webrtc/rtcpeerconnection.go

// Package webrtc implements the WebRTC 1.0 as defined in W3C WebRTC specification document.
package webrtc

import (
	"crypto/ecdsa"
	"crypto/elliptic"
	"crypto/rand"
	"encoding/binary"
	"fmt"
	"net"
	"strings"
	"sync"
	"time"

	"github.com/pions/sdp"
	"github.com/pions/webrtc/internal/mux"
	"github.com/pions/webrtc/internal/srtp"
	"github.com/pions/webrtc/pkg/ice"
	"github.com/pions/webrtc/pkg/logging"
	"github.com/pions/webrtc/pkg/media"
	"github.com/pions/webrtc/pkg/rtcerr"
	"github.com/pions/webrtc/pkg/rtcp"
	"github.com/pions/webrtc/pkg/rtp"
	"github.com/pkg/errors"
)

var pcLog = logging.NewScopedLogger("pc")

const (
	// Unknown defines default public constant to use for "enum" like struct
	// comparisons when no value was defined.
	Unknown    = iota
	unknownStr = "unknown"

	receiveMTU = 8192

	srtpMasterKeyLen     = 16
	srtpMasterKeySaltLen = 14
)

// RTCPeerConnection represents a WebRTC connection that establishes a
// peer-to-peer communications with another RTCPeerConnection instance in a
// browser, or to another endpoint implementing the required protocols.
type RTCPeerConnection struct {
	sync.RWMutex

	configuration RTCConfiguration

	// CurrentLocalDescription represents the local description that was
	// successfully negotiated the last time the RTCPeerConnection transitioned
	// into the stable state plus any local candidates that have been generated
	// by the IceAgent since the offer or answer was created.
	CurrentLocalDescription *RTCSessionDescription

	// PendingLocalDescription represents a local description that is in the
	// process of being negotiated plus any local candidates that have been
	// generated by the IceAgent since the offer or answer was created. If the
	// RTCPeerConnection is in the stable state, the value is null.
	PendingLocalDescription *RTCSessionDescription

	// CurrentRemoteDescription represents the last remote description that was
	// successfully negotiated the last time the RTCPeerConnection transitioned
	// into the stable state plus any remote candidates that have been supplied
	// via AddIceCandidate() since the offer or answer was created.
	CurrentRemoteDescription *RTCSessionDescription

	// PendingRemoteDescription represents a remote description that is in the
	// process of being negotiated, complete with any remote candidates that
	// have been supplied via AddIceCandidate() since the offer or answer was
	// created. If the RTCPeerConnection is in the stable state, the value is
	// null.
	PendingRemoteDescription *RTCSessionDescription

	// SignalingState attribute returns the signaling state of the
	// RTCPeerConnection instance.
	SignalingState RTCSignalingState

	// IceGatheringState attribute returns the ICE gathering state of the
	// RTCPeerConnection instance.
	IceGatheringState RTCIceGatheringState // FIXME NOT-USED

	// IceConnectionState attribute returns the ICE connection state of the
	// RTCPeerConnection instance.
	// IceConnectionState RTCIceConnectionState  // FIXME SWAP-FOR-THIS
	IceConnectionState ice.ConnectionState // FIXME REMOVE

	// ConnectionState attribute returns the connection state of the
	// RTCPeerConnection instance.
	ConnectionState RTCPeerConnectionState

	idpLoginURL *string

	isClosed          bool
	negotiationNeeded bool

	lastOffer  string
	lastAnswer string

	rtpTransceivers []*RTCRtpTransceiver

	// DataChannels
	dataChannels map[uint16]*RTCDataChannel

	// OnNegotiationNeeded        func() // FIXME NOT-USED
	// OnIceCandidate             func() // FIXME NOT-USED
	// OnIceCandidateError        func() // FIXME NOT-USED

	// OnIceGatheringStateChange  func() // FIXME NOT-USED
	// OnConnectionStateChange    func() // FIXME NOT-USED

	onSignalingStateChangeHandler     func(RTCSignalingState)
	onICEConnectionStateChangeHandler func(ice.ConnectionState)
	onTrackHandler                    func(*RTCTrack)
	onDataChannelHandler              func(*RTCDataChannel)

	iceGatherer   *RTCIceGatherer
	iceTransport  *RTCIceTransport
	dtlsTransport *RTCDtlsTransport
	sctpTransport *RTCSctpTransport

	srtpSession  *srtp.SessionSRTP
	srtpEndpoint *mux.Endpoint

	srtcpSession  *srtp.SessionSRTCP
	srtcpEndpoint *mux.Endpoint

	// A reference to the associated API state used by this connection
	api *API
}

// New creates a new RTCPeerConfiguration with the provided configuration against the received API object
func (api *API) New(configuration RTCConfiguration) (*RTCPeerConnection, error) {
	// https://w3c.github.io/webrtc-pc/#constructor (Step #2)
	// Some variables defined explicitly despite their implicit zero values to
	// allow better readability to understand what is happening.
	pc := RTCPeerConnection{
		configuration: RTCConfiguration{
			IceServers:           []RTCIceServer{},
			IceTransportPolicy:   RTCIceTransportPolicyAll,
			BundlePolicy:         RTCBundlePolicyBalanced,
			RtcpMuxPolicy:        RTCRtcpMuxPolicyRequire,
			Certificates:         []RTCCertificate{},
			IceCandidatePoolSize: 0,
		},
		isClosed:          false,
		negotiationNeeded: false,
		lastOffer:         "",
		lastAnswer:        "",
		SignalingState:    RTCSignalingStateStable,
		// IceConnectionState: RTCIceConnectionStateNew, // FIXME SWAP-FOR-THIS
		IceConnectionState: ice.ConnectionStateNew, // FIXME REMOVE
		IceGatheringState:  RTCIceGatheringStateNew,
		ConnectionState:    RTCPeerConnectionStateNew,
		dataChannels:       make(map[uint16]*RTCDataChannel),

		srtpSession:  srtp.CreateSessionSRTP(),
		srtcpSession: srtp.CreateSessionSRTCP(),

		api: api,
	}

	var err error
	if err = pc.initConfiguration(configuration); err != nil {
		return nil, err
	}

	// For now we eagerly allocate and start the gatherer
	gatherer, err := pc.createIceGatherer()
	if err != nil {
		return nil, err
	}
	pc.iceGatherer = gatherer

	err = pc.gather()

	if err != nil {
		return nil, err
	}

	return &pc, nil
}

// New creates a new RTCPeerConfiguration with the provided configuration
func New(configuration RTCConfiguration) (*RTCPeerConnection, error) {
	return defaultAPI.New(configuration)
}

// initConfiguration defines validation of the specified RTCConfiguration and
// its assignment to the internal configuration variable. This function differs
// from its SetConfiguration counterpart because most of the checks do not
// include verification statements related to the existing state. Thus the
// function describes only minor verification of some the struct variables.
func (pc *RTCPeerConnection) initConfiguration(configuration RTCConfiguration) error {
	if configuration.PeerIdentity != "" {
		pc.configuration.PeerIdentity = configuration.PeerIdentity
	}

