
+}
+
+// NewRTCRtpVP8Codec is a helper to create an VP8 codec
+func NewRTCRtpVP8Codec(payloadType uint8, clockrate uint32) *RTCRtpCodec {
+	c := NewRTCRtpCodec(RTCRtpCodecTypeVideo,
+		VP8,
+		clockrate,
+		0,
+		"",
+		payloadType,
+		&codecs.VP8Payloader{})
+	return c
+}
+
+// NewRTCRtpVP9Codec is a helper to create an VP9 codec
+func NewRTCRtpVP9Codec(payloadType uint8, clockrate uint32) *RTCRtpCodec {
+	c := NewRTCRtpCodec(RTCRtpCodecTypeVideo,
+		VP9,
+		clockrate,
+		0,
+		"",
+		payloadType,
+		nil) // TODO
+	return c
+}
+
+// NewRTCRtpH264Codec is a helper to create an H264 codec
+func NewRTCRtpH264Codec(payloadType uint8, clockrate uint32) *RTCRtpCodec {
+	c := NewRTCRtpCodec(RTCRtpCodecTypeVideo,
+		H264,
+		clockrate,
+		0,
+		"level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f",
+		payloadType,
+		&codecs.H264Payloader{})
+	return c
+}
+
+// RTCRtpCodecType determines the type of a codec
+type RTCRtpCodecType int
+
+const (
+
+	// RTCRtpCodecTypeAudio indicates this is an audio codec
+	RTCRtpCodecTypeAudio RTCRtpCodecType = iota + 1
+
+	// RTCRtpCodecTypeVideo indicates this is a video codec
+	RTCRtpCodecTypeVideo
+)
+
+func (t RTCRtpCodecType) String() string {
+	switch t {
+	case RTCRtpCodecTypeAudio:
+		return "audio"
+	case RTCRtpCodecTypeVideo:
+		return "video"
+	default:
+		return "Unknown"
+	}
+}
+
+// RTCRtpCodec represents a codec supported by the PeerConnection
+type RTCRtpCodec struct {
+	RTCRtpCodecCapability
+	Type        RTCRtpCodecType
+	Name        string
+	PayloadType uint8
+	Payloader   rtp.Payloader
+}
+
+// NewRTCRtpCodec is used to define a new codec
+func NewRTCRtpCodec(
+	typ RTCRtpCodecType,
+	name string,
+	clockrate uint32,
+	channels uint16,
+	fmtp string,
+	payloadType uint8,
+	payloader rtp.Payloader,
+) *RTCRtpCodec {
+	return &RTCRtpCodec{
+		RTCRtpCodecCapability: RTCRtpCodecCapability{
+			MimeType:    typ.String() + "/" + name,
+			ClockRate:   clockrate,
+			Channels:    channels,
+			SdpFmtpLine: fmtp,
+		},
+		PayloadType: payloadType,
+		Payloader:   payloader,
+		Type:        typ,
+		Name:        name,
+	}
+}
+
+// RTCRtpCodecCapability provides information about codec capabilities.
+type RTCRtpCodecCapability struct {
+	MimeType    string
+	ClockRate   uint32
+	Channels    uint16
+	SdpFmtpLine string
+}
+
+// RTCRtpHeaderExtensionCapability is used to define a RFC5285 RTP header extension supported by the codec.
+type RTCRtpHeaderExtensionCapability struct {
+	URI string
+}
+
+// RTCRtpCapabilities represents the capabilities of a transceiver
+type RTCRtpCapabilities struct {
+	Codecs           []RTCRtpCodecCapability
+	HeaderExtensions []RTCRtpHeaderExtensionCapability
+}

pkg/ice/address.go

@@ -0,0 +1,179 @@
+package ice
+
+import (
+	"log"
+	"strconv"
+	"strings"
+
+	"github.com/pkg/errors"
+)
+
+// TODO: Migrate address parsing to STUN/TURN packages?
+
+var (
+	// ErrServerType indicates the server type could not be parsed
+	ErrServerType = errors.New("unknown server type")
+
+	// ErrSTUNQuery indicates query arguments are provided in a STUN URL
+	ErrSTUNQuery = errors.New("queries not supported in stun address")
+
+	// ErrInvalidQuery indicates an unsupported query is provided
+	ErrInvalidQuery = errors.New("invalid query")
+
+	// ErrTransportType indicates an unsupported transport type was provided
+	ErrTransportType = errors.New("invalid transport type")
+
+	// ErrHost indicates the server hostname could not be parsed
+	ErrHost = errors.New("invalid hostname")
+
+	// ErrPort indicates the server port could not be parsed
+	ErrPort = errors.New("invalid port")
+)
+
+// ServerType indicates the type of server used
+type ServerType int
+
+const (
+	// ServerTypeSTUN indicates the URL represents a STUN server
+	ServerTypeSTUN ServerType = iota + 1
+
+	// ServerTypeTURN indicates the URL represents a TURN server
+	ServerTypeTURN
+)
+
+func (t ServerType) String() string {
+	switch t {
+	case ServerTypeSTUN:
+		return "stun"
+	case ServerTypeTURN:
+		return "turn"
+	default:
+		return "Unknown"
+	}
+}
+
+// TransportType indicates the transport that is used
+type TransportType int
+
+const (
+	// TransportUDP indicates the URL uses a UDP transport
+	TransportUDP TransportType = iota + 1
+
+	// TransportTCP indicates the URL uses a TCP transport
+	TransportTCP
+)
+
+func (t TransportType) String() string {
+	switch t {
+	case TransportUDP:
+		return "udp"
+	case TransportTCP:
+		return "tcp"
+	default:
+		return "Unknown"
+	}
+}
+
+// URL represents a STUN (rfc7064) or TRUN (rfc7065) URL
+type URL struct {
+	Type          ServerType
+	Secure        bool
+	Host          string
+	Port          int
+	TransportType TransportType
+}
+
+// NewURL creates a new URL by parsing a STUN (rfc7064) or TRUN (rfc7065) uri string
+func NewURL(address string) (URL, error) {
+	var result URL
+
+	var scheme string
+	scheme, address = split(address, ":")
+
+	switch strings.ToLower(scheme) {
+	case "stun":
+		result.Type = ServerTypeSTUN
+		result.Secure = false
+
+	case "stuns":
+		result.Type = ServerTypeSTUN
+		result.Secure = true
+
+	case "turn":
+		result.Type = ServerTypeTURN
+		result.Secure = false
+
+	case "turns":
+		result.Type = ServerTypeTURN
+		result.Secure = true
+
+	default:
+		return result, ErrServerType
+	}
+
+	var query string
+	address, query = split(address, "?")
+
+	if query != "" {
+		if result.Type == ServerTypeSTUN {
+			return result, ErrSTUNQuery
+		}
+		key, value := split(query, "=")
+		if strings.ToLower(key) != "transport" {
+			return result, ErrInvalidQuery
+		}
+		switch strings.ToLower(value) {
+		case "udp":
+			result.TransportType = TransportUDP
+		case "tcp":
+			result.TransportType = TransportTCP
+		default:
+			return result, ErrTransportType
+		}
+	} else {
+		if result.Secure {
+			result.TransportType = TransportTCP
+		} else {
+			result.TransportType = TransportUDP
+		}
+	}
+
+	var host string
+	var port string
+	colon := strings.IndexByte(address, ':')
+	if colon == -1 {
+		host = address
+		if result.Secure {
+			port = "5349"
+		} else {
+			port = "3478"
+		}
+	} else if i := strings.IndexByte(address, ']'); i != -1 {
+		host = strings.TrimPrefix(address[:i], "[")
+		port = address[i+1+len(":"):]
+		log.Println(port)
+	} else {
+		host = address[:colon]
+		port = address[colon+len(":"):]
+	}
+	if host == "" {
+		return result, ErrHost
+	}
+	result.Host = strings.ToLower(host)
+
+	var err error
+	result.Port, err = strconv.Atoi(port)
+	if err != nil {
+		return result, ErrPort
+	}
+
+	return result, nil
+}
+
+func split(s string, c string) (string, string) {
+	i := strings.Index(s, c)
+	if i < 0 {
+		return s, ""
+	}
+	return s[:i], s[i+len(c):]
+}

