mirror of
https://github.com/m1k1o/neko.git
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347 lines
10 KiB
Go
347 lines
10 KiB
Go
package webrtc
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import (
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"encoding/json"
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"fmt"
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"io"
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"net"
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"strings"
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"time"
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"github.com/pion/interceptor"
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"github.com/pion/webrtc/v3"
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"github.com/pion/webrtc/v3/pkg/media"
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"github.com/rs/zerolog"
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"github.com/rs/zerolog/log"
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"m1k1o/neko/internal/types"
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"m1k1o/neko/internal/types/config"
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)
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func New(sessions types.SessionManager, remote types.RemoteManager, config *config.WebRTC) *WebRTCManager {
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return &WebRTCManager{
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logger: log.With().Str("module", "webrtc").Logger(),
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remote: remote,
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sessions: sessions,
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config: config,
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}
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}
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type WebRTCManager struct {
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logger zerolog.Logger
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videoTrack *webrtc.TrackLocalStaticSample
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audioTrack *webrtc.TrackLocalStaticSample
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videoCodec webrtc.RTPCodecParameters
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audioCodec webrtc.RTPCodecParameters
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sessions types.SessionManager
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remote types.RemoteManager
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config *config.WebRTC
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api *webrtc.API
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}
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func (manager *WebRTCManager) Start() {
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var err error
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manager.audioTrack, manager.audioCodec, err = manager.createTrack(manager.remote.AudioCodec())
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if err != nil {
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manager.logger.Panic().Err(err).Msg("unable to create audio track")
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}
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manager.remote.OnAudioFrame(func(sample types.Sample) {
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if err := manager.audioTrack.WriteSample(media.Sample(sample)); err != nil && err != io.ErrClosedPipe {
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manager.logger.Warn().Err(err).Msg("audio pipeline failed to write")
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}
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})
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manager.videoTrack, manager.videoCodec, err = manager.createTrack(manager.remote.VideoCodec())
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if err != nil {
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manager.logger.Panic().Err(err).Msg("unable to create video track")
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}
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manager.remote.OnVideoFrame(func(sample types.Sample) {
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if err := manager.videoTrack.WriteSample(media.Sample(sample)); err != nil && err != io.ErrClosedPipe {
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manager.logger.Warn().Err(err).Msg("video pipeline failed to write")
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}
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})
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if err := manager.initAPI(); err != nil {
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manager.logger.Panic().Err(err).Msg("failed to initialize webrtc API")
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}
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manager.logger.Info().
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Str("ice_lite", fmt.Sprintf("%t", manager.config.ICELite)).
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Str("ice_servers", fmt.Sprintf("%+v", manager.config.ICEServers)).
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Str("ephemeral_port_range", fmt.Sprintf("%d-%d", manager.config.EphemeralMin, manager.config.EphemeralMax)).
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Str("nat_ips", strings.Join(manager.config.NAT1To1IPs, ",")).
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Msgf("webrtc starting")
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}
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func (manager *WebRTCManager) Shutdown() error {
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manager.logger.Info().Msgf("webrtc shutting down")
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return nil
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}
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func (manager *WebRTCManager) initAPI() error {
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logger := loggerFactory{
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logger: manager.logger,
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}
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settings := webrtc.SettingEngine{
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LoggerFactory: logger,
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}
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_ = settings.SetEphemeralUDPPortRange(manager.config.EphemeralMin, manager.config.EphemeralMax)
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settings.SetNAT1To1IPs(manager.config.NAT1To1IPs, webrtc.ICECandidateTypeHost)
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settings.SetICETimeouts(6*time.Second, 6*time.Second, 3*time.Second)
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settings.SetSRTPReplayProtectionWindow(512)
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settings.SetLite(manager.config.ICELite)
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var networkType []webrtc.NetworkType
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// Add TCP Mux
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if manager.config.TCPMUX > 0 {
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tcpListener, err := net.ListenTCP("tcp", &net.TCPAddr{
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IP: net.IP{0, 0, 0, 0},
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Port: manager.config.TCPMUX,
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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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tcpMux := webrtc.NewICETCPMux(logger.NewLogger("ice-tcp"), tcpListener, 32)
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settings.SetICETCPMux(tcpMux)
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networkType = append(networkType, webrtc.NetworkTypeTCP4)
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manager.logger.Info().Str("listener", tcpListener.Addr().String()).Msg("using TCP MUX")
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}
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// Add UDP Mux
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if manager.config.UDPMUX > 0 {
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udpListener, err := net.ListenUDP("udp", &net.UDPAddr{
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IP: net.IP{0, 0, 0, 0},
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Port: manager.config.UDPMUX,
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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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udpMux := webrtc.NewICEUDPMux(logger.NewLogger("ice-udp"), udpListener)
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settings.SetICEUDPMux(udpMux)
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networkType = append(networkType, webrtc.NetworkTypeUDP4)
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manager.logger.Info().Str("listener", udpListener.LocalAddr().String()).Msg("using UDP MUX")
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}
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// Enable support for TCP and UDP ICE candidates
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if len(networkType) > 0 {
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settings.SetNetworkTypes(networkType)
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}
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// Create MediaEngine with selected codecs
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engine := webrtc.MediaEngine{}
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_ = engine.RegisterCodec(manager.audioCodec, webrtc.RTPCodecTypeAudio)
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_ = engine.RegisterCodec(manager.videoCodec, webrtc.RTPCodecTypeVideo)
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// Register Interceptors
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i := &interceptor.Registry{}
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if err := webrtc.RegisterDefaultInterceptors(&engine, i); err != nil {
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return err
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}
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// Create API with MediaEngine and SettingEngine
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manager.api = webrtc.NewAPI(
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webrtc.WithMediaEngine(&engine),
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webrtc.WithSettingEngine(settings),
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webrtc.WithInterceptorRegistry(i),
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)
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return nil
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}
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func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (types.Peer, error) {
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configuration := webrtc.Configuration{
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SDPSemantics: webrtc.SDPSemanticsUnifiedPlanWithFallback,
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}
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if !manager.config.ICELite {
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configuration.ICEServers = manager.config.ICEServers
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}
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// Create new peer connection
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connection, err := manager.api.NewPeerConnection(configuration)
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if err != nil {
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return nil, err
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}
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negotiated := true
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_, err = connection.CreateDataChannel("data", &webrtc.DataChannelInit{
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Negotiated: &negotiated,
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})
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if err != nil {
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return nil, err
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}
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connection.OnDataChannel(func(d *webrtc.DataChannel) {
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d.OnMessage(func(msg webrtc.DataChannelMessage) {
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if err = manager.handle(id, msg); err != nil {
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manager.logger.Warn().Err(err).Msg("data handle failed")
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}
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})
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})
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// Set the handler for ICE connection state
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// This will notify you when the peer has connected/disconnected
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connection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
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manager.logger.Info().