	// https://www.w3.org/TR/webrtc/#constructor (step #3)
	if len(configuration.Certificates) > 0 {
		now := time.Now()
		for _, x509Cert := range configuration.Certificates {
			if !x509Cert.Expires().IsZero() && now.After(x509Cert.Expires()) {
				return &rtcerr.InvalidAccessError{Err: ErrCertificateExpired}
			}
			pc.configuration.Certificates = append(pc.configuration.Certificates, x509Cert)
		}
	} else {
		sk, err := ecdsa.GenerateKey(elliptic.P256(), rand.Reader)
		if err != nil {
			return &rtcerr.UnknownError{Err: err}
		}
		certificate, err := GenerateCertificate(sk)
		if err != nil {
			return err
		}
		pc.configuration.Certificates = []RTCCertificate{*certificate}
	}

	if configuration.BundlePolicy != RTCBundlePolicy(Unknown) {
		pc.configuration.BundlePolicy = configuration.BundlePolicy
	}

	if configuration.RtcpMuxPolicy != RTCRtcpMuxPolicy(Unknown) {
		pc.configuration.RtcpMuxPolicy = configuration.RtcpMuxPolicy
	}

	if configuration.IceCandidatePoolSize != 0 {
		pc.configuration.IceCandidatePoolSize = configuration.IceCandidatePoolSize
	}

	if configuration.IceTransportPolicy != RTCIceTransportPolicy(Unknown) {
		pc.configuration.IceTransportPolicy = configuration.IceTransportPolicy
	}

	if len(configuration.IceServers) > 0 {
		for _, server := range configuration.IceServers {
			if _, err := server.validate(); err != nil {
				return err
			}
		}
		pc.configuration.IceServers = configuration.IceServers
	}
	return nil
}

// OnSignalingStateChange sets an event handler which is invoked when the
// peer connection's signaling state changes
func (pc *RTCPeerConnection) OnSignalingStateChange(f func(RTCSignalingState)) {
	pc.Lock()
	defer pc.Unlock()
	pc.onSignalingStateChangeHandler = f
}

func (pc *RTCPeerConnection) onSignalingStateChange(newState RTCSignalingState) (done chan struct{}) {
	pc.RLock()
	hdlr := pc.onSignalingStateChangeHandler
	pc.RUnlock()

	pcLog.Infof("signaling state changed to %s", newState)
	done = make(chan struct{})
	if hdlr == nil {
		close(done)
		return
	}

	go func() {
		hdlr(newState)
		close(done)
	}()

	return
}

// OnDataChannel sets an event handler which is invoked when a data
// channel message arrives from a remote peer.
func (pc *RTCPeerConnection) OnDataChannel(f func(*RTCDataChannel)) {
	pc.Lock()
	defer pc.Unlock()
	pc.onDataChannelHandler = f
}

// OnTrack sets an event handler which is called when remote track
// arrives from a remote peer.
func (pc *RTCPeerConnection) OnTrack(f func(*RTCTrack)) {
	pc.Lock()
	defer pc.Unlock()
	pc.onTrackHandler = f
}

func (pc *RTCPeerConnection) onTrack(t *RTCTrack) (done chan struct{}) {
	pc.RLock()
	hdlr := pc.onTrackHandler
	pc.RUnlock()

	pcLog.Debugf("got new track: %+v", t)
	done = make(chan struct{})
	if hdlr == nil || t == nil {
		close(done)
		return
	}

	go func() {
		hdlr(t)
		close(done)
	}()

	return
}

// OnICEConnectionStateChange sets an event handler which is called
// when an ICE connection state is changed.
func (pc *RTCPeerConnection) OnICEConnectionStateChange(f func(ice.ConnectionState)) {
	pc.Lock()
	defer pc.Unlock()
	pc.onICEConnectionStateChangeHandler = f
}

func (pc *RTCPeerConnection) onICEConnectionStateChange(cs ice.ConnectionState) (done chan struct{}) {
	pc.RLock()
	hdlr := pc.onICEConnectionStateChangeHandler
	pc.RUnlock()

	pcLog.Infof("ICE connection state changed: %s", cs)
	done = make(chan struct{})
	if hdlr == nil {
		close(done)
		return
	}

	go func() {
		hdlr(cs)
		close(done)
	}()

	return
}

// SetConfiguration updates the configuration of this RTCPeerConnection object.
func (pc *RTCPeerConnection) SetConfiguration(configuration RTCConfiguration) error {
	// https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-setconfiguration (step #2)
	if pc.isClosed {
		return &rtcerr.InvalidStateError{Err: ErrConnectionClosed}
	}

	// https://www.w3.org/TR/webrtc/#set-the-configuration (step #3)
	if configuration.PeerIdentity != "" {
		if configuration.PeerIdentity != pc.configuration.PeerIdentity {
			return &rtcerr.InvalidModificationError{Err: ErrModifyingPeerIdentity}
		}
		pc.configuration.PeerIdentity = configuration.PeerIdentity
	}

	// https://www.w3.org/TR/webrtc/#set-the-configuration (step #4)
	if len(configuration.Certificates) > 0 {
		if len(configuration.Certificates) != len(pc.configuration.Certificates) {
			return &rtcerr.InvalidModificationError{Err: ErrModifyingCertificates}
		}

		for i, certificate := range configuration.Certificates {
			if !pc.configuration.Certificates[i].Equals(certificate) {
				return &rtcerr.InvalidModificationError{Err: ErrModifyingCertificates}
			}
		}
		pc.configuration.Certificates = configuration.Certificates
	}

	// https://www.w3.org/TR/webrtc/#set-the-configuration (step #5)
	if configuration.BundlePolicy != RTCBundlePolicy(Unknown) {
		if configuration.BundlePolicy != pc.configuration.BundlePolicy {
			return &rtcerr.InvalidModificationError{Err: ErrModifyingBundlePolicy}
		}
		pc.configuration.BundlePolicy = configuration.BundlePolicy
	}

	// https://www.w3.org/TR/webrtc/#set-the-configuration (step #6)
	if configuration.RtcpMuxPolicy != RTCRtcpMuxPolicy(Unknown) {
		if configuration.RtcpMuxPolicy != pc.configuration.RtcpMuxPolicy {
			return &rtcerr.InvalidModificationError{Err: ErrModifyingRtcpMuxPolicy}
		}
		pc.configuration.RtcpMuxPolicy = configuration.RtcpMuxPolicy
	}

	// https://www.w3.org/TR/webrtc/#set-the-configuration (step #7)
	if configuration.IceCandidatePoolSize != 0 {
		if pc.configuration.IceCandidatePoolSize != configuration.IceCandidatePoolSize &&
			pc.LocalDescription() != nil {
			return &rtcerr.InvalidModificationError{Err: ErrModifyingIceCandidatePoolSize}
		}
		pc.configuration.IceCandidatePoolSize = configuration.IceCandidatePoolSize
	}

	// https://www.w3.org/TR/webrtc/#set-the-configuration (step #8)
	if configuration.IceTransportPolicy != RTCIceTransportPolicy(Unknown) {
		pc.configuration.IceTransportPolicy = configuration.IceTransportPolicy
	}

	// https://www.w3.org/TR/webrtc/#set-the-configuration (step #11)
	if len(configuration.IceServers) > 0 {
		// https://www.w3.org/TR/webrtc/#set-the-configuration (step #11.3)
		for _, server := range configuration.IceServers {
			if _, err := server.validate(); err != nil {
				return err
			}
		}
		pc.configuration.IceServers = configuration.IceServers
	}
	return nil
}

// GetConfiguration returns an RTCConfiguration object representing the current
// configuration of this RTCPeerConnection object. The returned object is a
// copy and direct mutation on it will not take affect until SetConfiguration
// has been called with RTCConfiguration passed as its only argument.
// https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-getconfiguration
func (pc *RTCPeerConnection) GetConfiguration() RTCConfiguration {
	return pc.configuration
}

// ------------------------------------------------------------------------
// --- FIXME - BELOW CODE NEEDS REVIEW/CLEANUP
// ------------------------------------------------------------------------