pkg/ice/address_test.go

@@ -0,0 +1,68 @@
+package ice
+
+import "testing"
+
+func TestNewURL(t *testing.T) {
+	t.Run("Success", func(t *testing.T) {
+		testCases := []struct {
+			rawURL                string
+			expectedType          ServerType
+			expectedSecure        bool
+			expectedHost          string
+			expectedPort          int
+			expectedTransportType TransportType
+		}{
+			{"stun:google.de", ServerTypeSTUN, false, "google.de", 3478, TransportUDP},
+			{"stun:google.de:1234", ServerTypeSTUN, false, "google.de", 1234, TransportUDP},
+			{"stuns:google.de", ServerTypeSTUN, true, "google.de", 5349, TransportTCP},
+			{"stun:[::1]:123", ServerTypeSTUN, false, "::1", 123, TransportUDP},
+			{"turn:google.de", ServerTypeTURN, false, "google.de", 3478, TransportUDP},
+			{"turns:google.de", ServerTypeTURN, true, "google.de", 5349, TransportTCP},
+			{"turn:google.de?transport=udp", ServerTypeTURN, false, "google.de", 3478, TransportUDP},
+			{"turn:google.de?transport=tcp", ServerTypeTURN, false, "google.de", 3478, TransportTCP},
+		}
+
+		for i, testCase := range testCases {
+			url, err := NewURL(testCase.rawURL)
+			if err != nil {
+				t.Errorf("Case %d: got error: %v", i, err)
+			}
+
+			if url.Type != testCase.expectedType ||
+				url.Secure != testCase.expectedSecure ||
+				url.Host != testCase.expectedHost ||
+				url.Port != testCase.expectedPort ||
+				url.TransportType != testCase.expectedTransportType {
+				t.Errorf("Case %d: got %s %t %s %d %s",
+					i,
+					url.Type,
+					url.Secure,
+					url.Host,
+					url.Port,
+					url.TransportType,
+				)
+			}
+		}
+	})
+	t.Run("Failure", func(t *testing.T) {
+		testCases := []struct {
+			rawURL      string
+			expectedErr error
+		}{
+			{"", ErrServerType},
+			{":::", ErrServerType},
+			{"google.de", ErrServerType},
+			{"stun:", ErrHost},
+			{"stun:google.de:abc", ErrPort},
+			{"stun:google.de?transport=udp", ErrSTUNQuery},
+			{"turn:google.de?trans=udp", ErrInvalidQuery},
+			{"turn:google.de?transport=ip", ErrTransportType},
+		}
+
+		for i, testCase := range testCases {
+			if _, err := NewURL(testCase.rawURL); err != testCase.expectedErr {
+				t.Errorf("Case %d: got error '%v' expected '%v'", i, err, testCase.expectedErr)
+			}
+		}
+	})
+}

pkg/ice/agent.go

@@ -0,0 +1,23 @@
+package ice
+
+import "github.com/pions/webrtc/internal/util"
+
+// Agent represents the ICE agent
+type Agent struct {
+	Servers [][]URL
+	Ufrag   string
+	Pwd     string
+}
+
+// NewAgent creates a new Agent
+func NewAgent() *Agent {
+	return &Agent{
+		Ufrag: util.RandSeq(16),
+		Pwd:   util.RandSeq(32),
+	}
+}
+
+// SetServers is used to set the ICE servers used by the Agent
+func (a *Agent) SetServers(urls [][]URL) {
+	a.Servers = urls
+}

pkg/ice/ice.go

@@ -7,49 +7,76 @@ type ConnectionState int
 
 // List of supported States
 const (
-	// New ICE agent is gathering addresses
-	New = iota + 1
+	// ConnectionStateNew ICE agent is gathering addresses
+	ConnectionStateNew = iota + 1
 
-	// Checking ICE agent has been given local and remote candidates, and is attempting to find a match
-	Checking
+	// ConnectionStateChecking ICE agent has been given local and remote candidates, and is attempting to find a match
+	ConnectionStateChecking
 
-	// Connected ICE agent has a pairing, but is still checking other pairs
-	Connected
+	// ConnectionStateConnected ICE agent has a pairing, but is still checking other pairs
+	ConnectionStateConnected
 
-	// Completed ICE agent has finished
-	Completed
+	// ConnectionStateCompleted ICE agent has finished
+	ConnectionStateCompleted
 
-	// Failed ICE agent never could sucessfully connect
-	Failed
+	// ConnectionStateFailed ICE agent never could sucessfully connect
+	ConnectionStateFailed
 
-	// Failed ICE agent connected sucessfully, but has entered a failed state
-	Disconnected
+	// ConnectionStateDisconnected ICE agent connected sucessfully, but has entered a failed state
+	ConnectionStateDisconnected
 
-	// Closed ICE agent has finished and is no longer handling requests
-	Closed
+	// ConnectionStateClosed ICE agent has finished and is no longer handling requests
+	ConnectionStateClosed
 )
 
 func (c ConnectionState) String() string {
 	switch c {
-	case New:
+	case ConnectionStateNew:
 		return "New"
-	case Checking:
+	case ConnectionStateChecking:
 		return "Checking"
-	case Connected:
+	case ConnectionStateConnected:
 		return "Connected"
-	case Completed:
+	case ConnectionStateCompleted:
 		return "Completed"
-	case Failed:
+	case ConnectionStateFailed:
 		return "Failed"
-	case Disconnected:
+	case ConnectionStateDisconnected:
 		return "Disconnected"
-	case Closed:
+	case ConnectionStateClosed:
 		return "Closed"
 	default:
 		return "Invalid"
 	}
 }
 
+// GatheringState describes the state of the candidate gathering process
+type GatheringState int
+
+const (
+	// GatheringStateNew indicates candidate gatering is not yet started
+	GatheringStateNew GatheringState = iota + 1
+
+	// GatheringStateGathering indicates candidate gatering is ongoing
+	GatheringStateGathering
+
+	// GatheringStateComplete indicates candidate gatering has been completed
+	GatheringStateComplete
+)
+
+func (t GatheringState) String() string {
+	switch t {
+	case GatheringStateNew:
+		return "new"
+	case GatheringStateGathering:
+		return "gathering"
+	case GatheringStateComplete:
+		return "complete"
+	default:
+		return "Unknown"
+	}
+}
+
 // HostInterfaces generates a slice of all the IPs associated with interfaces
 func HostInterfaces() (ips []string) {
 	ifaces, err := net.Interfaces()

rtcconfiguration.go

@@ -1,68 +1,133 @@
 package webrtc
 
 import (
-	"fmt"
-	"strings"
+	"time"
+
+	"github.com/pions/webrtc/pkg/ice"
 )
 
