zishuo/webrtc
1
0
Fork 0
mirror of https://github.com/pion/webrtc.git synced 2025-11-01 11:12:50 +08:00
Code Issues Packages Projects Releases Wiki Activity

Finish RTCRtpReceiver and RTCRtpSender

Browse Source 
Allow sending of media via ORTC, examples will be added in the future.
But for now all of this state has been pulled out of RTCPeerConnection

Resolves #312
This commit is contained in:
Sean DuBois
2019-01-26 21:47:05 -08:00
parent 3c37b6c4b4
commit 56a5827d5c
7 changed files with 456 additions and 353 deletions
Download Patch File Download Diff File Expand all files Collapse all files

77
rtcdtlstransport.go
View File

@@ -13,9 +13,15 @@ import (
"github.com/pions/dtls"
"github.com/pions/webrtc/internal/mux"
"github.com/pions/webrtc/internal/srtp"
"github.com/pions/webrtc/pkg/rtcerr"
)
const (
srtpMasterKeyLen = 16
srtpMasterKeySaltLen = 14
)
// RTCDtlsTransport allows an application access to information about the DTLS
// transport over which RTP and RTCP packets are sent and received by
// RTCRtpSender and RTCRtpReceiver, as well other data such as SCTP packets sent
@@ -31,13 +37,23 @@ type RTCDtlsTransport struct {
// OnError func()
conn *dtls.Conn
srtpSession *srtp.SessionSRTP
srtpEndpoint *mux.Endpoint
srtcpSession *srtp.SessionSRTCP
srtcpEndpoint *mux.Endpoint
}
// NewRTCDtlsTransport creates a new RTCDtlsTransport.
// This constructor is part of the ORTC API. It is not
// meant to be used together with the basic WebRTC API.
func (api *API) NewRTCDtlsTransport(transport *RTCIceTransport, certificates []RTCCertificate) (*RTCDtlsTransport, error) {
t := &RTCDtlsTransport{iceTransport: transport}
t := &RTCDtlsTransport{
iceTransport: transport,
srtpSession: srtp.CreateSessionSRTP(),
srtcpSession: srtp.CreateSessionSRTCP(),
}
if len(certificates) > 0 {
now := time.Now()
@@ -132,7 +148,64 @@ func (t *RTCDtlsTransport) Start(remoteParameters RTCDtlsParameters) error {
fmt.Println("Warning: Certificate not checked")
}
return nil
return t.startSRTP(isClient)
}
// Start SRTP transport
// RTCDtlsTransport needs to own SRTP state to satisfy ORTC interfaces
func (t *RTCDtlsTransport) startSRTP(isClient bool) error {
keyingMaterial, err := t.conn.ExportKeyingMaterial([]byte("EXTRACTOR-dtls_srtp"), nil, (srtpMasterKeyLen*2)+(srtpMasterKeySaltLen*2))
if err != nil {
return err
}
t.srtpEndpoint = t.iceTransport.mux.NewEndpoint(mux.MatchSRTP)
t.srtcpEndpoint = t.iceTransport.mux.NewEndpoint(mux.MatchSRTCP)
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 !isClient {
err = t.srtpSession.Start(
serverWriteKey[0:16], serverWriteKey[16:],
clientWriteKey[0:16], clientWriteKey[16:],
srtp.ProtectionProfileAes128CmHmacSha1_80, t.srtpEndpoint,
)
if err == nil {
err = t.srtcpSession.Start(
serverWriteKey[0:16], serverWriteKey[16:],
clientWriteKey[0:16], clientWriteKey[16:],
srtp.ProtectionProfileAes128CmHmacSha1_80, t.srtcpEndpoint,
)
}
} else {
err = t.srtpSession.Start(
clientWriteKey[0:16], clientWriteKey[16:],
serverWriteKey[0:16], serverWriteKey[16:],
srtp.ProtectionProfileAes128CmHmacSha1_80, t.srtpEndpoint,
)
if err == nil {
err = t.srtcpSession.Start(
clientWriteKey[0:16], clientWriteKey[16:],
serverWriteKey[0:16], serverWriteKey[16:],
srtp.ProtectionProfileAes128CmHmacSha1_80, t.srtcpEndpoint,
)
}
}
return err
}
func (t *RTCDtlsTransport) validateFingerPrint(remoteParameters RTCDtlsParameters, remoteCert *x509.Certificate) error {

