Initial commit
This commit is contained in:
Alexey Khit
@@ -0,0 +1,104 @@
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package webrtc
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import (
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"github.com/pion/interceptor"
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"github.com/pion/webrtc/v3"
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"net"
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)
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func NewAPI(address string) (*webrtc.API, error) {
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// for debug logs add to env: `PION_LOG_DEBUG=all`
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m := &webrtc.MediaEngine{}
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//if err := m.RegisterDefaultCodecs(); err != nil {
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// return nil, err
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//}
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if err := RegisterDefaultCodecs(m); err != nil {
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return nil, err
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}
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i := &interceptor.Registry{}
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if err := webrtc.RegisterDefaultInterceptors(m, i); err != nil {
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return nil, err
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}
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if address == "" {
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return webrtc.NewAPI(
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webrtc.WithMediaEngine(m),
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webrtc.WithInterceptorRegistry(i),
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), nil
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}
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ln, err := net.Listen("tcp", address)
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if err != nil {
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return webrtc.NewAPI(
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webrtc.WithMediaEngine(m),
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webrtc.WithInterceptorRegistry(i),
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), err
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}
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s := webrtc.SettingEngine{
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//LoggerFactory: customLoggerFactory{},
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}
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s.SetNetworkTypes([]webrtc.NetworkType{
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webrtc.NetworkTypeUDP4, webrtc.NetworkTypeUDP6,
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webrtc.NetworkTypeTCP4, webrtc.NetworkTypeTCP6,
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})
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tcpMux := webrtc.NewICETCPMux(nil, ln, 8)
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s.SetICETCPMux(tcpMux)
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return webrtc.NewAPI(
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webrtc.WithMediaEngine(m),
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webrtc.WithInterceptorRegistry(i),
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webrtc.WithSettingEngine(s),
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), nil
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}
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func RegisterDefaultCodecs(m *webrtc.MediaEngine) error {
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for _, codec := range []webrtc.RTPCodecParameters{
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{webrtc.MimeTypeOpus, 48000, 2, "minptime=10;useinbandfec=1", nil},
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PayloadType: 101, //111,
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}, {
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RTPCodecCapability: webrtc.RTPCodecCapability{webrtc.MimeTypePCMU, 8000, 0, "", nil},
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PayloadType: 0,
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}, {
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RTPCodecCapability: webrtc.RTPCodecCapability{webrtc.MimeTypePCMA, 8000, 0, "", nil},
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PayloadType: 8,
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},
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} {
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if err := m.RegisterCodec(codec, webrtc.RTPCodecTypeAudio); err != nil {
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return err
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}
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}
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videoRTCPFeedback := []webrtc.RTCPFeedback{{"goog-remb", ""}, {"ccm", "fir"}, {"nack", ""}, {"nack", "pli"}}
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for _, codec := range []webrtc.RTPCodecParameters{
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// macOS Google Chrome 103.0.5060.134
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{webrtc.MimeTypeH264, 90000, 0, "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f", videoRTCPFeedback},
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PayloadType: 96, //102,
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}, {
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RTPCodecCapability: webrtc.RTPCodecCapability{webrtc.MimeTypeH264, 90000, 0, "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f", videoRTCPFeedback},
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PayloadType: 97, //125,
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}, {
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RTPCodecCapability: webrtc.RTPCodecCapability{webrtc.MimeTypeH264, 90000, 0, "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032", videoRTCPFeedback},
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PayloadType: 98, //123,
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},
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// macOS Safari 15.1
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{webrtc.MimeTypeH264, 90000, 0, "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640c1f", videoRTCPFeedback},
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PayloadType: 99,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{webrtc.MimeTypeH265, 90000, 0, "", videoRTCPFeedback},
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PayloadType: 100,
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},
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} {
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if err := m.RegisterCodec(codec, webrtc.RTPCodecTypeVideo); err != nil {
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return err
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}
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}
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return nil
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}
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@@ -0,0 +1,195 @@
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package webrtc
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import (
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"github.com/AlexxIT/go2rtc/pkg/streamer"
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"github.com/pion/webrtc/v3"
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)
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const (
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MsgTypeOffer = "webrtc/offer"
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MsgTypeOfferComplete = "webrtc/offer-complete"
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MsgTypeAnswer = "webrtc/answer"
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MsgTypeCandidate = "webrtc/candidate"
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)
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type Conn struct {
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streamer.Element
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UserAgent string
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Conn *webrtc.PeerConnection
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medias []*streamer.Media
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tracks []*streamer.Track
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receive int
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send int