-// RTCCredentialType specifies the type of credentials provided
-type RTCCredentialType string
+// RTCICECredentialType indicates the type of credentials used to connect to an ICE server
+type RTCICECredentialType int
 
-var (
-	// RTCCredentialTypePassword describes username+pasword based credentials
-	RTCCredentialTypePassword RTCCredentialType = "password"
-	// RTCCredentialTypeToken describes token based credentials
-	RTCCredentialTypeToken RTCCredentialType = "token"
+const (
+	// RTCICECredentialTypePassword describes username+pasword based credentials
+	RTCICECredentialTypePassword RTCICECredentialType = iota + 1
+	// RTCICECredentialTypeOauth describes token based credentials
+	RTCICECredentialTypeOauth
 )
 
-// RTCServerType is used to identify different ICE server types
-type RTCServerType string
+func (t RTCICECredentialType) String() string {
+	switch t {
+	case RTCICECredentialTypePassword:
+		return "password"
+	case RTCICECredentialTypeOauth:
+		return "oauth"
+	default:
+		return "Unknown"
+	}
+}
 
-var (
-	// RTCServerTypeSTUN is used to identify STUN servers. Prefix is stun:
-	RTCServerTypeSTUN RTCServerType = "STUN"
-	// RTCServerTypeTURN is used to identify TURN servers. Prefix is turn:
-	RTCServerTypeTURN RTCServerType = "TURN"
-	// RTCServerTypeUnknown is used when an ICE server can not be identified properly.
-	RTCServerTypeUnknown RTCServerType = "UnknownType"
-)
+// RTCCertificate represents a certificate used to authenticate WebRTC communications.
+type RTCCertificate struct {
+	expires time.Time
+	// TODO: Finish during DTLS implementation
+}
+
+// Equals determines if two certificates are identical
+func (c RTCCertificate) Equals(other RTCCertificate) bool {
+	return c.expires == other.expires
+}
 
 // RTCICEServer describes a single ICE server, as well as required credentials
 type RTCICEServer struct {
-	CredentialType RTCCredentialType
 	URLs           []string
 	Username       string
-	Credential     string
+	Credential     RTCICECredential
+	CredentialType RTCICECredentialType
 }
 
+// RTCICECredential represents credentials used to connect to an ICE server
+// Two types of credentials are supported:
+// - Password (type string)
+// - Password (type RTCOAuthCredential)
+type RTCICECredential interface{}
+
+// RTCOAuthCredential represents OAuth credentials used to connect to an ICE server
+type RTCOAuthCredential struct {
+	MacKey      string
+	AccessToken string
+}
+
+// RTCICETransportPolicy defines the ICE candidate policy [JSEP] (section 3.5.3.) used to surface the permitted candidates
+type RTCICETransportPolicy int
+
 const (
-	stunPrefix  = "stun:"
-	stunsPrefix = "stuns:"
-	turnPrefix  = "turn:"
-	turnsPrefix = "turns:"
+	// RTCICETransportPolicyRelay indicates only media relay candidates such as candidates passing through a TURN server are used
+	RTCICETransportPolicyRelay RTCICETransportPolicy = iota + 1
+
+	// RTCICETransportPolicyAll indicates any type of candidate is used
+	RTCICETransportPolicyAll
 )
 
-func (c RTCICEServer) serverType() RTCServerType {
-	for _, url := range c.URLs {
-		if strings.HasPrefix(url, stunPrefix) || strings.HasPrefix(url, stunsPrefix) {
-			return RTCServerTypeSTUN
+func (t RTCICETransportPolicy) String() string {
+	switch t {
+	case RTCICETransportPolicyRelay:
+		return "relay"
+	case RTCICETransportPolicyAll:
+		return "all"
+	default:
+		return "Unknown"
 	}
-		if strings.HasPrefix(url, turnPrefix) || strings.HasPrefix(url, turnsPrefix) {
-			return RTCServerTypeTURN
-		}
-	}
-	return RTCServerTypeUnknown
 }
 
-func protocolAndHost(url string) (string, string, error) {
-	if strings.HasPrefix(url, stunPrefix) {
-		return "udp", url[len(stunPrefix):], nil
+// RTCBundlePolicy affects which media tracks are negotiated if the remote endpoint is not bundle-aware, and what ICE candidates are gathered.
+type RTCBundlePolicy int
+
+const (
+
+	// RTCRtcpMuxPolicyBalanced indicates to gather ICE candidates for each media type in use (audio, video, and data).
+	RTCRtcpMuxPolicyBalanced RTCBundlePolicy = iota + 1
+
+	// RTCRtcpMuxPolicyMaxCompat indicates to gather ICE candidates for each track.
+	RTCRtcpMuxPolicyMaxCompat
+
+	// RTCRtcpMuxPolicyMaxBundle indicates to gather ICE candidates for only one track.
+	RTCRtcpMuxPolicyMaxBundle
+)
+
+func (t RTCBundlePolicy) String() string {
+	switch t {
+	case RTCRtcpMuxPolicyBalanced:
+		return "balanced"
+	case RTCRtcpMuxPolicyMaxCompat:
+		return "max-compat"
+	case RTCRtcpMuxPolicyMaxBundle:
+		return "max-bundle"
+	default:
+		return "Unknown"
 	}
-	if strings.HasPrefix(url, stunsPrefix) {
-		return "tcp", url[len(stunsPrefix):], nil
+}
+
+// RTCRtcpMuxPolicy affects what ICE candidates are gathered to support non-multiplexed RTCP
+type RTCRtcpMuxPolicy int
+
+const (
+	// RTCRtcpMuxPolicyNegotiate indicates to gather ICE candidates for both RTP and RTCP candidates.
+	RTCRtcpMuxPolicyNegotiate RTCRtcpMuxPolicy = iota + 1
+
+	// RTCRtcpMuxPolicyRequire indicates to gather ICE candidates only for RTP and multiplex RTCP on the RTP candidates
+	RTCRtcpMuxPolicyRequire
+)
+
+func (t RTCRtcpMuxPolicy) String() string {
+	switch t {
+	case RTCRtcpMuxPolicyNegotiate:
+		return "negotiate"
+	case RTCRtcpMuxPolicyRequire:
+		return "require"
+	default:
+		return "Unknown"
 	}
-	// TODO TURN urls
-	return "", "", fmt.Errorf("Unknown protocol in URL %q", url)
 }
 