432
rtcpeerconnection.go
View File

@@ -5,22 +5,18 @@ import (
"crypto/ecdsa"
"crypto/elliptic"
"crypto/rand"
"encoding/binary"
"fmt"
"net"
"strconv"
"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"
)
@@ -33,9 +29,6 @@ const (
unknownStr = "unknown"
receiveMTU = 8192
srtpMasterKeyLen = 16
srtpMasterKeySaltLen = 14
)
// RTCPeerConnection represents a WebRTC connection that establishes a
@@ -118,12 +111,6 @@ type RTCPeerConnection struct {
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
}
@@ -133,7 +120,7 @@ func (api *API) NewRTCPeerConnection(configuration RTCConfiguration) (*RTCPeerCo
// 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{
pc := &RTCPeerConnection{
configuration: RTCConfiguration{
IceServers: []RTCIceServer{},
IceTransportPolicy: RTCIceTransportPolicyAll,
@@ -153,9 +140,6 @@ func (api *API) NewRTCPeerConnection(configuration RTCConfiguration) (*RTCPeerCo
ConnectionState: RTCPeerConnectionStateNew,
dataChannels: make(map[uint16]*RTCDataChannel),
srtpSession: srtp.CreateSessionSRTP(),
srtcpSession: srtp.CreateSessionSRTCP(),
api: api,
}
@@ -177,7 +161,18 @@ func (api *API) NewRTCPeerConnection(configuration RTCConfiguration) (*RTCPeerCo
return nil, err
}
return &pc, nil
// Create the ice transport
iceTransport := pc.createICETransport()
pc.iceTransport = iceTransport
// Create the DTLS transport
dtlsTransport, err := pc.createDTLSTransport()
if err != nil {
return nil, err
}
pc.dtlsTransport = dtlsTransport
return pc, nil
}
// initConfiguration defines validation of the specified RTCConfiguration and
@@ -744,10 +739,6 @@ func (pc *RTCPeerConnection) SetRemoteDescription(desc RTCSessionDescription) er
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() {
@@ -772,13 +763,6 @@ func (pc *RTCPeerConnection) SetRemoteDescription(desc RTCSessionDescription) er
}
}
// 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")
@@ -844,19 +828,12 @@ func (pc *RTCPeerConnection) SetRemoteDescription(desc RTCSessionDescription) er
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
if pc.onTrackHandler != nil {
pc.openSRTP()
} else {
pcLog.Warnf("OnTrack unset, unable to handle incoming media streams")
}
go pc.acceptSRTP()
go pc.drainSRTCP()
// Start sctp
err = pc.sctpTransport.Start(RTCSctpCapabilities{
MaxMessageSize: 0,
@@ -885,56 +862,65 @@ func (pc *RTCPeerConnection) openDataChannels() {
}
}
// 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
}
// openSRTP opens knows inbound SRTP streams from the RemoteDescription
func (pc *RTCPeerConnection) openSRTP() {
incomingSSRCes := map[uint32]RTCRtpCodecType{}
offset := 0
clientWriteKey := append([]byte{}, keyingMaterial[offset:offset+srtpMasterKeyLen]...)
offset += srtpMasterKeyLen
for _, media := range pc.RemoteDescription().parsed.MediaDescriptions {
for _, attr := range media.Attributes {
var codecType RTCRtpCodecType
if media.MediaName.Media == "audio" {
codecType = RTCRtpCodecTypeAudio
} else if media.MediaName.Media == "video" {
codecType = RTCRtpCodecTypeVideo
} else {
continue
}
serverWriteKey := append([]byte{}, keyingMaterial[offset:offset+srtpMasterKeyLen]...)
offset += srtpMasterKeyLen
if attr.Key == sdp.AttrKeySsrc {
ssrc, err := strconv.ParseUint(strings.Split(attr.Value, " ")[0], 10, 32)
if err != nil {
pcLog.Warnf("Failed to parse SSRC: %v", err)
continue
}
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,
)
incomingSSRCes[uint32(ssrc)] = codecType
}
}
}
return err
for i := range incomingSSRCes {
go func(ssrc uint32, codecType RTCRtpCodecType) {
receiver := NewRTCRtpReceiver(codecType, pc.dtlsTransport)
<-receiver.Receive(RTCRtpReceiveParameters{
encodings: RTCRtpDecodingParameters{
RTCRtpCodingParameters{SSRC: ssrc},
}})
sdpCodec, err := pc.CurrentLocalDescription.parsed.GetCodecForPayloadType(receiver.Track.PayloadType)