// CreateOffer starts the RTCPeerConnection and generates the localDescription
func (pc *RTCPeerConnection) CreateOffer(options *RTCOfferOptions) (RTCSessionDescription, error) {
	useIdentity := pc.idpLoginURL != nil
	if options != nil {
		return RTCSessionDescription{}, errors.Errorf("TODO handle options")
	} else if useIdentity {
		return RTCSessionDescription{}, errors.Errorf("TODO handle identity provider")
	} else if pc.isClosed {
		return RTCSessionDescription{}, &rtcerr.InvalidStateError{Err: ErrConnectionClosed}
	}

	d := sdp.NewJSEPSessionDescription(useIdentity)
	pc.addFingerprint(d)

	iceParams, err := pc.iceGatherer.GetLocalParameters()
	if err != nil {
		return RTCSessionDescription{}, err
	}

	candidates, err := pc.iceGatherer.GetLocalCandidates()
	if err != nil {
		return RTCSessionDescription{}, err
	}

	bundleValue := "BUNDLE"

	if pc.addRTPMediaSection(d, RTCRtpCodecTypeAudio, "audio", iceParams, RTCRtpTransceiverDirectionSendrecv, candidates, sdp.ConnectionRoleActpass) {
		bundleValue += " audio"
	}
	if pc.addRTPMediaSection(d, RTCRtpCodecTypeVideo, "video", iceParams, RTCRtpTransceiverDirectionSendrecv, candidates, sdp.ConnectionRoleActpass) {
		bundleValue += " video"
	}

	pc.addDataMediaSection(d, "data", iceParams, candidates, sdp.ConnectionRoleActpass)
	d = d.WithValueAttribute(sdp.AttrKeyGroup, bundleValue+" data")

	for _, m := range d.MediaDescriptions {
		m.WithPropertyAttribute("setup:actpass")
	}

	desc := RTCSessionDescription{
		Type:   RTCSdpTypeOffer,
		Sdp:    d.Marshal(),
		parsed: d,
	}
	pc.lastOffer = desc.Sdp

	// FIXME: This doesn't follow the JS API spec, but removing it
	// would mean our examples and existing code have to change
	if err := pc.SetLocalDescription(desc); err != nil {
		return RTCSessionDescription{}, err
	}
	return desc, nil
}

func (pc *RTCPeerConnection) createIceGatherer() (*RTCIceGatherer, error) {
	g, err := NewRTCIceGatherer(RTCIceGatherOptions{
		ICEServers: pc.configuration.IceServers,
		// TODO: GatherPolicy
	})
	if err != nil {
		return nil, err
	}

	return g, nil
}

func (pc *RTCPeerConnection) gather() error {
	return pc.iceGatherer.Gather()
}

func (pc *RTCPeerConnection) createICETransport() *RTCIceTransport {
	t := NewRTCIceTransport(pc.iceGatherer)

	t.OnConnectionStateChange(func(state RTCIceTransportState) {
		// We convert the state back to the ICE state to not brake the
		// existing public API at this point.
		iceState := state.toICE()
		pc.iceStateChange(iceState)
	})

	return t
}

func (pc *RTCPeerConnection) createDTLSTransport() (*RTCDtlsTransport, error) {
	dtlsTransport, err := NewRTCDtlsTransport(pc.iceTransport, pc.configuration.Certificates)
	return dtlsTransport, err
}

// CreateAnswer starts the RTCPeerConnection and generates the localDescription
func (pc *RTCPeerConnection) CreateAnswer(options *RTCAnswerOptions) (RTCSessionDescription, error) {
	useIdentity := pc.idpLoginURL != nil
	if options != nil {
		return RTCSessionDescription{}, errors.Errorf("TODO handle options")
	} else if useIdentity {
		return RTCSessionDescription{}, errors.Errorf("TODO handle identity provider")
	} else if pc.isClosed {
		return RTCSessionDescription{}, &rtcerr.InvalidStateError{Err: ErrConnectionClosed}
	}

	iceParams, err := pc.iceGatherer.GetLocalParameters()
	if err != nil {
		return RTCSessionDescription{}, err
	}

	candidates, err := pc.iceGatherer.GetLocalCandidates()
	if err != nil {
		return RTCSessionDescription{}, err
	}

	d := sdp.NewJSEPSessionDescription(useIdentity)
	pc.addFingerprint(d)

	bundleValue := "BUNDLE"
	for _, remoteMedia := range pc.RemoteDescription().parsed.MediaDescriptions {
		// TODO @trivigy better SDP parser
		var peerDirection RTCRtpTransceiverDirection
		midValue := ""
		for _, a := range remoteMedia.Attributes {
			if strings.HasPrefix(*a.String(), "mid") {
				midValue = (*a.String())[len("mid:"):]
			} else if strings.HasPrefix(*a.String(), "sendrecv") {
				peerDirection = RTCRtpTransceiverDirectionSendrecv
			} else if strings.HasPrefix(*a.String(), "sendonly") {
				peerDirection = RTCRtpTransceiverDirectionSendonly
			} else if strings.HasPrefix(*a.String(), "recvonly") {
				peerDirection = RTCRtpTransceiverDirectionRecvonly
			}
		}

		appendBundle := func() {
			bundleValue += " " + midValue
		}

		if strings.HasPrefix(*remoteMedia.MediaName.String(), "audio") {
			if pc.addRTPMediaSection(d, RTCRtpCodecTypeAudio, midValue, iceParams, peerDirection, candidates, sdp.ConnectionRoleActive) {
				appendBundle()
			}
		} else if strings.HasPrefix(*remoteMedia.MediaName.String(), "video") {
			if pc.addRTPMediaSection(d, RTCRtpCodecTypeVideo, midValue, iceParams, peerDirection, candidates, sdp.ConnectionRoleActive) {
				appendBundle()
			}
		} else if strings.HasPrefix(*remoteMedia.MediaName.String(), "application") {
			pc.addDataMediaSection(d, midValue, iceParams, candidates, sdp.ConnectionRoleActive)
			appendBundle()
		}
	}

	d = d.WithValueAttribute(sdp.AttrKeyGroup, bundleValue)

	desc := RTCSessionDescription{
		Type:   RTCSdpTypeAnswer,
		Sdp:    d.Marshal(),
		parsed: d,
	}
	pc.lastAnswer = desc.Sdp

	// FIXME: This doesn't follow the JS API spec, but removing it
	// would mean our examples and existing code have to change
	if err := pc.SetLocalDescription(desc); err != nil {
		return RTCSessionDescription{}, err
	}
	return desc, nil
}

// 4.4.1.6 Set the RTCSessionDescription
func (pc *RTCPeerConnection) setDescription(sd *RTCSessionDescription, op rtcStateChangeOp) error {
	if pc.isClosed {
		return &rtcerr.InvalidStateError{Err: ErrConnectionClosed}
	}

	cur := pc.SignalingState
	setLocal := rtcStateChangeOpSetLocal
	setRemote := rtcStateChangeOpSetRemote
	newSdpDoesNotMatchOffer := &rtcerr.InvalidModificationError{Err: errors.New("New sdp does not match previous offer")}
	newSdpDoesNotMatchAnswer := &rtcerr.InvalidModificationError{Err: errors.New("New sdp does not match previous answer")}