 // RTCConfiguration contains RTCPeerConfiguration options
@@ -71,4 +136,152 @@ type RTCConfiguration struct {
 	// these are typically STUN and/or TURN servers. If this isn't specified, the ICE agent may choose to use its own ICE servers;
 	// otherwise, the connection attempt will be made with no STUN or TURN server available, which limits the connection to local peers.
 	ICEServers           []RTCICEServer
+	ICETransportPolicy   RTCICETransportPolicy
+	BundlePolicy         RTCBundlePolicy
+	RtcpMuxPolicy        RTCRtcpMuxPolicy
+	PeerIdentity         string
+	Certificates         []RTCCertificate
+	ICECandidatePoolSize uint8
+}
+
+// SetConfiguration updates the configuration of the RTCPeerConnection
+func (r *RTCPeerConnection) SetConfiguration(config RTCConfiguration) error {
+	err := r.validatePeerIdentity(config)
+	if err != nil {
+		return err
+	}
+	err = r.validateCertificates(config)
+	if err != nil {
+		return err
+	}
+	err = r.validateBundlePolicy(config)
+	if err != nil {
+		return err
+	}
+	err = r.validateRtcpMuxPolicy(config)
+	if err != nil {
+		return err
+	}
+	err = r.validateICECandidatePoolSize(config)
+	if err != nil {
+		return err
+	}
+
+	err = r.setICEServers(config)
+	if err != nil {
+		return err
+	}
+
+	r.config = config
+
+	return nil
+}
+
+func (r *RTCPeerConnection) validatePeerIdentity(config RTCConfiguration) error {
+	current := r.config
+	if current.PeerIdentity != "" &&
+		config.PeerIdentity != "" &&
+		config.PeerIdentity != current.PeerIdentity {
+		return &InvalidModificationError{Err: ErrModPeerIdentity}
+	}
+	return nil
+}
+
+func (r *RTCPeerConnection) validateCertificates(config RTCConfiguration) error {
+	current := r.config
+	if len(current.Certificates) > 0 &&
+		len(config.Certificates) > 0 {
+		if len(config.Certificates) != len(current.Certificates) {
+			return &InvalidModificationError{Err: ErrModCertificates}
+		}
+		for i, cert := range config.Certificates {
+			if !current.Certificates[i].Equals(cert) {
+				return &InvalidModificationError{Err: ErrModCertificates}
+			}
+		}
+	}
+
+	now := time.Now()
+	for _, cert := range config.Certificates {
+		if now.After(cert.expires) {
+			return &InvalidAccessError{Err: ErrCertificateExpired}
+		}
+		// TODO: Check certificate 'origin'
+	}
+	return nil
+}
+
+func (r *RTCPeerConnection) validateBundlePolicy(config RTCConfiguration) error {
+	current := r.config
+	if config.BundlePolicy != current.BundlePolicy {
+		return &InvalidModificationError{Err: ErrModRtcpMuxPolicy}
+	}
+	return nil
+}
+
+func (r *RTCPeerConnection) validateRtcpMuxPolicy(config RTCConfiguration) error {
+	current := r.config
+	if config.RtcpMuxPolicy != current.RtcpMuxPolicy {
+		return &InvalidModificationError{Err: ErrModRtcpMuxPolicy}
+	}
+	return nil
+}
+
+func (r *RTCPeerConnection) validateICECandidatePoolSize(config RTCConfiguration) error {
+	current := r.config
+	if r.LocalDescription != nil &&
+		config.ICECandidatePoolSize != current.ICECandidatePoolSize {
+		return &InvalidModificationError{Err: ErrModIceCandidatePoolSize}
+	}
+	return nil
+}
+
+func (r *RTCPeerConnection) setICEServers(config RTCConfiguration) error {
+	if len(config.ICEServers) > 0 {
+		var servers [][]ice.URL
+		for _, server := range config.ICEServers {
+			var urls []ice.URL
+			for _, rawURL := range server.URLs {
+				url, err := parseICEServer(server, rawURL)
+				if err != nil {
+					return err
+				}
+				urls = append(urls, url)
+			}
+			if len(urls) > 0 {
+				servers = append(servers, urls)
+			}
+		}
+		if len(servers) > 0 {
+			r.iceAgent.SetServers(servers)
+		}
+	}
+	return nil
+}
+
+func parseICEServer(server RTCICEServer, rawURL string) (ice.URL, error) {
+	iceurl, err := ice.NewURL(rawURL)
+	if err != nil {
+		return iceurl, &SyntaxError{Err: err}
+	}
+
+	_, isPass := server.Credential.(string)
+	_, isOauth := server.Credential.(RTCOAuthCredential)
+	noPass := !isPass && !isOauth
+
+	if iceurl.Type == ice.ServerTypeTURN {
+		if server.Username == "" ||
+			noPass {
+			return iceurl, &InvalidAccessError{Err: ErrNoTurnCred}
+		}
+		if server.CredentialType == RTCICECredentialTypePassword &&
+			!isPass {
+			return iceurl, &InvalidAccessError{Err: ErrTurnCred}
+		}
+		if server.CredentialType == RTCICECredentialTypeOauth &&
+			!isOauth {
+			return iceurl, &InvalidAccessError{Err: ErrTurnCred}
+		}
+	}
+	return iceurl, nil
 }

rtcconfiguration_test.go

@@ -1,44 +0,0 @@
-package webrtc
-
-import "testing"
-
-func TestRTCICEServer_isStun(t *testing.T) {
-	testCases := []struct {
-		expectedType RTCServerType
-		server       RTCICEServer
-	}{
-		{RTCServerTypeSTUN, RTCICEServer{URLs: []string{"stun:google.de"}}},
-		{RTCServerTypeTURN, RTCICEServer{URLs: []string{"turn:google.de"}}},
-		{RTCServerTypeUnknown, RTCICEServer{URLs: []string{"google.de"}}},
-	}
-
-	for _, testCase := range testCases {
-		if serverType := testCase.server.serverType(); serverType != testCase.expectedType {
-			t.Errorf("Expected %q to be %s, but got %s", testCase.server.URLs, testCase.expectedType, serverType)
-		}
-	}
-}
-
-func TestPortAndHost(t *testing.T) {
-	testCases := []struct {
-		url              string
-		expectedHost     string
-		expectedProtocol string
-	}{
-		{"stun:stun.l.google.com:19302", "stun.l.google.com:19302", "udp"},
-		{"stuns:stun.l.google.com:19302", "stun.l.google.com:19302", "tcp"},
-	}
-
-	for _, testCase := range testCases {
-		proto, host, err := protocolAndHost(testCase.url)
-		if err != nil {
-			t.Fatalf("unable to get proto and host: %v", err)
-		}
-		if proto != testCase.expectedProtocol {
-			t.Fatalf("expected %s, got %s", testCase.expectedProtocol, proto)
-		}
-		if host != testCase.expectedHost {
-			t.Fatalf("expected %s, got %s", testCase.expectedHost, host)
-		}
-	}
-}

rtcpeerconnection.go

@@ -3,18 +3,14 @@ package webrtc
 import (
 	"fmt"
 	"math/rand"
-	"net"
 	"sync"
 	"time"
 
-	"github.com/pions/pkg/stun"
 	"github.com/pions/webrtc/internal/dtls"
 	"github.com/pions/webrtc/internal/network"
 	"github.com/pions/webrtc/internal/sdp"
-	"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"
 )
@@ -23,212 +19,119 @@ func init() {
 	rand.Seed(time.Now().UTC().UnixNano())
 }
 
-// RTCSample contains media, and the amount of samples in it
-type RTCSample struct {
-	Data    []byte
-	Samples uint32
-}
+// RTCPeerConnectionState indicates the state of the peer connection
+type RTCPeerConnectionState int
 