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Str("connection_state", connectionState.String()).
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Msg("connection state has changed")
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})
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rtpVideo, err := connection.AddTrack(manager.videoTrack)
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if err != nil {
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return nil, err
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}
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rtpAudio, err := connection.AddTrack(manager.audioTrack)
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if err != nil {
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return nil, err
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}
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connection.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
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switch state {
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case webrtc.PeerConnectionStateDisconnected:
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manager.logger.Info().Str("id", id).Msg("peer disconnected")
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manager.sessions.Destroy(id)
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case webrtc.PeerConnectionStateFailed:
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manager.logger.Warn().Str("id", id).Msg("peer failed")
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manager.sessions.Destroy(id)
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case webrtc.PeerConnectionStateClosed:
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manager.logger.Info().Str("id", id).Msg("peer closed")
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manager.sessions.Destroy(id)
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case webrtc.PeerConnectionStateConnected:
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manager.logger.Info().Str("id", id).Msg("peer connected")
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if err = session.SetConnected(true); err != nil {
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manager.logger.Warn().Err(err).Msg("unable to set connected on peer")
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manager.sessions.Destroy(id)
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}
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}
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})
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peer := &Peer{
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id: id,
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manager: manager,
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connection: connection,
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}
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connection.OnNegotiationNeeded(func() {
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manager.logger.Warn().Msg("negotiation is needed")
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sdp, err := peer.CreateOffer()
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if err != nil {
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manager.logger.Err(err).Msg("creating offer failed")
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return
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}
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err = session.SignalLocalOffer(sdp)
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if err != nil {
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manager.logger.Warn().Err(err).Msg("sending SignalLocalOffer failed")
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return
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}
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})
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connection.OnICECandidate(func(i *webrtc.ICECandidate) {
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if i == nil {
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manager.logger.Info().Msg("sent all ICECandidates")
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return
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}
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candidateString, err := json.Marshal(i.ToJSON())
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if err != nil {
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manager.logger.Warn().Err(err).Msg("converting ICECandidate to json failed")
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return
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}
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if err := session.SignalCandidate(string(candidateString)); err != nil {
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manager.logger.Warn().Err(err).Msg("sending SignalCandidate failed")
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return
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}
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})
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if err := session.SetPeer(peer); err != nil {
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return nil, err
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}
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go func() {
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rtcpBuf := make([]byte, 1500)
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for {
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if _, _, rtcpErr := rtpVideo.Read(rtcpBuf); rtcpErr != nil {
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return
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}
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}
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}()
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go func() {
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rtcpBuf := make([]byte, 1500)
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for {
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if _, _, rtcpErr := rtpAudio.Read(rtcpBuf); rtcpErr != nil {
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return
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}
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}
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}()
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return peer, nil
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}
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func (manager *WebRTCManager) ICELite() bool {
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return manager.config.ICELite
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}
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func (manager *WebRTCManager) ICEServers() []webrtc.ICEServer {
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return manager.config.ICEServers
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}
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func (manager *WebRTCManager) createTrack(codecName string) (*webrtc.TrackLocalStaticSample, webrtc.RTPCodecParameters, error) {
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var codec webrtc.RTPCodecParameters
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id := ""
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fb := []webrtc.RTCPFeedback{}
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switch codecName {
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case "VP8":
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codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeVP8, ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 96}
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id = "video"
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case "VP9":
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codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeVP9, ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 98}
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id = "video"
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case "H264":
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codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeH264, ClockRate: 90000, Channels: 0, SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f", RTCPFeedback: fb}, PayloadType: 102}
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id = "video"
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case "H265":
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// TODO: Fix this (correct payload type and SDPFmtpLine if needed).
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codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "video/H265", ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 0}
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id = "video"
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case "Opus":
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codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeOpus, ClockRate: 48000, Channels: 2, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 111}
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id = "audio"
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case "G722":
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codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeG722, ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 9}
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id = "audio"
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case "PCMU":
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codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypePCMU, ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 0}
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id = "audio"
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case "PCMA":
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codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypePCMA, ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 8}
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id = "audio"
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default:
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return nil, codec, fmt.Errorf("unknown codec %s", codecName)
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}
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track, err := webrtc.NewTrackLocalStaticSample(codec.RTPCodecCapability, id, "stream")
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if err != nil {
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return nil, codec, err
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}
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return track, codec, nil
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}
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