	var nextState RTCSignalingState
	var err error
	switch op {
	case setLocal:
		switch sd.Type {
		// stable->SetLocal(offer)->have-local-offer
		case RTCSdpTypeOffer:
			if sd.Sdp != pc.lastOffer {
				return newSdpDoesNotMatchOffer
			}
			nextState, err = checkNextSignalingState(cur, RTCSignalingStateHaveLocalOffer, setLocal, sd.Type)
			if err == nil {
				pc.PendingLocalDescription = sd
			}
		// have-remote-offer->SetLocal(answer)->stable
		// have-local-pranswer->SetLocal(answer)->stable
		case RTCSdpTypeAnswer:
			if sd.Sdp != pc.lastAnswer {
				return newSdpDoesNotMatchAnswer
			}
			nextState, err = checkNextSignalingState(cur, RTCSignalingStateStable, setLocal, sd.Type)
			if err == nil {
				pc.CurrentLocalDescription = sd
				pc.CurrentRemoteDescription = pc.PendingRemoteDescription
				pc.PendingRemoteDescription = nil
				pc.PendingLocalDescription = nil
			}
		case RTCSdpTypeRollback:
			nextState, err = checkNextSignalingState(cur, RTCSignalingStateStable, setLocal, sd.Type)
			if err == nil {
				pc.PendingLocalDescription = nil
			}
		// have-remote-offer->SetLocal(pranswer)->have-local-pranswer
		case RTCSdpTypePranswer:
			if sd.Sdp != pc.lastAnswer {
				return newSdpDoesNotMatchAnswer
			}
			nextState, err = checkNextSignalingState(cur, RTCSignalingStateHaveLocalPranswer, setLocal, sd.Type)
			if err == nil {
				pc.PendingLocalDescription = sd
			}
		default:
			return &rtcerr.OperationError{Err: fmt.Errorf("Invalid state change op: %s(%s)", op, sd.Type)}
		}
	case setRemote:
		switch sd.Type {
		// stable->SetRemote(offer)->have-remote-offer
		case RTCSdpTypeOffer:
			nextState, err = checkNextSignalingState(cur, RTCSignalingStateHaveRemoteOffer, setRemote, sd.Type)
			if err == nil {
				pc.PendingRemoteDescription = sd
			}
		// have-local-offer->SetRemote(answer)->stable
		// have-remote-pranswer->SetRemote(answer)->stable
		case RTCSdpTypeAnswer:
			nextState, err = checkNextSignalingState(cur, RTCSignalingStateStable, setRemote, sd.Type)
			if err == nil {
				pc.CurrentRemoteDescription = sd
				pc.CurrentLocalDescription = pc.PendingLocalDescription
				pc.PendingRemoteDescription = nil
				pc.PendingLocalDescription = nil
			}
		case RTCSdpTypeRollback:
			nextState, err = checkNextSignalingState(cur, RTCSignalingStateStable, setRemote, sd.Type)
			if err == nil {
				pc.PendingRemoteDescription = nil
			}
		// have-local-offer->SetRemote(pranswer)->have-remote-pranswer
		case RTCSdpTypePranswer:
			nextState, err = checkNextSignalingState(cur, RTCSignalingStateHaveRemotePranswer, setRemote, sd.Type)
			if err == nil {
				pc.PendingRemoteDescription = sd
			}
		default:
			return &rtcerr.OperationError{Err: fmt.Errorf("Invalid state change op: %s(%s)", op, sd.Type)}
		}
	default:
		return &rtcerr.OperationError{Err: fmt.Errorf("Unhandled state change op: %q", op)}
	}

	if err == nil {
		pc.SignalingState = nextState
		pc.onSignalingStateChange(nextState)
	}
	return err
}

// SetLocalDescription sets the SessionDescription of the local peer
func (pc *RTCPeerConnection) SetLocalDescription(desc RTCSessionDescription) error {
	if pc.isClosed {
		return &rtcerr.InvalidStateError{Err: ErrConnectionClosed}
	}

	// JSEP 5.4
	if desc.Sdp == "" {
		switch desc.Type {
		case RTCSdpTypeAnswer, RTCSdpTypePranswer:
			desc.Sdp = pc.lastAnswer
		case RTCSdpTypeOffer:
			desc.Sdp = pc.lastOffer
		default:
			return &rtcerr.InvalidModificationError{
				Err: fmt.Errorf("Invalid SDP type supplied to SetLocalDescription(): %s", desc.Type),
			}
		}
	}

	// TODO: Initiate ICE candidate gathering?

	desc.parsed = &sdp.SessionDescription{}
	if err := desc.parsed.Unmarshal(desc.Sdp); err != nil {
		return err
	}
	return pc.setDescription(&desc, rtcStateChangeOpSetLocal)
}

// LocalDescription returns PendingLocalDescription if it is not null and
// otherwise it returns CurrentLocalDescription. This property is used to
// determine if setLocalDescription has already been called.
// https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-localdescription
func (pc *RTCPeerConnection) LocalDescription() *RTCSessionDescription {
	if pc.PendingLocalDescription != nil {
		return pc.PendingLocalDescription
	}
	return pc.CurrentLocalDescription
}

// SetRemoteDescription sets the SessionDescription of the remote peer
func (pc *RTCPeerConnection) SetRemoteDescription(desc RTCSessionDescription) error {
	// FIXME: Remove this when renegotiation is supported
	if pc.CurrentRemoteDescription != nil {
		return errors.Errorf("remoteDescription is already defined, SetRemoteDescription can only be called once")
	}
	if pc.isClosed {
		return &rtcerr.InvalidStateError{Err: ErrConnectionClosed}
	}

	desc.parsed = &sdp.SessionDescription{}
	if err := desc.parsed.Unmarshal(desc.Sdp); err != nil {
		return err
	}
	if err := pc.setDescription(&desc, rtcStateChangeOpSetRemote); err != nil {
		return err
	}

	weOffer := true
	remoteUfrag := ""
	remotePwd := ""
	if desc.Type == RTCSdpTypeOffer {
		weOffer = false
	}

	// Create the ice transport
	iceTransport := pc.createICETransport()
	pc.iceTransport = iceTransport

	for _, m := range pc.RemoteDescription().parsed.MediaDescriptions {
		for _, a := range m.Attributes {
			if a.IsICECandidate() {
				sdpCandidate, err := a.ToICECandidate()
				if err != nil {
					return err
				}

				candidate, err := newRTCIceCandidateFromSDP(sdpCandidate)
				if err != nil {
					return err
				}

				if err = pc.iceTransport.AddRemoteCandidate(candidate); err != nil {
					return err
				}
			} else if strings.HasPrefix(*a.String(), "ice-ufrag") {
				remoteUfrag = (*a.String())[len("ice-ufrag:"):]
			} else if strings.HasPrefix(*a.String(), "ice-pwd") {
				remotePwd = (*a.String())[len("ice-pwd:"):]
			}
		}
	}

	// Create the DTLS transport
	dtlsTransport, err := pc.createDTLSTransport()
	if err != nil {
		return err
	}
	pc.dtlsTransport = dtlsTransport

	fingerprint, ok := desc.parsed.Attribute("fingerprint")
	if !ok {
		fingerprint, ok = desc.parsed.MediaDescriptions[0].Attribute("fingerprint")
		if !ok {
			return errors.New("could not find fingerprint")
		}
	}
	var fingerprintHash string
	parts := strings.Split(fingerprint, " ")
	if len(parts) != 2 {
		return errors.New("invalid fingerprint")
	}
	fingerprint = parts[1]
	fingerprintHash = parts[0]

	// Create the SCTP transport
	sctp := NewRTCSctpTransport(pc.dtlsTransport)
	pc.sctpTransport = sctp

	// Wire up the on datachannel handler
	sctp.OnDataChannel(func(d *RTCDataChannel) {
		pc.RLock()
		hdlr := pc.onDataChannelHandler
		pc.RUnlock()
		if hdlr != nil {
			hdlr(d)
		}
	})

	go func() {
		// Star the networking in a new routine since it will block until
		// the connection is actually established.