-// TrackType determines the type of media we are sending receiving
-type TrackType int
-
-// List of supported TrackTypes
 const (
-	VP8 TrackType = iota + 1
-	VP9
-	H264
-	Opus
+
+	// RTCPeerConnectionStateNew indicates some of the ICE or DTLS transports are in status new
+	RTCPeerConnectionStateNew RTCPeerConnectionState = iota + 1
+
+	// RTCPeerConnectionStateConnecting indicates some of the ICE or DTLS transports are in status connecting or checking
+	RTCPeerConnectionStateConnecting
+
+	// RTCPeerConnectionStateConnected indicates all of the ICE or DTLS transports are in status connected or completed
+	RTCPeerConnectionStateConnected
+
+	// RTCPeerConnectionStateDisconnected indicates some of the ICE or DTLS transports are in status disconnected
+	RTCPeerConnectionStateDisconnected
+
+	// RTCPeerConnectionStateFailed indicates some of the ICE or DTLS transports are in status failed
+	RTCPeerConnectionStateFailed
+
+	// RTCPeerConnectionStateClosed indicates the peer connection is closed
+	RTCPeerConnectionStateClosed
 )
 
-func (t TrackType) String() string {
+func (t RTCPeerConnectionState) String() string {
 	switch t {
-	case VP8:
-		return "VP8"
-	case VP9:
-		return "VP9"
-	case H264:
-		return "H264"
-	case Opus:
-		return "Opus"
+	case RTCPeerConnectionStateNew:
+		return "new"
+	case RTCPeerConnectionStateConnecting:
+		return "connecting"
+	case RTCPeerConnectionStateConnected:
+		return "connected"
+	case RTCPeerConnectionStateDisconnected:
+		return "disconnected"
+	case RTCPeerConnectionStateFailed:
+		return "failed"
+	case RTCPeerConnectionStateClosed:
+		return "closed"
 	default:
 		return "Unknown"
 	}
 }
 
-// New creates a new RTCPeerConfiguration with the provided configuration
-func New(config *RTCConfiguration) (*RTCPeerConnection, error) {
-	return &RTCPeerConnection{
-		config: config,
-	}, nil
-}
-
 // RTCPeerConnection represents a WebRTC connection between itself and a remote peer
 type RTCPeerConnection struct {
-	Ontrack                    func(mediaType TrackType, buffers <-chan *rtp.Packet)
-	LocalDescription           *sdp.SessionDescription
+	// ICE
 	OnICEConnectionStateChange func(iceConnectionState ice.ConnectionState)
 
-	config *RTCConfiguration
+	config RTCConfiguration
 	tlscfg *dtls.TLSCfg
 
-	iceUfrag string
-	icePwd   string
+	// ICE: TODO: Move to ICEAgent
+	iceAgent           *ice.Agent
 	iceState           ice.ConnectionState
+	iceGatheringState  ice.GatheringState
+	iceConnectionState ice.ConnectionState
 
 	portsLock sync.RWMutex
 	ports     []*network.Port
 
+	// Signaling
+	// pendingLocalDescription *RTCSessionDescription
+	// currentLocalDescription *RTCSessionDescription
+	LocalDescription *sdp.SessionDescription
+
+	// pendingRemoteDescription *RTCSessionDescription
+	currentRemoteDescription *RTCSessionDescription
 	remoteDescription        *sdp.SessionDescription
 
-	localTracks []*sdp.SessionBuilderTrack
+	idpLoginURL *string
+
+	IsClosed          bool
+	NegotiationNeeded bool
+
+	// lastOffer  string
+	// lastAnswer string
+
+	signalingState  RTCSignalingState
+	connectionState RTCPeerConnectionState
+
+	// Media
+	rtpTransceivers []*RTCRtpTransceiver
+	Ontrack         func(*RTCTrack)
+}
+
+// New creates a new RTCPeerConfiguration with the provided configuration
+func New(config RTCConfiguration) (*RTCPeerConnection, error) {
+
+	r := &RTCPeerConnection{
+		config:             config,
+		signalingState:     RTCSignalingStateStable,
+		iceAgent:           ice.NewAgent(),
+		iceGatheringState:  ice.GatheringStateNew,
+		iceConnectionState: ice.ConnectionStateNew,
+		connectionState:    RTCPeerConnectionStateNew,
+	}
+	err := r.SetConfiguration(config)
+	if err != nil {
+		return nil, err
+	}
+
+	r.tlscfg = dtls.NewTLSCfg()
+
+	// TODO: Initialize ICE Agent
+
+	return r, nil
 }
 
 // Public
 
-// SetRemoteDescription sets the SessionDescription of the remote peer
-func (r *RTCPeerConnection) SetRemoteDescription(rawSessionDescription string) error {
-	if r.remoteDescription != nil {
-		return errors.Errorf("remoteDescription is already defined, SetRemoteDescription can only be called once")
-	}
-
-	r.remoteDescription = &sdp.SessionDescription{}
-	return r.remoteDescription.Unmarshal(rawSessionDescription)
-}
-
-// CreateOffer starts the RTCPeerConnection and generates the localDescription
-func (r *RTCPeerConnection) CreateOffer() error {
-	return errors.Errorf("CreateOffer is not implemented")
-}
-
-// CreateAnswer starts the RTCPeerConnection and generates the localDescription
-func (r *RTCPeerConnection) CreateAnswer() error {
-	if r.tlscfg != nil {
-		return errors.Errorf("tlscfg is already defined, CreateOffer can only be called once")
-	}
-
-	r.tlscfg = dtls.NewTLSCfg()
-	r.iceUfrag = util.RandSeq(16)
-	r.icePwd = util.RandSeq(32)
-
-	r.portsLock.Lock()
-	defer r.portsLock.Unlock()
-
-	candidates := []string{}
-	basePriority := uint16(rand.Uint32() & (1<<16 - 1))
-	for _, c := range ice.HostInterfaces() {
-		port, err := network.NewPort(c+":0", []byte(r.icePwd), r.tlscfg, r.generateChannel, r.iceStateChange)
-		if err != nil {
-			return err
-		}
-		candidates = append(candidates, fmt.Sprintf("candidate:udpcandidate 1 udp %d %s %d typ host", basePriority, c, port.ListeningAddr.Port))
-		basePriority = basePriority + 1
-		r.ports = append(r.ports, port)
-	}
-	if r.config != nil {
-		for _, server := range r.config.ICEServers {
-			if server.serverType() != RTCServerTypeSTUN {
-				continue
-			}
-
-			for _, iceURL := range server.URLs {
-				proto, host, err := protocolAndHost(iceURL)
-				// TODO if one of the URLs does not work we should just ignore it.
-				if err != nil {
-					return errors.Wrapf(err, "Failed to parse ICE URL")
-				}
-				// TODO Do we want the timeout to be configurable?
-				client, err := stun.NewClient(proto, host, time.Second*5)
-				if err != nil {
-					return errors.Wrapf(err, "Failed to create STUN client")
-				}
-				localAddr, ok := client.LocalAddr().(*net.UDPAddr)
-				if !ok {
-					return errors.Errorf("Failed to cast STUN client to UDPAddr")
-				}
-
-				resp, err := client.Request()
-				if err != nil {
-					return errors.Wrapf(err, "Failed to make STUN request")
-				}
-
-				if err := client.Close(); err != nil {
-					return errors.Wrapf(err, "Failed to close STUN client")
-				}
-
-				attr, ok := resp.GetOneAttribute(stun.AttrXORMappedAddress)
-				if !ok {
-					return errors.Errorf("Got respond from STUN server that did not contain XORAddress")
-				}
-
-				var addr stun.XorAddress
-				if err := addr.Unpack(resp, attr); err != nil {
-					return errors.Wrapf(err, "Failed to unpack STUN XorAddress response")
-				}
-
-				port, err := network.NewPort(fmt.Sprintf("0.0.0.0:%d", localAddr.Port), []byte(r.icePwd), r.tlscfg, r.generateChannel, r.iceStateChange)
-				if err != nil {
-					return errors.Wrapf(err, "Failed to build network/port")
-				}
-				candidates = append(candidates, fmt.Sprintf("candidate:%scandidate 1 %s %d %s %d typ srflx", proto, proto, basePriority, addr.IP.String(), localAddr.Port))
-				basePriority = basePriority + 1
-				r.ports = append(r.ports, port)
-			}
-		}
-	}
-
-	r.LocalDescription = sdp.BaseSessionDescription(&sdp.SessionBuilder{
-		IceUsername: r.iceUfrag,
-		IcePassword: r.icePwd,
-		Fingerprint: r.tlscfg.Fingerprint(),
-		Candidates:  candidates,
-		Tracks:      r.localTracks,
-	})
-
-	return nil
-}
-
-// AddTrack adds a new track to the RTCPeerConnection
-// This function returns a channel to push buffers on, and an error if the channel can't be added
-// Closing the channel ends this stream
-func (r *RTCPeerConnection) AddTrack(mediaType TrackType, clockRate uint32) (samples chan<- RTCSample, err error) {
-	if mediaType != VP8 && mediaType != H264 && mediaType != Opus {
-		panic("TODO Discarding packet, need media parsing")
-	}
-
-	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
-		}
-
-		r.localTracks = append(r.localTracks, sdpTrack)
-		packetizer := rtp.NewPacketizer(1400, payloadType, ssrc, payloader, rtp.NewRandomSequencer(), clockRate)
-		for {
-			in := <-trackInput
-			packets := packetizer.Packetize(in.Data, in.Samples)
-			for _, p := range packets {
-				for _, port := range r.ports {
-					port.Send(p)
-				}
-			}
-		}
-	}()
-	return trackInput, nil
+// SetIdentityProvider is used to configure an identity provider to generate identity assertions
+func (r *RTCPeerConnection) SetIdentityProvider(provider string) error {
+	panic("TODO SetIdentityProvider")
 }
 