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}
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func (c *Conn) Init() {
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c.Conn.OnICECandidate(func(candidate *webrtc.ICECandidate) {
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if candidate != nil {
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c.Fire(&streamer.Message{
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Type: MsgTypeCandidate, Value: candidate.ToJSON().Candidate,
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})
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}
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})
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c.Conn.OnTrack(func(remote *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) {
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for _, track := range c.tracks {
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if track.Direction != streamer.DirectionRecvonly {
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continue
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}
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if track.Codec.PayloadType != uint8(remote.PayloadType()) {
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continue
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}
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for {
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packet, _, err := remote.ReadRTP()
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if err != nil {
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return
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}
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if len(packet.Payload) == 0 {
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continue
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}
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c.receive += len(packet.Payload)
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_ = track.WriteRTP(packet)
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}
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}
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panic("something wrong")
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})
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c.Conn.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
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c.Fire(state)
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// TODO: remove
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switch state {
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case webrtc.PeerConnectionStateConnected:
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c.Fire(streamer.StatePlaying)
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case webrtc.PeerConnectionStateDisconnected:
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c.Fire(streamer.StateNull)
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case webrtc.PeerConnectionStateFailed:
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_ = c.Conn.Close()
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}
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})
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}
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func (c *Conn) ExchangeSDP(offer string, complete bool) (answer string, err error) {
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sdOffer := webrtc.SessionDescription{
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Type: webrtc.SDPTypeOffer, SDP: offer,
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}
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if err = c.Conn.SetRemoteDescription(sdOffer); err != nil {
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return
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}
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//for _, tr := range c.Conn.GetTransceivers() {
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// switch tr.Direction() {
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// case webrtc.RTPTransceiverDirectionSendonly:
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// // disable transceivers if we don't have track
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// // make direction=inactive
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// // don't really necessary, but anyway
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// if tr.Sender() == nil {
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// if err = tr.Stop(); err != nil {
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// return
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// }
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// }
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// case webrtc.RTPTransceiverDirectionRecvonly:
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// // TODO: change codecs list
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// caps := webrtc.RTPCodecCapability{
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// MimeType: webrtc.MimeTypePCMU,
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// ClockRate: 8000,
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// }
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// codecs := []webrtc.RTPCodecParameters{
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// {RTPCodecCapability: caps},
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// }
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// if err = tr.SetCodecPreferences(codecs); err != nil {
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// return
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// }
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// }
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//}
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var sdAnswer webrtc.SessionDescription
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sdAnswer, err = c.Conn.CreateAnswer(nil)
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if err != nil {
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return
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}
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//var sd *sdp.SessionDescription
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//sd, err = sdAnswer.Unmarshal()
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//for _, media := range sd.MediaDescriptions {
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// if media.MediaName.Media != "audio" {
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// continue
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// }
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// for i, attr := range media.Attributes {
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// if attr.Key == "sendonly" {
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// attr.Key = "inactive"
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// media.Attributes[i] = attr
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// break
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// }
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// }
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//}
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//var b []byte
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//b, err = sd.Marshal()
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//sdAnswer.SDP = string(b)
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if err = c.Conn.SetLocalDescription(sdAnswer); err != nil {
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return
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}
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if complete {
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<-webrtc.GatheringCompletePromise(c.Conn)
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return c.Conn.LocalDescription().SDP, nil
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}
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return sdAnswer.SDP, nil
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}
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func (c *Conn) SetOffer(offer string) (err error) {
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sdOffer := webrtc.SessionDescription{
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Type: webrtc.SDPTypeOffer, SDP: offer,
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}
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if err = c.Conn.SetRemoteDescription(sdOffer); err != nil {
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return
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}
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rawSDP := []byte(c.Conn.RemoteDescription().SDP)