if err != nil {
pcLog.Warnf("no codec could be found in RemoteDescription for payloadType %d", receiver.Track.PayloadType)
return
}
codec, err := pc.api.mediaEngine.getCodecSDP(sdpCodec)
if err != nil {
pcLog.Warnf("codec %s in not registered", sdpCodec)
return
}
receiver.Track.Kind = codec.Type
receiver.Track.Codec = codec
pc.newRTCRtpTransceiver(
receiver,
nil,
RTCRtpTransceiverDirectionRecvonly,
)
pc.onTrack(receiver.Track)
}(i, incomingSSRCes[i])
}
}
// drainSRTCP pulls and discards RTCP packets that don't match any SRTP
@@ -943,24 +929,22 @@ func (pc *RTCPeerConnection) startSRTP(isOffer bool) error {
// and provides useful debugging messages
func (pc *RTCPeerConnection) drainSRTCP() {
for {
r, ssrc, err := pc.srtcpSession.AcceptStream()
r, ssrc, err := pc.dtlsTransport.srtcpSession.AcceptStream()
if err != nil {
pcLog.Warnf("Failed to accept RTCP %v \n", err)
return
}
go func() {
var rtcpPacket rtcp.Packet
rtcpBuf := make([]byte, receiveMTU)
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)
pcLog.Warnf("Failed to read, drainSRTCP done for: %v %d \n", err, ssrc)
return
}
rtcpPacket, _, err = rtcp.Unmarshal(rtcpBuf[:i])
rtcpPacket, _, err := rtcp.Unmarshal(rtcpBuf[:i])
if err != nil {
pcLog.Warnf("Failed to unmarshal RTCP packet, discarding: %v \n", err)
continue
@@ -972,77 +956,6 @@ func (pc *RTCPeerConnection) drainSRTCP() {
}
}
// 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.
@@ -1138,13 +1051,16 @@ func (pc *RTCPeerConnection) AddTrack(track *RTCTrack) (*RTCRtpSender, error) {
return nil, err
}
} else {
var receiver *RTCRtpReceiver
sender := newRTCRtpSender(track)
sender := NewRTCRtpSender(track, pc.dtlsTransport)
transceiver = pc.newRTCRtpTransceiver(
receiver,
nil,
sender,
RTCRtpTransceiverDirectionSendonly,
)
sender.Send(RTCRtpSendParameters{
encodings: RTCRtpEncodingParameters{
RTCRtpCodingParameters{SSRC: track.Ssrc, PayloadType: track.PayloadType},
}})
}
transceiver.Mid = track.Kind.String() // TODO: Mid generation
@@ -1332,7 +1248,7 @@ func (pc *RTCPeerConnection) SendRTCP(pkt rtcp.Packet) error {
return err
}
writeStream, err := pc.srtcpSession.OpenWriteStream()
writeStream, err := pc.dtlsTransport.srtcpSession.OpenWriteStream()
if err != nil {
return fmt.Errorf("SendRTCP failed to open WriteStream: %v", err)
}
@@ -1382,13 +1298,13 @@ func (pc *RTCPeerConnection) Close() error {
// 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.srtpSession.Close(); err != nil {
// closeErrs = append(closeErrs, err)
// }
if err := pc.srtcpSession.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 {
@@ -1426,44 +1342,6 @@ func flattenErrs(errs []error) error {
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
@@ -1568,20 +1446,17 @@ func (pc *RTCPeerConnection) addDataMediaSection(d *sdp.SessionDescription, midV
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)
}
// NewRawRTPTrack Creates a new RTCTrack
//
// See NewRTCSampleTrack for documentation
func (pc *RTCPeerConnection) NewRawRTPTrack(payloadType uint8, ssrc uint32, id, label string) (*RTCTrack, error) {
return NewRawRTPTrack(payloadType, ssrc, id, label)
}
func (pc *RTCPeerConnection) newRTCTrack(payloadType uint8, ssrc uint32, id, label string) (*RTCTrack, error) {
// NewRTCSampleTrack Creates a new RTCTrack
//
// See NewRTCSampleTrack for documentation
func (pc *RTCPeerConnection) NewRTCSampleTrack(payloadType uint8, id, label string) (*RTCTrack, error) {
codec, err := pc.api.mediaEngine.getCodec(payloadType)
if err != nil {
return nil, err
@@ -1589,118 +1464,7 @@ func (pc *RTCPeerConnection) newRTCTrack(payloadType uint8, ssrc uint32, id, lab
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)
return NewRTCSampleTrack(payloadType, id, label, codec)
}
// NewRTCTrack is used to create a new RTCTrack

