neko/server/internal/webrtc/manager.go

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package webrtc
import (
"fmt"
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"net"
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"strings"
"sync"
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"sync/atomic"
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"time"
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"github.com/pion/ice/v2"
"github.com/pion/interceptor"
"github.com/pion/interceptor/pkg/cc"
"github.com/pion/interceptor/pkg/gcc"
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"github.com/pion/rtcp"
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"github.com/pion/webrtc/v3"
"github.com/rs/zerolog"
"github.com/rs/zerolog/log"
"github.com/demodesk/neko/internal/config"
"github.com/demodesk/neko/internal/webrtc/cursor"
"github.com/demodesk/neko/internal/webrtc/pionlog"
"github.com/demodesk/neko/pkg/types"
"github.com/demodesk/neko/pkg/types/codec"
"github.com/demodesk/neko/pkg/types/event"
"github.com/demodesk/neko/pkg/types/message"
"github.com/demodesk/neko/pkg/utils"
)
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const (
// size of receiving channel used to buffer incoming TCP packets
tcpReadChanBufferSize = 50
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// size of buffer used to buffer outgoing TCP packets. Default is 4MB
tcpWriteBufferSizeInBytes = 4 * 1024 * 1024
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// the duration without network activity before a Agent is considered disconnected. Default is 5 Seconds
disconnectedTimeout = 4 * time.Second
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// the duration without network activity before a Agent is considered failed after disconnected. Default is 25 Seconds
failedTimeout = 6 * time.Second
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// how often the ICE Agent sends extra traffic if there is no activity, if media is flowing no traffic will be sent. Default is 2 seconds
keepAliveInterval = 2 * time.Second
// send a PLI on an interval so that the publisher is pushing a keyframe every rtcpPLIInterval
rtcpPLIInterval = 3 * time.Second
)
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func New(desktop types.DesktopManager, capture types.CaptureManager, config *config.WebRTC) *WebRTCManagerCtx {
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logger := log.With().Str("module", "webrtc").Logger()
configuration := webrtc.Configuration{
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SDPSemantics: webrtc.SDPSemanticsUnifiedPlan,
}
if !config.ICELite {
ICEServers := []webrtc.ICEServer{}
for _, server := range config.ICEServersBackend {
var credential any
if server.Credential != "" {
credential = server.Credential
} else {
credential = false
}
ICEServers = append(ICEServers, webrtc.ICEServer{
URLs: server.URLs,
Username: server.Username,
Credential: credential,
})
}
configuration.ICEServers = ICEServers
}
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return &WebRTCManagerCtx{
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logger: logger,
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config: config,
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metrics: newMetricsManager(),
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webrtcConfiguration: configuration,
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desktop: desktop,
capture: capture,
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curImage: cursor.NewImage(logger, desktop),
curPosition: cursor.NewPosition(logger),
}
}
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type WebRTCManagerCtx struct {
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logger zerolog.Logger
config *config.WebRTC
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metrics *metricsManager
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peerId int32
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desktop types.DesktopManager
capture types.CaptureManager
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curImage cursor.Image
curPosition cursor.Position
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webrtcConfiguration webrtc.Configuration
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tcpMux ice.TCPMux
udpMux ice.UDPMux
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camStop, micStop *func()
}
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func (manager *WebRTCManagerCtx) Start() {
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manager.curImage.Start()
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logger := pionlog.New(manager.logger)
// add TCP Mux listener
if manager.config.TCPMux > 0 {
tcpListener, err := net.ListenTCP("tcp", &net.TCPAddr{
IP: net.IP{0, 0, 0, 0},
Port: manager.config.TCPMux,
})
if err != nil {
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manager.logger.Fatal().Err(err).Msg("unable to setup ice TCP mux")
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}
manager.tcpMux = ice.NewTCPMuxDefault(ice.TCPMuxParams{
Listener: tcpListener,
Logger: logger.NewLogger("ice-tcp"),
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ReadBufferSize: tcpReadChanBufferSize,
WriteBufferSize: tcpWriteBufferSizeInBytes,
})
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}
// add UDP Mux listener
if manager.config.UDPMux > 0 {
var err error
manager.udpMux, err = ice.NewMultiUDPMuxFromPort(manager.config.UDPMux,
ice.UDPMuxFromPortWithLogger(logger.NewLogger("ice-udp")),
)
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if err != nil {
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manager.logger.Fatal().Err(err).Msg("unable to setup ice UDP mux")
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}
}
manager.logger.Info().