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c.medias, err = streamer.UnmarshalSDP(rawSDP)
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return
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}
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func (c *Conn) GetAnswer() (answer string, err error) {
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for _, tr := range c.Conn.GetTransceivers() {
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if tr.Direction() != webrtc.RTPTransceiverDirectionSendonly {
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continue
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}
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// disable transceivers if we don't have track
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// make direction=inactive
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// don't really necessary, but anyway
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if tr.Sender() == nil {
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if err = tr.Stop(); err != nil {
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return
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}
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}
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}
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var sdAnswer webrtc.SessionDescription
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sdAnswer, err = c.Conn.CreateAnswer(nil)
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if err != nil {
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return
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}
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if err = c.Conn.SetLocalDescription(sdAnswer); err != nil {
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return
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}
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return sdAnswer.SDP, nil
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}
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func (c *Conn) remote() string {
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for _, trans := range c.Conn.GetTransceivers() {
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pair, _ := trans.Receiver().Transport().ICETransport().GetSelectedCandidatePair()
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return pair.Remote.String()
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}
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return ""
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}
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@@ -0,0 +1,85 @@
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package webrtc
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import (
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"github.com/AlexxIT/go2rtc/pkg/streamer"
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"github.com/pion/ice/v2"
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"github.com/pion/stun"
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"github.com/pion/webrtc/v3"
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"net"
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"strconv"
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)
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func NewCandidate(address string) (string, error) {
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host, port, err := net.SplitHostPort(address)
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if err != nil {
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return "", err
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}
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i, err := strconv.Atoi(port)
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if err != nil {
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return "", err
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}
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cand, err := ice.NewCandidateHost(&ice.CandidateHostConfig{
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Network: "tcp",
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Address: host,
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Port: i,
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Component: ice.ComponentRTP,
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TCPType: ice.TCPTypePassive,
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})
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if err != nil {
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return "", err
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}
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return "candidate:" + cand.Marshal(), nil
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}
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// GetPublicIP example from https://github.com/pion/stun
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func GetPublicIP() (net.IP, error) {
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c, err := stun.Dial("udp", "stun.l.google.com:19302")
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if err != nil {
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return nil, err
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}
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var res stun.Event
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message := stun.MustBuild(stun.TransactionID, stun.BindingRequest)
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if err = c.Do(message, func(e stun.Event) { res = e }); err != nil {
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return nil, err
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}
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if res.Error != nil {
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return nil, res.Error
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}
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var xorAddr stun.XORMappedAddress
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if err = xorAddr.GetFrom(res.Message); err != nil {
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return nil, err
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}
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return xorAddr.IP, nil
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}
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func MimeType(codec *streamer.Codec) string {
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switch codec.Name {
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case streamer.CodecH264:
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return webrtc.MimeTypeH264
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case streamer.CodecH265:
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return webrtc.MimeTypeH265
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case streamer.CodecVP8:
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return webrtc.MimeTypeVP8
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case streamer.CodecVP9:
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return webrtc.MimeTypeVP9
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case streamer.CodecAV1:
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return webrtc.MimeTypeAV1
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case streamer.CodecPCMU:
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return webrtc.MimeTypePCMU
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case streamer.CodecPCMA:
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return webrtc.MimeTypePCMA
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case streamer.CodecOpus:
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return webrtc.MimeTypeOpus
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case streamer.CodecG722:
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return webrtc.MimeTypeG722
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}
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panic("not implemented")
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}
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@@ -0,0 +1,125 @@
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package webrtc
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import (
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"encoding/json"
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"github.com/AlexxIT/go2rtc/pkg/h264"
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"github.com/AlexxIT/go2rtc/pkg/streamer"
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"github.com/pion/rtp"
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"github.com/pion/webrtc/v3"
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)