		// Start the ice transport
		iceRole := RTCIceRoleControlled
		if weOffer {
			iceRole = RTCIceRoleControlling
		}
		err := pc.iceTransport.Start(
			pc.iceGatherer,
			RTCIceParameters{
				UsernameFragment: remoteUfrag,
				Password:         remotePwd,
				IceLite:          false,
			},
			&iceRole,
		)

		if err != nil {
			// TODO: Handle error
			pcLog.Warnf("Failed to start manager: %s", err)
			return
		}

		// Start the dtls transport
		err = pc.dtlsTransport.Start(RTCDtlsParameters{
			Role:         RTCDtlsRoleAuto,
			Fingerprints: []RTCDtlsFingerprint{{Algorithm: fingerprintHash, Value: fingerprint}},
		})
		if err != nil {
			// TODO: Handle error
			pcLog.Warnf("Failed to start manager: %s", err)
			return
		}

		pc.srtpEndpoint = pc.iceTransport.mux.NewEndpoint(mux.MatchSRTP)
		pc.srtcpEndpoint = pc.iceTransport.mux.NewEndpoint(mux.MatchSRTCP)

		err = pc.startSRTP(weOffer)
		if err != nil {
			// TODO: Handle error
			pcLog.Warnf("Failed to start RTP: %s", err)
			return
		}

		go pc.acceptSRTP()
		go pc.drainSRTCP()

		// Start sctp
		err = pc.sctpTransport.Start(RTCSctpCapabilities{
			MaxMessageSize: 0,
		})
		if err != nil {
			// TODO: Handle error
			pcLog.Warnf("Failed to start SCTP: %s", err)
			return
		}

		// Open data channels that where created before signaling
		pc.openDataChannels()
	}()

	return nil
}

// openDataChannels opens the existing data channels
func (pc *RTCPeerConnection) openDataChannels() {
	for _, d := range pc.dataChannels {
		err := d.open(pc.sctpTransport)
		if err != nil {
			pcLog.Warnf("failed to open data channel: %s", err)
			continue
		}
	}
}

// startSRTP initializes all the cryptographic context needed for encrypted RTP
func (pc *RTCPeerConnection) startSRTP(isOffer bool) error {
	keyingMaterial, err := pc.dtlsTransport.conn.ExportKeyingMaterial([]byte("EXTRACTOR-dtls_srtp"), nil, (srtpMasterKeyLen*2)+(srtpMasterKeySaltLen*2))
	if err != nil {
		return err
	}

	offset := 0
	clientWriteKey := append([]byte{}, keyingMaterial[offset:offset+srtpMasterKeyLen]...)
	offset += srtpMasterKeyLen

	serverWriteKey := append([]byte{}, keyingMaterial[offset:offset+srtpMasterKeyLen]...)
	offset += srtpMasterKeyLen

	clientWriteKey = append(clientWriteKey, keyingMaterial[offset:offset+srtpMasterKeySaltLen]...)
	offset += srtpMasterKeySaltLen

	serverWriteKey = append(serverWriteKey, keyingMaterial[offset:offset+srtpMasterKeySaltLen]...)

	if isOffer {
		err = pc.srtpSession.Start(
			serverWriteKey[0:16], serverWriteKey[16:],
			clientWriteKey[0:16], clientWriteKey[16:],
			srtp.ProtectionProfileAes128CmHmacSha1_80, pc.srtpEndpoint,
		)

		if err == nil {
			err = pc.srtcpSession.Start(
				serverWriteKey[0:16], serverWriteKey[16:],
				clientWriteKey[0:16], clientWriteKey[16:],
				srtp.ProtectionProfileAes128CmHmacSha1_80, pc.srtcpEndpoint,
			)
		}
	} else {
		err = pc.srtpSession.Start(
			clientWriteKey[0:16], clientWriteKey[16:],
			serverWriteKey[0:16], serverWriteKey[16:],
			srtp.ProtectionProfileAes128CmHmacSha1_80, pc.srtpEndpoint,
		)

		if err == nil {
			err = pc.srtcpSession.Start(
				clientWriteKey[0:16], clientWriteKey[16:],
				serverWriteKey[0:16], serverWriteKey[16:],
				srtp.ProtectionProfileAes128CmHmacSha1_80, pc.srtcpEndpoint,
			)
		}
	}

	return err
}

// drainSRTCP pulls and discards RTCP packets that don't match any SRTP
// These could be sent to the user, but right now we don't provide an API
// to distribute orphaned RTCP messages. This is needed to make sure we don't block
// and provides useful debugging messages
func (pc *RTCPeerConnection) drainSRTCP() {
	for {
		r, ssrc, err := pc.srtcpSession.AcceptStream()
		if err != nil {
			pcLog.Warnf("Failed to accept RTCP %v \n", err)
			return
		}

		go func() {
			var rtcpPacket rtcp.Packet

			for {
				rtcpBuf := make([]byte, receiveMTU)
				i, err := r.Read(rtcpBuf)
				if err != nil {
					pcLog.Warnf("Failed to read, RTCTrack done for: %v %d \n", err, ssrc)
					return
				}

				rtcpPacket, _, err = rtcp.Unmarshal(rtcpBuf[:i])
				if err != nil {
					pcLog.Warnf("Failed to unmarshal RTCP packet, discarding: %v \n", err)
					continue
				}
				pcLog.Debugf("got RTCP: %+v", rtcpPacket)
			}
		}()

	}
}

// TODO: Move to RTCRTpSender?
func (pc *RTCPeerConnection) acceptSRTP() {
	for {
		r, ssrc, err := pc.srtpSession.AcceptStream()
		if err != nil {
			return
		}

		_, h, err := r.ReadRTP(make([]byte, receiveMTU))
		if err != nil {
			pcLog.Warnf("Failed to read, ignoring AcceptStream: %v \n", err)
			continue
		}

		rtpChannel, rtcpChannel, err := pc.generateChannel(h)
		if err != nil {
			pcLog.Warnf("Failed to create output channels, ignoring AcceptStream: %v \n", err)
			continue
		}

		// RTP
		go func() {
			for {
				rtpBuf := make([]byte, receiveMTU)
				rtpLen, h, err := r.ReadRTP(rtpBuf)
				if err != nil {
					pcLog.Warnf("Failed to read, RTCTrack done for: %v %d \n", err, ssrc)
					return
				}

				select {
				case rtpChannel <- &rtp.Packet{Header: *h, Raw: rtpBuf[:rtpLen], Payload: rtpBuf[h.PayloadOffset:rtpLen]}:
				default:
				}
			}
		}()

		// RTCP
		go func() {
			readStream, err := pc.srtcpSession.OpenReadStream(ssrc)
			if err != nil {
				pcLog.Warnf("Failed to open RTCP ReadStream, RTCTrack done for: %v %d \n", err, ssrc)
				return
			}

			for {
				var (
					rtcpPacket rtcp.Packet
					rtcpLen    int
				)
				rtcpBuf := make([]byte, receiveMTU)
				rtcpLen, err = readStream.Read(rtcpBuf)
				if err != nil {
					pcLog.Warnf("Failed to read, RTCTrack done for: %v %d \n", err, ssrc)
					return
				}

				rtcpPacket, _, err = rtcp.Unmarshal(rtcpBuf[:rtcpLen])
				if err != nil {
					pcLog.Warnf("Failed to unmarshal RTCP packet, discarding: %v \n", err)
					continue
				}
				select {
				case rtcpChannel <- rtcpPacket:
				default:
				}
			}
		}()
	}
}

// RemoteDescription returns PendingRemoteDescription if it is not null and
// otherwise it returns CurrentRemoteDescription. This property is used to
// determine if setRemoteDescription has already been called.
// https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-remotedescription
func (pc *RTCPeerConnection) RemoteDescription() *RTCSessionDescription {
	if pc.PendingRemoteDescription != nil {
		return pc.PendingRemoteDescription
	}
	return pc.CurrentRemoteDescription
}

// AddIceCandidate accepts an ICE candidate string and adds it
// to the existing set of candidates
func (pc *RTCPeerConnection) AddIceCandidate(s string) error {
	// TODO: AddIceCandidate should take RTCIceCandidateInit
	s = strings.TrimPrefix(s, "candidate:")

	attribute := sdp.NewAttribute("candidate", s)
	sdpCandidate, err := attribute.ToICECandidate()
	if err != nil {
		return err
	}

	candidate, err := newRTCIceCandidateFromSDP(sdpCandidate)
	if err != nil {
		return err
	}

	return pc.iceTransport.AddRemoteCandidate(candidate)
}

// ------------------------------------------------------------------------
// --- FIXME - BELOW CODE NEEDS RE-ORGANIZATION - https://w3c.github.io/webrtc-pc/#rtp-media-api
// ------------------------------------------------------------------------