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Bool("icelite", manager.config.ICELite).
Bool("icetrickle", manager.config.ICETrickle).
Interface("iceservers-frontend", manager.config.ICEServersFrontend).
Interface("iceservers-backend", manager.config.ICEServersBackend).
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Str("nat1to1", strings.Join(manager.config.NAT1To1IPs, ",")).
Str("epr", fmt.Sprintf("%d-%d", manager.config.EphemeralMin, manager.config.EphemeralMax)).
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Int("tcpmux", manager.config.TCPMux).
Int("udpmux", manager.config.UDPMux).
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Msg("webrtc starting")
}
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func (manager *WebRTCManagerCtx) Shutdown() error {
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manager.logger.Info().Msg("shutdown")
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manager.curImage.Shutdown()
manager.curPosition.Shutdown()
return nil
}
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func (manager *WebRTCManagerCtx) ICEServers() []types.ICEServer {
return manager.config.ICEServersFrontend
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}
func (manager *WebRTCManagerCtx) newPeerConnection(logger zerolog.Logger, codecs []codec.RTPCodec) (*webrtc.PeerConnection, cc.BandwidthEstimator, error) {
// create media engine
engine := &webrtc.MediaEngine{}
for _, codec := range codecs {
if err := codec.Register(engine); err != nil {
return nil, nil, err
}
}
// create setting engine
settings := webrtc.SettingEngine{
LoggerFactory: pionlog.New(logger),
}
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settings.DisableMediaEngineCopy(true)
settings.SetICETimeouts(disconnectedTimeout, failedTimeout, keepAliveInterval)
settings.SetNAT1To1IPs(manager.config.NAT1To1IPs, webrtc.ICECandidateTypeHost)
settings.SetLite(manager.config.ICELite)
// make sure server answer sdp setup as passive, to not force DTLS renegotiation
// otherwise iOS renegotiation fails with: Failed to set SSL role for the transport.
settings.SetAnsweringDTLSRole(webrtc.DTLSRoleServer)
var networkType []webrtc.NetworkType
// udp candidates
if manager.udpMux != nil {
settings.SetICEUDPMux(manager.udpMux)
networkType = append(networkType,
webrtc.NetworkTypeUDP4,
webrtc.NetworkTypeUDP6,
)
} else if manager.config.EphemeralMax != 0 {
_ = settings.SetEphemeralUDPPortRange(manager.config.EphemeralMin, manager.config.EphemeralMax)
networkType = append(networkType,
webrtc.NetworkTypeUDP4,
webrtc.NetworkTypeUDP6,
)
}
// tcp candidates
if manager.tcpMux != nil {
settings.SetICETCPMux(manager.tcpMux)
networkType = append(networkType,
webrtc.NetworkTypeTCP4,
webrtc.NetworkTypeTCP6,
)
}
// enable support for TCP and UDP ICE candidates
settings.SetNetworkTypes(networkType)
// create interceptor registry
registry := &interceptor.Registry{}
// create bandwidth estimator
estimatorChan := make(chan cc.BandwidthEstimator, 1)
if manager.config.Estimator.Enabled {
congestionController, err := cc.NewInterceptor(func() (cc.BandwidthEstimator, error) {
return gcc.NewSendSideBWE(
gcc.SendSideBWEInitialBitrate(manager.config.Estimator.InitialBitrate),
gcc.SendSideBWEPacer(gcc.NewNoOpPacer()),
)
})
if err != nil {
return nil, nil, err
}
congestionController.OnNewPeerConnection(func(id string, estimator cc.BandwidthEstimator) {
estimatorChan <- estimator
})
registry.Add(congestionController)
if err = webrtc.ConfigureTWCCHeaderExtensionSender(engine, registry); err != nil {
return nil, nil, err
}
} else {
// no estimator, send nil
estimatorChan <- nil
}
if err := webrtc.RegisterDefaultInterceptors(engine, registry); err != nil {
return nil, nil, err
}
// create new API
api := webrtc.NewAPI(
webrtc.WithMediaEngine(engine),
webrtc.WithSettingEngine(settings),
webrtc.WithInterceptorRegistry(registry),
)
// create new peer connection
configuration := manager.webrtcConfiguration
connection, err := api.NewPeerConnection(configuration)
return connection, <-estimatorChan, err
}
func (manager *WebRTCManagerCtx) CreatePeer(session types.Session) (*webrtc.SessionDescription, types.WebRTCPeer, error) {