 // Close ends the RTCPeerConnection
@@ -252,29 +155,32 @@ func (r *RTCPeerConnection) generateChannel(ssrc uint32, payloadType uint8) (buf
 		return nil
 	}
 
-	var codec TrackType
-	ok, codecStr := sdp.GetCodecForPayloadType(payloadType, r.remoteDescription)
-	if !ok {
+	sdpCodec, err := r.remoteDescription.GetCodecForPayloadType(payloadType)
+	if err != nil {
 		fmt.Printf("No codec could be found in RemoteDescription for payloadType %d \n", payloadType)
 		return nil
 	}
 
-	switch codecStr {
-	case "VP8":
-		codec = VP8
-	case "VP9":
-		codec = VP9
-	case "opus":
-		codec = Opus
-	case "H264":
-		codec = H264
-	default:
-		fmt.Printf("Codec %s in not supported by pion-WebRTC \n", codecStr)
-		return nil
+	codec, err := rtcMediaEngine.getCodecSDP(sdpCodec)
+	if err != nil {
+		fmt.Printf("Codec %s in not registered\n", sdpCodec)
 	}
 
 	bufferTransport := make(chan *rtp.Packet, 15)
-	go r.Ontrack(codec, bufferTransport)
+
+	track := &RTCTrack{
+		PayloadType: payloadType,
+		Kind:        codec.Type,
+		ID:          "0", // TODO extract from remoteDescription
+		Label:       "",  // TODO extract from remoteDescription
+		Ssrc:        ssrc,
+		Codec:       codec,
+		Packets:     bufferTransport,
+	}
+
+	// TODO: Register the receiving Track
+
+	go r.Ontrack(track)
 	return bufferTransport
 }
 
@@ -287,7 +193,7 @@ func (r *RTCPeerConnection) iceStateChange(p *network.Port) {
 		r.iceState = newState
 	}
 
-	if p.ICEState == ice.Failed {
+	if p.ICEState == ice.ConnectionStateFailed {
 		if err := p.Close(); err != nil {
 			fmt.Println(errors.Wrap(err, "Failed to close Port when ICE went to failed"))
 		}
@@ -301,9 +207,9 @@ func (r *RTCPeerConnection) iceStateChange(p *network.Port) {
 		}
 
 		if len(r.ports) == 0 {
-			updateAndNotify(ice.Disconnected)
+			updateAndNotify(ice.ConnectionStateDisconnected)
 		}
 	} else {
-		updateAndNotify(ice.Connected)
+		updateAndNotify(ice.ConnectionStateConnected)
 	}
 }