// GetSenders returns the RTCRtpSender that are currently attached to this RTCPeerConnection
func (pc *RTCPeerConnection) GetSenders() []RTCRtpSender {
	result := make([]RTCRtpSender, len(pc.rtpTransceivers))
	for i, tranceiver := range pc.rtpTransceivers {
		result[i] = *tranceiver.Sender
	}
	return result
}

// GetReceivers returns the RTCRtpReceivers that are currently attached to this RTCPeerConnection
func (pc *RTCPeerConnection) GetReceivers() []RTCRtpReceiver {
	result := make([]RTCRtpReceiver, len(pc.rtpTransceivers))
	for i, tranceiver := range pc.rtpTransceivers {
		result[i] = *tranceiver.Receiver
	}
	return result
}

// GetTransceivers returns the RTCRtpTransceiver that are currently attached to this RTCPeerConnection
func (pc *RTCPeerConnection) GetTransceivers() []RTCRtpTransceiver {
	result := make([]RTCRtpTransceiver, len(pc.rtpTransceivers))
	for i, tranceiver := range pc.rtpTransceivers {
		result[i] = *tranceiver
	}
	return result
}

// AddTrack adds a RTCTrack to the RTCPeerConnection
func (pc *RTCPeerConnection) AddTrack(track *RTCTrack) (*RTCRtpSender, error) {
	if pc.isClosed {
		return nil, &rtcerr.InvalidStateError{Err: ErrConnectionClosed}
	}
	for _, transceiver := range pc.rtpTransceivers {
		if transceiver.Sender.Track == nil {
			continue
		}
		if track.ID == transceiver.Sender.Track.ID {
			return nil, &rtcerr.InvalidAccessError{Err: ErrExistingTrack}
		}
	}
	var transceiver *RTCRtpTransceiver
	for _, t := range pc.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 {
			transceiver = t
			break
		}
	}
	if transceiver != nil {
		if err := transceiver.setSendingTrack(track); err != nil {
			return nil, err
		}
	} else {
		var receiver *RTCRtpReceiver
		sender := newRTCRtpSender(track)
		transceiver = pc.newRTCRtpTransceiver(
			receiver,
			sender,
			RTCRtpTransceiverDirectionSendonly,
		)
	}

	transceiver.Mid = track.Kind.String() // TODO: Mid generation

	return transceiver.Sender, nil
}

// func (pc *RTCPeerConnection) RemoveTrack() {
// 	panic("not implemented yet") // FIXME NOT-IMPLEMENTED nolint
// }

// func (pc *RTCPeerConnection) AddTransceiver() RTCRtpTransceiver {
// 	panic("not implemented yet") // FIXME NOT-IMPLEMENTED nolint
// }

// ------------------------------------------------------------------------
// --- FIXME - BELOW CODE NEEDS RE-ORGANIZATION - https://w3c.github.io/webrtc-pc/#peer-to-peer-data-api
// ------------------------------------------------------------------------

// CreateDataChannel creates a new RTCDataChannel object with the given label
// and optional RTCDataChannelInit used to configure properties of the
// underlying channel such as data reliability.
func (pc *RTCPeerConnection) CreateDataChannel(label string, options *RTCDataChannelInit) (*RTCDataChannel, error) {
	// https://w3c.github.io/webrtc-pc/#peer-to-peer-data-api (Step #2)
	if pc.isClosed {
		return nil, &rtcerr.InvalidStateError{Err: ErrConnectionClosed}
	}

	// TODO: Add additional options once implemented. RTCDataChannelInit
	// implements all options. RTCDataChannelParameters implements the
	// options that actually have an effect at this point.
	params := &RTCDataChannelParameters{
		Label: label,
	}

	// https://w3c.github.io/webrtc-pc/#peer-to-peer-data-api (Step #19)
	if options != nil {
		if options.ID == nil {
			var err error
			if params.ID, err = pc.generateDataChannelID(true); err != nil {
				return nil, err
			}
		} else {
			params.ID = *options.ID
		}
	}

	// TODO: Re-enable validation of the parameters once they are implemented.
	/*
		// https://w3c.github.io/webrtc-pc/#peer-to-peer-data-api (Step #3)
		// Some variables defined explicitly despite their implicit zero values to
		// allow better readability to understand what is happening. Additionally,
		// some members are set to a non zero value default due to the default
		// definitions in https://w3c.github.io/webrtc-pc/#dom-rtcdatachannelinit
		// which are later overwriten by the options if any were specified.
		channel := RTCDataChannel{
			rtcPeerConnection: pc,
			// https://w3c.github.io/webrtc-pc/#peer-to-peer-data-api (Step #4)
			Label:             label,
			Ordered:           true,
			MaxPacketLifeTime: nil,
			MaxRetransmits:    nil,
			Protocol:          "",
			Negotiated:        false,
			ID:                nil,
			Priority:          RTCPriorityTypeLow,
			// https://w3c.github.io/webrtc-pc/#dfn-create-an-rtcdatachannel (Step #3)
			BufferedAmount: 0,
		}

		if options != nil {
			// https://w3c.github.io/webrtc-pc/#peer-to-peer-data-api (Step #7)
			if options.MaxPacketLifeTime != nil {
				channel.MaxPacketLifeTime = options.MaxPacketLifeTime
			}

			// https://w3c.github.io/webrtc-pc/#peer-to-peer-data-api (Step #8)
			if options.MaxRetransmits != nil {
				channel.MaxRetransmits = options.MaxRetransmits
			}

			// https://w3c.github.io/webrtc-pc/#peer-to-peer-data-api (Step #9)
			if options.Ordered != nil {
				channel.Ordered = *options.Ordered
			}

			// https://w3c.github.io/webrtc-pc/#peer-to-peer-data-api (Step #10)
			if options.Protocol != nil {
				channel.Protocol = *options.Protocol
			}

			// https://w3c.github.io/webrtc-pc/#peer-to-peer-da	ta-api (Step #12)
			if options.Negotiated != nil {
				channel.Negotiated = *options.Negotiated
			}

			// https://w3c.github.io/webrtc-pc/#peer-to-peer-data-api (Step #13)
			if options.ID != nil && channel.Negotiated {
				channel.ID = options.ID
			}

			// https://w3c.github.io/webrtc-pc/#peer-to-peer-data-api (Step #15)
			if options.Priority != nil {
				channel.Priority = *options.Priority
			}
		}

		// https://w3c.github.io/webrtc-pc/#peer-to-peer-data-api (Step #11)
		if len(channel.Protocol) > 65535 {
			return nil, &rtcerr.TypeError{Err: ErrStringSizeLimit}
		}

		// https://w3c.github.io/webrtc-pc/#peer-to-peer-data-api (Step #14)
		if channel.Negotiated && channel.ID == nil {
			return nil, &rtcerr.TypeError{Err: ErrNegotiatedWithoutID}
		}

		// https://w3c.github.io/webrtc-pc/#peer-to-peer-data-api (Step #16)
		if channel.MaxPacketLifeTime != nil && channel.MaxRetransmits != nil {
			return nil, &rtcerr.TypeError{Err: ErrRetransmitsOrPacketLifeTime}
		}

		// FIXME https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-createdatachannel (Step #17)