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// Consumer
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func (c *Conn) GetMedias() []*streamer.Media {
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return c.medias
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}
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func (c *Conn) AddTrack(media *streamer.Media, track *streamer.Track) *streamer.Track {
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switch track.Direction {
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// send our track to WebRTC consumer
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case streamer.DirectionSendonly:
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codec := track.Codec
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// webrtc.codecParametersFuzzySearch
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caps := webrtc.RTPCodecCapability{
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MimeType: MimeType(codec),
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Channels: codec.Channels,
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ClockRate: codec.ClockRate,
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}
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if codec.Name == streamer.CodecH264 {
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// don't know if this really neccessary
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// I have tested multiple browsers and H264 profile has no effect on anything
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caps.SDPFmtpLine = "packetization-mode=1;profile-level-id=42e01f"
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}
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// important to use same streamID so JS will automatically
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// join two tracks as one source/stream
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trackLocal, err := webrtc.NewTrackLocalStaticRTP(
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caps, caps.MimeType[:5], "go2rtc",
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)
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if err != nil {
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return nil
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}
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if _, err = c.Conn.AddTrack(trackLocal); err != nil {
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return nil
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}
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push := func(packet *rtp.Packet) error {
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c.send += packet.MarshalSize()
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return trackLocal.WriteRTP(packet)
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}
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if codec.Name == streamer.CodecH264 {
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wrapper := h264.RTPPay(1200)
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push = wrapper(push)
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if codec.PayloadType != 255 {
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wrapper = h264.RTPDepay(track)
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push = wrapper(push)
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}
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}
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track = track.Bind(push)
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c.tracks = append(c.tracks, track)
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return track
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// receive track from WebRTC consumer (microphone, backchannel, two way audio)
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case streamer.DirectionRecvonly:
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for _, tr := range c.Conn.GetTransceivers() {
|
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if tr.Mid() != media.MID {
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continue
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}
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codec := track.Codec
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caps := webrtc.RTPCodecCapability{
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MimeType: MimeType(codec),
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ClockRate: codec.ClockRate,
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Channels: codec.Channels,
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}
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codecs := []webrtc.RTPCodecParameters{
|
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{RTPCodecCapability: caps},
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}
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if err := tr.SetCodecPreferences(codecs); err != nil {
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return nil
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}
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c.tracks = append(c.tracks, track)
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return track
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}
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}
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panic("wrong direction")
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}
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//
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func (c *Conn) Push(msg interface{}) {
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if msg := msg.(*streamer.Message); msg != nil {
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if msg.Type == MsgTypeCandidate {
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_ = c.Conn.AddICECandidate(webrtc.ICECandidateInit{
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Candidate: msg.Value.(string),
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})
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}
|
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}
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}
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func (c *Conn) MarshalJSON() ([]byte, error) {
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v := map[string]interface{}{
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streamer.JSONType: "WebRTC server consumer",
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streamer.JSONRemoteAddr: c.remote(),
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}
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if c.receive > 0 {
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v[streamer.JSONReceive] = c.receive
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}
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if c.send > 0 {
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v[streamer.JSONSend] = c.send
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}
|
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if c.UserAgent != "" {
|
||||
v[streamer.JSONUserAgent] = c.UserAgent
|
||||
}
|
||||
|
||||
return json.Marshal(v)
|
||||
}
|
||||
@@ -0,0 +1,56 @@
|
||||
package webrtc
|
||||
|
||||
import (
|
||||
"github.com/pion/ice/v2"
|
||||
"github.com/pion/sdp/v3"
|
||||
"github.com/pion/webrtc/v3"
|
||||
"github.com/stretchr/testify/assert"
|
||||
"testing"
|
||||
)
|
||||
|
||||
func TestName(t *testing.T) {
|
||||
i, _ := ice.NewCandidateHost(&ice.CandidateHostConfig{
|
||||
Network: "tcp",
|
||||
Address: "192.168.1.123",
|
||||
Port: 8555,
|
||||
Component: ice.ComponentRTP,
|
||||
TCPType: ice.TCPTypePassive,
|
||||
})
|
||||
a := i.Marshal()
|
||||
println(a)
|
||||
}
|
||||
|
||||
func TestPublicIP(t *testing.T) {
|
||||
ip, err := GetPublicIP()
|
||||
assert.Nil(t, err)
|
||||
assert.NotNil(t, ip)
|
||||
t.Logf("your public IP: %s", ip.String())
|
||||
}
|
||||
|
||||
func TestMedia(t *testing.T) {
|
||||
codec := webrtc.RTPCodecParameters{
|
||||
RTPCodecCapability: webrtc.RTPCodecCapability{
|
||||
MimeType: webrtc.MimeTypeH264,
|
||||
ClockRate: 90000,
|
||||
SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f",
|
||||
},
|
||||
PayloadType: 96,
|
||||
}
|
||||
|
||||
md := &sdp.MediaDescription{
|
||||
MediaName: sdp.MediaName{
|
||||
Media: "video", Protos: []string{"RTP", "AVP"},
|
||||
},
|
||||
}
|
||||
md.WithCodec(
|
||||
uint8(codec.PayloadType), codec.MimeType[6:], codec.ClockRate,
|
||||
codec.Channels, codec.SDPFmtpLine,
|
||||
)
|
||||
|
||||
sd := &sdp.SessionDescription{
|
||||
MediaDescriptions: []*sdp.MediaDescription{md},
|
||||
}
|
||||
data, err := sd.Marshal()
|
||||
assert.Nil(t, err)
|
||||
assert.NotNil(t, data)
|
||||
}
|
||||
Reference in New Issue
Block a user