6
rtcrtpreceiveparameters.go Normal file
View File

@@ -0,0 +1,6 @@
package webrtc
// RTCRtpReceiveParameters contains the RTP stack settings used by receivers
type RTCRtpReceiveParameters struct {
encodings RTCRtpDecodingParameters
}

113
rtcrtpreceiver.go
View File

@@ -1,14 +1,113 @@
package webrtc
import (
"github.com/pions/webrtc/pkg/rtcp"
"github.com/pions/webrtc/pkg/rtp"
)
// RTCRtpReceiver allows an application to inspect the receipt of a RTCTrack
type RTCRtpReceiver struct {
kind RTCRtpCodecType
transport *RTCDtlsTransport
hasRecv chan bool
Track *RTCTrack
// receiverTrack *RTCTrack
// receiverTransport
// receiverRtcpTransport
rtpOut chan *rtp.Packet
rtcpOut chan rtcp.Packet
}
// TODO: receiving side
// func newRTCRtpReceiver(kind, id string) {
//
// }
// NewRTCRtpReceiver constructs a new RTCRtpReceiver
func NewRTCRtpReceiver(kind RTCRtpCodecType, transport *RTCDtlsTransport) *RTCRtpReceiver {
return &RTCRtpReceiver{
kind: kind,
transport: transport,
hasRecv: make(chan bool),
rtpOut: make(chan *rtp.Packet, 15),
rtcpOut: make(chan rtcp.Packet, 15),
}
}
// Receive blocks until the RTCTrack is available
func (r *RTCRtpReceiver) Receive(parameters RTCRtpReceiveParameters) chan bool {
// TODO atomic only allow this to fire once
r.Track = &RTCTrack{
Kind: r.kind,
Ssrc: parameters.encodings.SSRC,
Packets: r.rtpOut,
RTCPPackets: r.rtcpOut,
}
// RTP ReadLoop
go func() {
payloadSet := false
defer func() {
if !payloadSet {
close(r.hasRecv)
}
}()
readStream, err := r.transport.srtpSession.OpenReadStream(parameters.encodings.SSRC)
if err != nil {
pcLog.Warnf("Failed to open RTCP ReadStream, RTCTrack done for: %v %d \n", err, parameters.encodings.SSRC)
return
}
readBuf := make([]byte, receiveMTU)
for {
rtpLen, err := readStream.Read(readBuf)
if err != nil {
pcLog.Warnf("Failed to read, RTCTrack done for: %v %d \n", err, parameters.encodings.SSRC)
return
}
var rtpPacket rtp.Packet
if err = rtpPacket.Unmarshal(append([]byte{}, readBuf[:rtpLen]...)); err != nil {
pcLog.Warnf("Failed to unmarshal RTP packet, discarding: %v \n", err)
continue
}
if !payloadSet {
r.Track.PayloadType = rtpPacket.PayloadType
payloadSet = true
close(r.hasRecv)
}
select {
case r.rtpOut <- &rtpPacket:
default:
}
}
}()
// RTCP ReadLoop
go func() {
readStream, err := r.transport.srtcpSession.OpenReadStream(parameters.encodings.SSRC)
if err != nil {
pcLog.Warnf("Failed to open RTCP ReadStream, RTCTrack done for: %v %d \n", err, parameters.encodings.SSRC)
return
}
readBuf := make([]byte, receiveMTU)
for {
rtcpLen, err := readStream.Read(readBuf)
if err != nil {
pcLog.Warnf("Failed to read, RTCTrack done for: %v %d \n", err, parameters.encodings.SSRC)
return
}
rtcpPacket, _, err := rtcp.Unmarshal(append([]byte{}, readBuf[:rtcpLen]...))
if err != nil {
pcLog.Warnf("Failed to unmarshal RTCP packet, discarding: %v \n", err)
continue
}
select {
case r.rtcpOut <- rtcpPacket:
default:
}
}
}()
return r.hasRecv
}