		// // https://w3c.github.io/webrtc-pc/#peer-to-peer-data-api (Step #18)
		if *channel.ID > 65534 {
			return nil, &rtcerr.TypeError{Err: ErrMaxDataChannelID}
		}

		if pc.sctpTransport.State == RTCSctpTransportStateConnected &&
			*channel.ID >= *pc.sctpTransport.MaxChannels {
			return nil, &rtcerr.OperationError{Err: ErrMaxDataChannelID}
		}
	*/

	d, err := pc.api.newRTCDataChannel(params)
	if err != nil {
		return nil, err
	}

	// Remember datachannel
	pc.dataChannels[params.ID] = d

	// Open if networking already started
	if pc.sctpTransport != nil {
		err = d.open(pc.sctpTransport)
		if err != nil {
			return nil, err
		}
	}

	return d, nil
}

func (pc *RTCPeerConnection) generateDataChannelID(client bool) (uint16, error) {
	var id uint16
	if !client {
		id++
	}

	max := sctpMaxChannels
	if pc.sctpTransport != nil {
		max = *pc.sctpTransport.MaxChannels
	}

	for ; id < max-1; id += 2 {
		_, ok := pc.dataChannels[id]
		if !ok {
			return id, nil
		}
	}
	return 0, &rtcerr.OperationError{Err: ErrMaxDataChannelID}
}

// SetIdentityProvider is used to configure an identity provider to generate identity assertions
func (pc *RTCPeerConnection) SetIdentityProvider(provider string) error {
	return errors.Errorf("TODO SetIdentityProvider")
}

// SendRTCP sends a user provided RTCP packet to the connected peer
// If no peer is connected the packet is discarded
func (pc *RTCPeerConnection) SendRTCP(pkt rtcp.Packet) error {
	raw, err := pkt.Marshal()
	if err != nil {
		return err
	}

	writeStream, err := pc.srtcpSession.OpenWriteStream()
	if err != nil {
		return fmt.Errorf("SendRTCP failed to open WriteStream: %v", err)
	}

	if _, err := writeStream.Write(raw); err != nil {
		return fmt.Errorf("SendRTCP failed to write: %v", err)
	}
	return nil
}

// Close ends the RTCPeerConnection
func (pc *RTCPeerConnection) Close() error {
	// https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-close (step #2)
	if pc.isClosed {
		return nil
	}

	for _, t := range pc.rtpTransceivers {
		if track := t.Sender.Track; track != nil {
			if track.isRawRTP {
				close(track.RawRTP)
			} else {
				close(track.Samples)
			}
		}
	}

	// https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-close (step #3)
	pc.isClosed = true

	// https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-close (step #4)
	pc.SignalingState = RTCSignalingStateClosed

	// https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-close (step #11)
	// pc.IceConnectionState = RTCIceConnectionStateClosed
	pc.iceStateChange(ice.ConnectionStateClosed) // FIXME REMOVE

	// https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-close (step #12)
	pc.ConnectionState = RTCPeerConnectionStateClosed

	// Try closing everything and collect the errors
	var closeErrs []error

	// Shutdown strategy:
	// 1. All Conn close by closing their underlying Conn.
	// 2. A Mux stops this chain. It won't close the underlying
	//    Conn if one of the endpoints is closed down. To
	//    continue the chain the Mux has to be closed.

	if err := pc.srtpSession.Close(); err != nil {
		closeErrs = append(closeErrs, err)
	}

	if err := pc.srtcpSession.Close(); err != nil {
		closeErrs = append(closeErrs, err)
	}

	if pc.sctpTransport != nil {
		if err := pc.sctpTransport.Stop(); err != nil {
			closeErrs = append(closeErrs, err)
		}
	}

	// TODO: Close DTLS?

	if pc.iceTransport != nil {
		if err := pc.iceTransport.Stop(); err != nil {
			closeErrs = append(closeErrs, err)
		}
	}

	// TODO: Figure out stopping ICE transport & Gatherer independently.
	// pc.iceGatherer()

	return flattenErrs(closeErrs)
}

func flattenErrs(errs []error) error {
	var errstrings []string

	for _, err := range errs {
		if err != nil {
			errstrings = append(errstrings, err.Error())
		}
	}

	if len(errstrings) == 0 {
		return nil
	}

	return fmt.Errorf(strings.Join(errstrings, "\n"))
}

/* Everything below is private */
func (pc *RTCPeerConnection) generateChannel(h *rtp.Header) (chan *rtp.Packet, chan rtcp.Packet, error) {
	pc.RLock()
	if pc.onTrackHandler == nil {
		pc.RUnlock()
		return nil, nil, fmt.Errorf("OnTrack unset, unable to handle incoming")
	}
	pc.RUnlock()

	sdpCodec, err := pc.CurrentLocalDescription.parsed.GetCodecForPayloadType(h.PayloadType)
	if err != nil {
		return nil, nil, fmt.Errorf("no codec could be found in RemoteDescription for payloadType %d", h.PayloadType)
	}

	codec, err := pc.api.mediaEngine.getCodecSDP(sdpCodec)
	if err != nil {
		return nil, nil, fmt.Errorf("codec %s in not registered", sdpCodec)
	}

	rtpTransport := make(chan *rtp.Packet, 15)
	rtcpTransport := make(chan rtcp.Packet, 15)

	track := &RTCTrack{
		PayloadType: h.PayloadType,
		Kind:        codec.Type,
		ID:          "0", // TODO extract from remoteDescription
		Label:       "",  // TODO extract from remoteDescription
		Ssrc:        h.SSRC,
		Codec:       codec,
		Packets:     rtpTransport,
		RTCPPackets: rtcpTransport,
	}

	// TODO: Register the receiving Track
	pc.onTrack(track)
	return rtpTransport, rtcpTransport, nil
}

func (pc *RTCPeerConnection) iceStateChange(newState ice.ConnectionState) {
	pc.Lock()
	pc.IceConnectionState = newState
	pc.Unlock()

	pc.onICEConnectionStateChange(newState)
}

func localDirection(weSend bool, peerDirection RTCRtpTransceiverDirection) RTCRtpTransceiverDirection {
	theySend := (peerDirection == RTCRtpTransceiverDirectionSendrecv || peerDirection == RTCRtpTransceiverDirectionSendonly)
	if weSend && theySend {
		return RTCRtpTransceiverDirectionSendrecv
	} else if weSend && !theySend {
		return RTCRtpTransceiverDirectionSendonly
	} else if !weSend && theySend {
		return RTCRtpTransceiverDirectionRecvonly
	}

	return RTCRtpTransceiverDirectionInactive
}

func (pc *RTCPeerConnection) addFingerprint(d *sdp.SessionDescription) {
	// TODO: Handle multiple certificates
	for _, fingerprint := range pc.configuration.Certificates[0].GetFingerprints() {
		d.WithFingerprint(fingerprint.Algorithm, strings.ToUpper(fingerprint.Value))
	}
}

func (pc *RTCPeerConnection) addRTPMediaSection(d *sdp.SessionDescription, codecType RTCRtpCodecType, midValue string, iceParams RTCIceParameters, peerDirection RTCRtpTransceiverDirection, candidates []RTCIceCandidate, dtlsRole sdp.ConnectionRole) bool {
	if codecs := pc.api.mediaEngine.getCodecsByKind(codecType); len(codecs) == 0 {
		return false
	}
	media := sdp.NewJSEPMediaDescription(codecType.String(), []string{}).
		WithValueAttribute(sdp.AttrKeyConnectionSetup, dtlsRole.String()). // TODO: Support other connection types
		WithValueAttribute(sdp.AttrKeyMID, midValue).
		WithICECredentials(iceParams.UsernameFragment, iceParams.Password).
		WithPropertyAttribute(sdp.AttrKeyRtcpMux).  // TODO: support RTCP fallback
		WithPropertyAttribute(sdp.AttrKeyRtcpRsize) // TODO: Support Reduced-Size RTCP?