signaling.go

@@ -0,0 +1,407 @@
+package webrtc
+
+import (
+	"fmt"
+	"math/rand"
+	"net"
+	"time"
+
+	"github.com/pions/pkg/stun"
+	"github.com/pions/webrtc/internal/network"
+	"github.com/pions/webrtc/internal/sdp"
+	"github.com/pions/webrtc/pkg/ice"
+	"github.com/pkg/errors"
+)
+
+/*
+                      setRemote(OFFER)               setLocal(PRANSWER)
+                          /-----\                               /-----\
+                          |     |                               |     |
+                          v     |                               v     |
+           +---------------+    |                +---------------+    |
+           |               |----/                |               |----/
+           |  have-        | setLocal(PRANSWER)  | have-         |
+           |  remote-offer |------------------- >| local-pranswer|
+           |               |                     |               |
+           |               |                     |               |
+           +---------------+                     +---------------+
+                ^   |                                   |
+                |   | setLocal(ANSWER)                  |
+  setRemote(OFFER)  |                                   |
+                |   V                  setLocal(ANSWER) |
+           +---------------+                            |
+           |               |                            |
+           |               |<---------------------------+
+           |    stable     |
+           |               |<---------------------------+
+           |               |                            |
+           +---------------+          setRemote(ANSWER) |
+                ^   |                                   |
+                |   | setLocal(OFFER)                   |
+  setRemote(ANSWER) |                                   |
+                |   V                                   |
+           +---------------+                     +---------------+
+           |               |                     |               |
+           |  have-        | setRemote(PRANSWER) |have-          |
+           |  local-offer  |------------------- >|remote-pranswer|
+           |               |                     |               |
+           |               |----\                |               |----\
+           +---------------+    |                +---------------+    |
+                          ^     |                               ^     |
+                          |     |                               |     |
+                          \-----/                               \-----/
+                      setLocal(OFFER)               setRemote(PRANSWER)
+*/
+
+// RTCSignalingState indicates the state of the offer/answer process
+type RTCSignalingState int
+
+const (
+
+	// RTCSignalingStateStable indicates there is no offer­answer exchange in progress.
+	RTCSignalingStateStable RTCSignalingState = iota + 1
+
+	// RTCSignalingStateHaveLocalOffer indicates A local description, of type "offer", has been successfully applied.
+	RTCSignalingStateHaveLocalOffer
+
+	// RTCSignalingStateHaveRemoteOffer indicates A remote description, of type "offer", has been successfully applied.
+	RTCSignalingStateHaveRemoteOffer
+
+	// RTCSignalingStateHaveLocalPranswer indicates A remote description of type "offer" has been successfully applied and a local description of type "pranswer" has been successfully applied.
+	RTCSignalingStateHaveLocalPranswer
+
+	// RTCSignalingStateHaveRemotePranswer indicates A local description of type "offer" has been successfully applied and a remote description of type "pranswer" has been successfully applied.
+	RTCSignalingStateHaveRemotePranswer
+
+	// RTCSignalingStateClosed indicates The RTCPeerConnection has been closed.
+	RTCSignalingStateClosed
+)
+
+func (t RTCSignalingState) String() string {
+	switch t {
+	case RTCSignalingStateStable:
+		return "stable"
+	case RTCSignalingStateHaveLocalOffer:
+		return "have-local-offer"
+	case RTCSignalingStateHaveRemoteOffer:
+		return "have-remote-offer"
+	case RTCSignalingStateHaveLocalPranswer:
+		return "have-local-pranswer"
+	case RTCSignalingStateHaveRemotePranswer:
+		return "have-remote-pranswer"
+	case RTCSignalingStateClosed:
+		return "closed"
+	default:
+		return "Unknown"
+	}
+}
+
+// RTCSdpType describes the type of an RTCSessionDescription
+type RTCSdpType int
+
+const (
+
+	// RTCSdpTypeOffer indicates that a description MUST be treated as an SDP offer.
+	RTCSdpTypeOffer RTCSdpType = iota + 1
+
+	// RTCSdpTypePranswer indicates that a description MUST be treated as an SDP answer, but not a final answer.
+	RTCSdpTypePranswer
+
+	// RTCSdpTypeAnswer indicates that a description MUST be treated as an SDP final answer, and the offer-answer exchange MUST be considered complete.
+	RTCSdpTypeAnswer
+
+	// RTCSdpTypeRollback indicates that a description MUST be treated as canceling the current SDP negotiation and moving the SDP offer and answer back to what it was in the previous stable state.
+	RTCSdpTypeRollback
+)
+
+func (t RTCSdpType) String() string {
+	switch t {
+	case RTCSdpTypeOffer:
+		return "offer"
+	case RTCSdpTypePranswer:
+		return "pranswer"
+	case RTCSdpTypeAnswer:
+		return "answer"
+	case RTCSdpTypeRollback:
+		return "rollback"
+	default:
+		return "Unknown"
+	}
+}
+
+// RTCSessionDescription is used to expose local and remote session descriptions.
+type RTCSessionDescription struct {
+	Typ RTCSdpType
+	Sdp string
+}
+
+// SetRemoteDescription sets the SessionDescription of the remote peer
+func (r *RTCPeerConnection) SetRemoteDescription(desc RTCSessionDescription) error {
+	if r.remoteDescription != nil {
+		return errors.Errorf("remoteDescription is already defined, SetRemoteDescription can only be called once")
+	}
+
+	r.currentRemoteDescription = &desc
+
+	r.remoteDescription = &sdp.SessionDescription{}
+
+	return r.remoteDescription.Unmarshal(desc.Sdp)
+}
+
+// RTCOfferOptions describes the options used to control the offer creation process
+type RTCOfferOptions struct {
+	VoiceActivityDetection bool
+	IceRestart             bool
+}
+
+// CreateOffer starts the RTCPeerConnection and generates the localDescription
+func (r *RTCPeerConnection) CreateOffer(options *RTCOfferOptions) (RTCSessionDescription, error) {
+	if options != nil {
+		panic("TODO handle options")
+	}
+	if r.IsClosed {
+		return RTCSessionDescription{}, &InvalidStateError{Err: ErrConnectionClosed}
+	}
+	useIdentity := r.idpLoginURL != nil
+	if useIdentity {
+		panic("TODO handle identity provider")
+	}
+
+	d := sdp.NewJSEPSessionDescription(
+		r.tlscfg.Fingerprint(),
+		useIdentity).
+		WithValueAttribute(sdp.AttrKeyGroup, "BUNDLE audio video") // TODO: Support BUNDLE
+
+	var streamlabels string
+	for _, tranceiver := range r.rtpTransceivers {
+		if tranceiver.Sender == nil ||
+			tranceiver.Sender.Track == nil {
+			continue
+		}
+		track := tranceiver.Sender.Track
+		cname := "poins"      // TODO: Support RTP streams synchronisation
+		steamlabel := "poins" // TODO: Support steam labels
+		codec, err := rtcMediaEngine.getCodec(track.PayloadType)
+		if err != nil {
+			return RTCSessionDescription{}, err
+		}
+		media := sdp.NewJSEPMediaDescription(track.Kind.String(), []string{}).
+			WithValueAttribute(sdp.AttrKeyConnectionSetup, sdp.ConnectionRoleActive.String()). // TODO: Support other connection types
+			WithValueAttribute(sdp.AttrKeyMID, tranceiver.Mid).
+			WithPropertyAttribute(tranceiver.Direction.String()).
+			WithICECredentials(r.iceAgent.Ufrag, r.iceAgent.Pwd).
+			WithPropertyAttribute(sdp.AttrKeyICELite).   // TODO: get ICE type from ICE Agent