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id := atomic.AddInt32(&manager.peerId, 1)
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// get metrics for session
metrics := manager.metrics.getBySession(session)
metrics.NewConnection()
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// add session id to logger context
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logger := manager.logger.With().Str("session_id", session.ID()).Int32("peer_id", id).Logger()
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logger.Info().Msg("creating webrtc peer")
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// all audios must have the same codec
audio := manager.capture.Audio()
audioCodec := audio.Codec()
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// all videos must have the same codec
video := manager.capture.Video()
videoCodec := video.Codec()
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connection, estimator, err := manager.newPeerConnection(
logger, []codec.RTPCodec{audioCodec, videoCodec})
if err != nil {
return nil, nil, err
}
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// asynchronously send local ICE Candidates
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if manager.config.ICETrickle {
connection.OnICECandidate(func(candidate *webrtc.ICECandidate) {
if candidate == nil {
logger.Debug().Msg("all local ice candidates sent")
return
}
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session.Send(
event.SIGNAL_CANDIDATE,
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message.SignalCandidate{
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ICECandidateInit: candidate.ToJSON(),
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})
})
}
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// audio track
audioTrack, err := NewTrack(logger, audioCodec, connection)
if err != nil {
return nil, nil, err
}
// we disable audio by default manually
audioTrack.SetPaused(true)
// set stream for audio track
_, err = audioTrack.SetStream(audio)
if err != nil {
return nil, nil, err
}
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// video track
videoRtcp := make(chan []rtcp.Packet, 1)
videoTrack, err := NewTrack(logger, videoCodec, connection, WithRtcpChan(videoRtcp))
if err != nil {
return nil, nil, err
}
//
// stream for video track will be set later
//
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// data channel
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dataChannel, err := connection.CreateDataChannel("data", nil)
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if err != nil {
return nil, nil, err
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}
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peer := &WebRTCPeerCtx{
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logger: logger,
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session: session,
metrics: metrics,
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connection: connection,
// bandwidth estimator
estimator: estimator,
estimateTrend: utils.NewTrendDetector(
utils.TrendDetectorParams{
// Probing
//RequiredSamples: 3,
//DownwardTrendThreshold: 0.0,
//CollapseValues: false,
// Non-Probing
RequiredSamples: 8,
DownwardTrendThreshold: -0.5,
CollapseValues: true,
}),
// stream selectors
video: video,
audio: audio,
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// tracks & channels
audioTrack: audioTrack,
videoTrack: videoTrack,
dataChannel: dataChannel,
rtcpChannel: videoRtcp,
// config
iceTrickle: manager.config.ICETrickle,
estimatorConfig: manager.config.Estimator,
audioDisabled: true, // we disable audio by default manually
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}
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connection.OnTrack(func(track *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) {
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logger := logger.With().
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Str("kind", track.Kind().String()).
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Str("mime", track.Codec().RTPCodecCapability.MimeType).