	for _, codec := range pc.api.mediaEngine.getCodecsByKind(codecType) {
		media.WithCodec(codec.PayloadType, codec.Name, codec.ClockRate, codec.Channels, codec.SdpFmtpLine)
	}

	weSend := false
	for _, transceiver := range pc.rtpTransceivers {
		if transceiver.Sender == nil ||
			transceiver.Sender.Track == nil ||
			transceiver.Sender.Track.Kind != codecType {
			continue
		}
		weSend = true
		track := transceiver.Sender.Track
		media = media.WithMediaSource(track.Ssrc, track.Label /* cname */, track.Label /* streamLabel */, track.Label)
	}
	media = media.WithPropertyAttribute(localDirection(weSend, peerDirection).String())

	for _, c := range candidates {
		sdpCandidate := c.toSDP()
		sdpCandidate.ExtensionAttributes = append(sdpCandidate.ExtensionAttributes, sdp.ICECandidateAttribute{Key: "generation", Value: "0"})
		sdpCandidate.Component = 1
		media.WithICECandidate(sdpCandidate)
		sdpCandidate.Component = 2
		media.WithICECandidate(sdpCandidate)
	}
	media.WithPropertyAttribute("end-of-candidates")
	d.WithMedia(media)
	return true
}

func (pc *RTCPeerConnection) addDataMediaSection(d *sdp.SessionDescription, midValue string, iceParams RTCIceParameters, candidates []RTCIceCandidate, dtlsRole sdp.ConnectionRole) {
	media := (&sdp.MediaDescription{
		MediaName: sdp.MediaName{
			Media:   "application",
			Port:    sdp.RangedPort{Value: 9},
			Protos:  []string{"DTLS", "SCTP"},
			Formats: []string{"5000"},
		},
		ConnectionInformation: &sdp.ConnectionInformation{
			NetworkType: "IN",
			AddressType: "IP4",
			Address: &sdp.Address{
				IP: net.ParseIP("0.0.0.0"),
			},
		},
	}).
		WithValueAttribute(sdp.AttrKeyConnectionSetup, dtlsRole.String()). // TODO: Support other connection types
		WithValueAttribute(sdp.AttrKeyMID, midValue).
		WithPropertyAttribute(RTCRtpTransceiverDirectionSendrecv.String()).
		WithPropertyAttribute("sctpmap:5000 webrtc-datachannel 1024").
		WithICECredentials(iceParams.UsernameFragment, iceParams.Password)

	for _, c := range candidates {
		sdpCandidate := c.toSDP()
		sdpCandidate.ExtensionAttributes = append(sdpCandidate.ExtensionAttributes, sdp.ICECandidateAttribute{Key: "generation", Value: "0"})
		sdpCandidate.Component = 1
		media.WithICECandidate(sdpCandidate)
		sdpCandidate.Component = 2
		media.WithICECandidate(sdpCandidate)
	}
	media.WithPropertyAttribute("end-of-candidates")

	d.WithMedia(media)
}

// TODO RTCRtpSender
func (pc *RTCPeerConnection) sendRTP(packet *rtp.Packet) {
	writeStream, err := pc.srtpSession.OpenWriteStream()
	if err != nil {
		pcLog.Warnf("SendRTP failed to open WriteStream: %v", err)
		return
	}

	if _, err := writeStream.WriteRTP(&packet.Header, packet.Payload); err != nil {
		pcLog.Warnf("SendRTP failed to write: %v", err)
	}
}

func (pc *RTCPeerConnection) newRTCTrack(payloadType uint8, ssrc uint32, id, label string) (*RTCTrack, error) {
	codec, err := pc.api.mediaEngine.getCodec(payloadType)
	if err != nil {
		return nil, err
	} else if codec.Payloader == nil {
		return nil, errors.New("codec payloader not set")
	}

	trackInput := make(chan media.RTCSample, 15) // Is the buffering needed?
	rawPackets := make(chan *rtp.Packet, 15)     // Is the buffering needed?
	rtcpPackets := make(chan rtcp.Packet, 15)    // Is the buffering needed?
	isRawRTP := false

	if ssrc == 0 {
		buf := make([]byte, 4)
		_, err = rand.Read(buf)
		if err != nil {
			return nil, errors.New("failed to generate random value")
		}
		ssrc = binary.LittleEndian.Uint32(buf)

		go func() {
			packetizer := rtp.NewPacketizer(
				1400,
				payloadType,
				ssrc,
				codec.Payloader,
				rtp.NewRandomSequencer(),
				codec.ClockRate,
			)

			for {
				in, ok := <-trackInput
				if !ok {
					return
				}
				packets := packetizer.Packetize(in.Data, in.Samples)
				for _, p := range packets {
					pc.sendRTP(p)
				}
			}
		}()
		close(rawPackets)
	} else {
		// If SSRC is not 0, then we are working with an established RTP stream
		// and need to accept raw RTP packets for forwarding.
		isRawRTP = true
		go func() {
			for {
				p, ok := <-rawPackets
				if !ok {
					return
				}

				pc.sendRTP(p)
			}
		}()
		close(trackInput)
	}

	t := &RTCTrack{
		isRawRTP:    isRawRTP,
		PayloadType: payloadType,
		Kind:        codec.Type,
		ID:          id,
		Label:       label,
		Ssrc:        ssrc,
		Codec:       codec,
		RTCPPackets: rtcpPackets,
		Samples:     trackInput,
		RawRTP:      rawPackets,
	}

	// Inbound RTCP
	go func() {
		readStream, err := pc.srtcpSession.OpenReadStream(ssrc)
		if err != nil {
			pcLog.Warnf("Failed to open RTCP ReadStream, RTCTrack done for: %v %d \n", err, ssrc)
			return
		}

		var rtcpPacket rtcp.Packet

		for {
			rtcpBuf := make([]byte, receiveMTU)
			i, err := readStream.Read(rtcpBuf)
			if err != nil {
				pcLog.Warnf("Failed to read, RTCTrack done for: %v %d \n", err, ssrc)
				return
			}

			rtcpPacket, _, err = rtcp.Unmarshal(rtcpBuf[:i])
			if err != nil {
				pcLog.Warnf("Failed to unmarshal RTCP packet, discarding: %v \n", err)
				continue
			}
			select {
			case rtcpPackets <- rtcpPacket:
			default:
			}
		}
	}()

	return t, nil
}

// NewRawRTPTrack initializes a new *RTCTrack configured to accept raw *rtp.Packet
//
// NB: If the source RTP stream is being broadcast to multiple tracks, each track
// must receive its own copies of the source packets in order to avoid packet corruption.
func (pc *RTCPeerConnection) NewRawRTPTrack(payloadType uint8, ssrc uint32, id, label string) (*RTCTrack, error) {
	if ssrc == 0 {
		return nil, errors.New("SSRC supplied to NewRawRTPTrack() must be non-zero")
	}
	return pc.newRTCTrack(payloadType, ssrc, id, label)
}

// NewRTCSampleTrack initializes a new *RTCTrack configured to accept media.RTCSample
func (pc *RTCPeerConnection) NewRTCSampleTrack(payloadType uint8, id, label string) (*RTCTrack, error) {
	return pc.newRTCTrack(payloadType, 0, id, label)
}

// NewRTCTrack is used to create a new RTCTrack
//
// Deprecated: Use NewRTCSampleTrack() instead
func (pc *RTCPeerConnection) NewRTCTrack(payloadType uint8, id, label string) (*RTCTrack, error) {
	return pc.NewRTCSampleTrack(payloadType, id, label)
}

func (pc *RTCPeerConnection) newRTCRtpTransceiver(
	receiver *RTCRtpReceiver,
	sender *RTCRtpSender,
	direction RTCRtpTransceiverDirection,
) *RTCRtpTransceiver {

	t := &RTCRtpTransceiver{
		Receiver:  receiver,
		Sender:    sender,
		Direction: direction,
	}
	pc.rtpTransceivers = append(pc.rtpTransceivers, t)
	return t
}