+			WithPropertyAttribute(sdp.AttrKeyRtcpMux).   // TODO: support RTCP fallback
+			WithPropertyAttribute(sdp.AttrKeyRtcpRsize). // TODO: Support Reduced-Size RTCP?
+			WithCodec(
+				codec.PayloadType,
+				codec.Name,
+				codec.ClockRate,
+				codec.Channels,
+				codec.SdpFmtpLine,
+			).
+			WithMediaSource(track.Ssrc, cname, steamlabel, track.Label)
+		err = r.addICECandidates(media)
+		if err != nil {
+			return RTCSessionDescription{}, err
+		}
+		streamlabels = streamlabels + " " + steamlabel
+
+		d.WithMedia(media)
+	}
+
+	d.WithValueAttribute(sdp.AttrKeyMsidSemantic, " "+sdp.SemanticTokenWebRTCMediaStreams+streamlabels)
+
+	return RTCSessionDescription{
+		Typ: RTCSdpTypeOffer,
+		Sdp: d.Marshal(),
+	}, nil
+}
+
+// RTCAnswerOptions describes the options used to control the answer creation process
+type RTCAnswerOptions struct {
+	VoiceActivityDetection bool
+}
+
+// CreateAnswer starts the RTCPeerConnection and generates the localDescription
+func (r *RTCPeerConnection) CreateAnswer(options *RTCOfferOptions) (RTCSessionDescription, error) {
+	if options != nil {
+		panic("TODO handle options")
+	}
+	if r.IsClosed {
+		return RTCSessionDescription{}, &InvalidStateError{Err: ErrConnectionClosed}
+	}
+	useIdentity := r.idpLoginURL != nil
+	if useIdentity {
+		panic("TODO handle identity provider")
+	}
+
+	d := sdp.NewJSEPSessionDescription(
+		r.tlscfg.Fingerprint(),
+		useIdentity).
+		WithValueAttribute(sdp.AttrKeyGroup, "BUNDLE audio video") // TODO: Support BUNDLE
+
+	// TODO: Actually reply based on the offer (#21)
+	// TODO: Media lines should be in the same order as the offer
+
+	audioStreamLabels, err := r.addAnswerMedia(d, RTCRtpCodecTypeAudio)
+	if err != nil {
+		return RTCSessionDescription{}, err
+	}
+	videoStreamLabels, err := r.addAnswerMedia(d, RTCRtpCodecTypeVideo)
+	if err != nil {
+		return RTCSessionDescription{}, err
+	}
+	d.WithValueAttribute(sdp.AttrKeyMsidSemantic, " "+sdp.SemanticTokenWebRTCMediaStreams+audioStreamLabels+videoStreamLabels)
+
+	return RTCSessionDescription{
+		Typ: RTCSdpTypeAnswer,
+		Sdp: d.Marshal(),
+	}, nil
+}
+
+func (r *RTCPeerConnection) addAnswerMedia(d *sdp.SessionDescription, typ RTCRtpCodecType) (string, error) {
+	added := false
+
+	var streamlabels string
+	for _, tranceiver := range r.rtpTransceivers {
+		if tranceiver.Sender == nil ||
+			tranceiver.Sender.Track == nil ||
+			tranceiver.Sender.Track.Kind != typ {
+			continue
+		}
+		track := tranceiver.Sender.Track
+		cname := track.Label      // TODO: Support RTP streams synchronisation
+		steamlabel := track.Label // TODO: Support steam labels
+		codec, err := rtcMediaEngine.getCodec(track.PayloadType)
+		if err != nil {
+			return "", err
+		}
+		media := sdp.NewJSEPMediaDescription(track.Kind.String(), []string{}).
+			WithValueAttribute(sdp.AttrKeyConnectionSetup, sdp.ConnectionRoleActive.String()). // TODO: Support other connection types
+			WithValueAttribute(sdp.AttrKeyMID, tranceiver.Mid).
+			WithPropertyAttribute(tranceiver.Direction.String()).
+			WithICECredentials(r.iceAgent.Ufrag, r.iceAgent.Pwd).
+			WithPropertyAttribute(sdp.AttrKeyICELite).   // TODO: get ICE type from ICE Agent
+			WithPropertyAttribute(sdp.AttrKeyRtcpMux).   // TODO: support RTCP fallback
+			WithPropertyAttribute(sdp.AttrKeyRtcpRsize). // TODO: Support Reduced-Size RTCP?
+			WithCodec(
+				codec.PayloadType,
+				codec.Name,
+				codec.ClockRate,
+				codec.Channels,
+				codec.SdpFmtpLine,
+			).
+			WithMediaSource(track.Ssrc, cname, steamlabel, track.Label)
+
+		err = r.addICECandidates(media)
+		if err != nil {
+			return "", err
+		}
+		d.WithMedia(media)
+		streamlabels = streamlabels + " " + steamlabel
+		added = true
+	}
+
+	if !added {
+		// Add media line to advertise capabilities
+		media := sdp.NewJSEPMediaDescription(typ.String(), []string{}).
+			WithValueAttribute(sdp.AttrKeyConnectionSetup, sdp.ConnectionRoleActive.String()). // TODO: Support other connection types
+			WithValueAttribute(sdp.AttrKeyMID, typ.String()).
+			WithPropertyAttribute(RTCRtpTransceiverDirectionSendrecv.String()).
+			WithICECredentials(r.iceAgent.Ufrag, r.iceAgent.Pwd). // TODO: get credendials form ICE agent
+			WithPropertyAttribute(sdp.AttrKeyICELite).            // TODO: get ICE type from ICE Agent (#23)
+			WithPropertyAttribute(sdp.AttrKeyRtcpMux).            // TODO: support RTCP fallback
+			WithPropertyAttribute(sdp.AttrKeyRtcpRsize)           // TODO: Support Reduced-Size RTCP?
+
+		for _, codec := range rtcMediaEngine.getCodecsByKind(typ) {
+			media.WithCodec(
+				codec.PayloadType,
+				codec.Name,
+				codec.ClockRate,
+				codec.Channels,
+				codec.SdpFmtpLine,
+			)
+		}
+
+		err := r.addICECandidates(media)
+		if err != nil {
+			return "", err
+		}
+		d.WithMedia(media)
+	}
+
+	return streamlabels, nil
+}
+
+func (r *RTCPeerConnection) addICECandidates(d *sdp.MediaDescription) error {
+	r.portsLock.Lock()
+	defer r.portsLock.Unlock()
+
+	// TODO: Move gathering to ice.Agent
+
+	basePriority := uint16(rand.Uint32() & (1<<16 - 1))
+
+	for _, c := range ice.HostInterfaces() {
+		port, err := network.NewPort(c+":0", []byte(r.iceAgent.Pwd), r.tlscfg, r.generateChannel, r.iceStateChange)
+		if err != nil {
+			return err
+		}
+
+		d.WithCandidate(1, "udp", basePriority, c, port.ListeningAddr.Port, "host")
+
+		basePriority = basePriority + 1
+		r.ports = append(r.ports, port)
+	}
+
+	for _, servers := range r.iceAgent.Servers {
+		for _, server := range servers {
+			if server.Type != ice.ServerTypeSTUN {
+				continue
+			}
+			// TODO Do we want the timeout to be configurable?
+			proto := server.TransportType.String()
+			client, err := stun.NewClient(proto, fmt.Sprintf("%s:%d", server.Host, server.Port), time.Second*5)
+			if err != nil {
+				return errors.Wrapf(err, "Failed to create STUN client")
+			}
+			localAddr, ok := client.LocalAddr().(*net.UDPAddr)
+			if !ok {
+				return errors.Errorf("Failed to cast STUN client to UDPAddr")
+			}
+
+			resp, err := client.Request()
+			if err != nil {
+				return errors.Wrapf(err, "Failed to make STUN request")
+			}
+
+			if err := client.Close(); err != nil {
+				return errors.Wrapf(err, "Failed to close STUN client")
+			}
+
+			attr, ok := resp.GetOneAttribute(stun.AttrXORMappedAddress)
+			if !ok {
+				return errors.Errorf("Got respond from STUN server that did not contain XORAddress")
+			}
+
+			var addr stun.XorAddress
+			if err := addr.Unpack(resp, attr); err != nil {
+				return errors.Wrapf(err, "Failed to unpack STUN XorAddress response")
+			}
+
+			port, err := network.NewPort(fmt.Sprintf("0.0.0.0:%d", localAddr.Port), []byte(r.iceAgent.Pwd), r.tlscfg, r.generateChannel, r.iceStateChange)
+			if err != nil {
+				return errors.Wrapf(err, "Failed to build network/port")
+			}
+
+			d.WithCandidate(1, proto, basePriority, addr.IP.String(), localAddr.Port, "srflx")
+
+			basePriority = basePriority + 1
+			r.ports = append(r.ports, port)
+		}
+	}
+
+	d.WithPropertyAttribute("end-of-candidates") // TODO: Support full trickle-ice
+
+	return nil
+}

signaling_test.go

@@ -0,0 +1,26 @@
+package webrtc
+
+import (
+	"testing"
+
+	"github.com/pions/webrtc/internal/sdp"
+)
+
+func TestSetRemoteDescription(t *testing.T) {
+	testCases := []struct {
+		desc RTCSessionDescription
+	}{
+		{RTCSessionDescription{RTCSdpTypeOffer, sdp.NewJSEPSessionDescription("", false).Marshal()}},
+	}
+
+	for i, testCase := range testCases {
+		peerConn, err := New(RTCConfiguration{})
+		if err != nil {
+			t.Errorf("Case %d: got error: %v", i, err)
+		}
+		err = peerConn.SetRemoteDescription(testCase.desc)
+		if err != nil {
+			t.Errorf("Case %d: got error: %v", i, err)
+		}
+	}
+}