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Logger()
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logger.Info().Msgf("received new remote track")
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if !session.Profile().CanShareMedia {
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err := receiver.Stop()
logger.Warn().Err(err).Msg("media sharing is disabled for this session")
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return
}
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// parse codec from remote track
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codec, ok := codec.ParseRTC(track.Codec())
if !ok {
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err := receiver.Stop()
logger.Warn().Err(err).Msg("remote track with unknown codec")
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return
}
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var srcManager types.StreamSrcManager
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stopped := false
stopFn := func() {
if stopped {
return
}
stopped = true
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err := receiver.Stop()
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srcManager.Stop()
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logger.Err(err).Msg("remote track stopped")
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}
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if track.Kind() == webrtc.RTPCodecTypeAudio {
// audio -> microphone
srcManager = manager.capture.Microphone()
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defer stopFn()
if manager.micStop != nil {
(*manager.micStop)()
}
manager.micStop = &stopFn
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} else if track.Kind() == webrtc.RTPCodecTypeVideo {
// video -> webcam
srcManager = manager.capture.Webcam()
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defer stopFn()
if manager.camStop != nil {
(*manager.camStop)()
}
manager.camStop = &stopFn
} else {
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err := receiver.Stop()
logger.Warn().Err(err).Msg("remote track with unsupported codec type")
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return
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}
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err := srcManager.Start(codec)
if err != nil {
logger.Err(err).Msg("failed to start pipeline")
return
}
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ticker := time.NewTicker(rtcpPLIInterval)
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defer ticker.Stop()
go func() {
for range ticker.C {
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err := connection.WriteRTCP([]rtcp.Packet{
&rtcp.PictureLossIndication{
MediaSSRC: uint32(track.SSRC()),
},
})
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if err != nil {
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logger.Err(err).Msg("remote track rtcp send err")
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}
}
}()
buf := make([]byte, 1400)
for {
i, _, err := track.Read(buf)
if err != nil {
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logger.Warn().Err(err).Msg("failed read from remote track")
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break
}
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srcManager.Push(buf[:i])
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}
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logger.Info().Msg("remote track data finished")
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})
connection.OnDataChannel(func(dc *webrtc.DataChannel) {
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logger.Info().Interface("data_channel", dc).Msg("got remote data channel")
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//
// old implementation created a new data channel on client side
// new implementation creates a new data channel on server side
//
// handle legacy data channel
dc.OnMessage(func(message webrtc.DataChannelMessage) {
if err := manager.handleLegacy(logger, message.Data, session); err != nil {
logger.Err(err).Msg("data handle failed")
}
})
// handle legacy data channel
peer.dataChannel = dc
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})
var once sync.Once
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connection.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
switch state {
case webrtc.PeerConnectionStateConnected:
session.SetWebRTCConnected(peer, true)
case webrtc.PeerConnectionStateDisconnected,
webrtc.PeerConnectionStateFailed:
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peer.Destroy()
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case webrtc.PeerConnectionStateClosed:
// ensure we only run this once
once.Do(func() {
session.SetWebRTCConnected(peer, false)
//
// TODO: Shutdown peer?
//
audioTrack.Shutdown()
videoTrack.Shutdown()
close(videoRtcp)
})
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}
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metrics.SetState(state)
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})
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dataChannel.OnOpen(func() {
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manager.curImage.AddListener(peer)
manager.curPosition.AddListener(peer)
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// send initial cursor image
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cur, img, err := manager.curImage.GetCurrent()
if err == nil {
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err := peer.SendCursorImage(cur, img)
if err != nil {
logger.Err(err).Msg("failed to set cursor image")
}
} else {
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logger.Err(err).Msg("failed to get cursor image")
}
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// send initial cursor position
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x, y := manager.desktop.GetCursorPosition()
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err = peer.SendCursorPosition(x, y)
if err != nil {
logger.Err(err).Msg("failed to set cursor position")
}
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})
dataChannel.OnClose(func() {
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manager.curImage.RemoveListener(peer)
manager.curPosition.RemoveListener(peer)
})
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dataChannel.OnMessage(func(message webrtc.DataChannelMessage) {
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if err := manager.handle(logger, message.Data, dataChannel, session); err != nil {
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logger.Err(err).Msg("data handle failed")
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}
})
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session.SetWebRTCPeer(peer)
offer, err := peer.CreateOffer(false)
if err != nil {
return nil, nil, err
}
// on negotiation needed handler must be registered after creating initial
// offer, otherwise it can fire and intercept sucessful negotiation
connection.OnNegotiationNeeded(func() {
logger.Warn().Msg("negotiation is needed")
if connection.SignalingState() != webrtc.SignalingStateStable {
logger.Warn().Msg("connection isn't stable yet; postponing...")
return
}
offer, err := peer.CreateOffer(false)
if err != nil {
logger.Err(err).Msg("sdp offer failed")
return
}
session.Send(
event.SIGNAL_OFFER,
message.SignalDescription{
SDP: offer.SDP,
})
})
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// start metrics collectors
go metrics.rtcpReceiver(videoRtcp)
go metrics.connectionStats(connection)
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// start estimator reader
go peer.estimatorReader()
return offer, peer, nil
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}
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func (manager *WebRTCManagerCtx) SetCursorPosition(x, y int) {
manager.curPosition.Set(x, y)
}