120
rtcrtpsender.go
View File

@@ -1,16 +1,122 @@
package webrtc
import (
"github.com/pions/webrtc/pkg/media"
"github.com/pions/webrtc/pkg/rtcp"
"github.com/pions/webrtc/pkg/rtp"
)
const rtpOutboundMTU = 1400
// 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
transport *RTCDtlsTransport
}
func newRTCRtpSender(track *RTCTrack) *RTCRtpSender {
s := &RTCRtpSender{
Track: track,
// NewRTCRtpSender constructs a new RTCRtpSender
func NewRTCRtpSender(track *RTCTrack, transport *RTCDtlsTransport) *RTCRtpSender {
return &RTCRtpSender{
Track: track,
transport: transport,
}
}
// Send Attempts to set the parameters controlling the sending of media.
func (r *RTCRtpSender) Send(parameters RTCRtpSendParameters) {
sampleInput := make(chan media.RTCSample, 15) // Is the buffering needed?
rawInput := make(chan *rtp.Packet, 15) // Is the buffering needed?
rtcpInput := make(chan rtcp.Packet, 15) // Is the buffering needed?
r.Track.Samples = sampleInput
r.Track.RawRTP = rawInput
r.Track.RTCPPackets = rtcpInput
if r.Track.isRawRTP {
close(r.Track.Samples)
go r.handleRawRTP(r.transport, rawInput)
} else {
close(r.Track.RawRTP)
go r.handleSampleRTP(r.transport, sampleInput)
}
go r.handleRTCP(r.transport, rtcpInput)
}
func (r *RTCRtpSender) handleRawRTP(transport *RTCDtlsTransport, rtpPackets chan *rtp.Packet) {
for {
p, ok := <-rtpPackets
if !ok {
return
}
r.sendRTP(p)
}
}
func (r *RTCRtpSender) handleSampleRTP(transport *RTCDtlsTransport, rtpPackets chan media.RTCSample) {
packetizer := rtp.NewPacketizer(
rtpOutboundMTU,
r.Track.PayloadType,
r.Track.Ssrc,
r.Track.Codec.Payloader,
rtp.NewRandomSequencer(),
r.Track.Codec.ClockRate,
)
for {
in, ok := <-rtpPackets
if !ok {
return
}
packets := packetizer.Packetize(in.Data, in.Samples)
for _, p := range packets {
r.sendRTP(p)
}
}
}
func (r *RTCRtpSender) handleRTCP(transport *RTCDtlsTransport, rtcpPackets chan rtcp.Packet) {
readStream, err := transport.srtcpSession.OpenReadStream(r.Track.Ssrc)
if err != nil {
pcLog.Warnf("Failed to open RTCP ReadStream, RTCTrack done for: %v %d \n", err, r.Track.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, r.Track.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:
}
}
}
func (r *RTCRtpSender) sendRTP(packet *rtp.Packet) {
writeStream, err := r.transport.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)
}
return s
}

6
rtcrtpsendparameters.go Normal file
View File

@@ -0,0 +1,6 @@
package webrtc
// RTCRtpSendParameters contains the RTP stack settings used by receivers
type RTCRtpSendParameters struct {
encodings RTCRtpEncodingParameters
}

55
rtctrack.go
View File

@@ -1,22 +1,71 @@
package webrtc
import (
"crypto/rand"
"encoding/binary"
"github.com/pions/webrtc/pkg/media"
"github.com/pions/webrtc/pkg/rtcp"
"github.com/pions/webrtc/pkg/rtp"
"github.com/pkg/errors"
)
// RTCTrack represents a track that is communicated
type RTCTrack struct {
isRawRTP bool
ID string
PayloadType uint8
Kind RTCRtpCodecType
Label string
Ssrc uint32
Codec *RTCRtpCodec
Packets <-chan *rtp.Packet
RTCPPackets <-chan rtcp.Packet
Samples chan<- media.RTCSample
RawRTP chan<- *rtp.Packet
isRawRTP bool
Samples chan<- media.RTCSample
RawRTP chan<- *rtp.Packet
}
// 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 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 &RTCTrack{
isRawRTP: true,
ID: id,
PayloadType: payloadType,
Label: label,
Ssrc: ssrc,
}, nil
}
// NewRTCSampleTrack initializes a new *RTCTrack configured to accept media.RTCSample
func NewRTCSampleTrack(payloadType uint8, id, label string, codec *RTCRtpCodec) (*RTCTrack, error) {
if codec == nil {
return nil, errors.New("codec supplied to NewRTCSampleTrack() must not be nil")
}
buf := make([]byte, 4)
if _, err := rand.Read(buf); err != nil {
return nil, errors.New("failed to generate random value")
}
return &RTCTrack{
isRawRTP: false,
ID: id,
PayloadType: payloadType,
Kind: codec.Type,
Label: label,
Ssrc: binary.LittleEndian.Uint32(buf),
Codec: codec,
}, nil
}