2020-09-29 15:04:23 +00:00
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package webrtc
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2020-09-24 09:24:13 +00:00
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import (
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"fmt"
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2020-09-24 17:01:26 +00:00
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"log"
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2020-09-24 09:24:13 +00:00
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"math/rand"
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"github.com/pion/webrtc/v3"
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2020-09-29 16:17:55 +00:00
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"gitlab.crans.org/nounous/ghostream/internal/monitoring"
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2020-10-04 16:22:10 +00:00
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"gitlab.crans.org/nounous/ghostream/stream/srt"
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2020-09-24 09:24:13 +00:00
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)
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2020-09-29 15:04:23 +00:00
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// Options holds web package configuration
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type Options struct {
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MinPortUDP uint16
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MaxPortUDP uint16
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STUNServers []string
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}
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// SessionDescription contains SDP data
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// to initiate a WebRTC connection between one client and this app
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type SessionDescription = webrtc.SessionDescription
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var (
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videoTracks map[string][]*webrtc.Track
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audioTracks map[string][]*webrtc.Track
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2020-09-24 17:01:26 +00:00
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)
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2020-09-25 13:12:28 +00:00
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// Helper to reslice tracks
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func removeTrack(tracks []*webrtc.Track, track *webrtc.Track) []*webrtc.Track {
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for i, t := range tracks {
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if t == track {
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return append(tracks[:i], tracks[i+1:]...)
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}
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}
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return nil
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}
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2020-09-29 16:17:55 +00:00
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// GetNumberConnectedSessions get the number of currently connected clients
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2020-10-05 20:08:39 +00:00
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func GetNumberConnectedSessions(streamID string) int {
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return len(videoTracks[streamID])
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2020-09-29 16:17:55 +00:00
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}
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2020-09-24 17:01:26 +00:00
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// newPeerHandler is called when server receive a new session description
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// this initiates a WebRTC connection and return server description
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2020-10-05 20:00:08 +00:00
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func newPeerHandler(remoteSdp struct {
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StreamID string
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RemoteDescription webrtc.SessionDescription
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}, cfg *Options) webrtc.SessionDescription {
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// Create media engine using client SDP
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mediaEngine := webrtc.MediaEngine{}
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2020-10-05 20:00:08 +00:00
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if err := mediaEngine.PopulateFromSDP(remoteSdp.RemoteDescription); err != nil {
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2020-09-25 13:12:28 +00:00
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log.Println("Failed to create new media engine", err)
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return webrtc.SessionDescription{}
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}
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2020-09-24 17:01:26 +00:00
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// Create a new PeerConnection
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2020-09-29 12:11:56 +00:00
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settingsEngine := webrtc.SettingEngine{}
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2020-09-29 15:04:23 +00:00
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if err := settingsEngine.SetEphemeralUDPPortRange(cfg.MinPortUDP, cfg.MaxPortUDP); err != nil {
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2020-09-29 12:11:56 +00:00
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log.Println("Failed to set min/max UDP ports", err)
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return webrtc.SessionDescription{}
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}
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api := webrtc.NewAPI(
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webrtc.WithMediaEngine(mediaEngine),
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webrtc.WithSettingEngine(settingsEngine),
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)
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2020-09-25 10:03:40 +00:00
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peerConnection, err := api.NewPeerConnection(webrtc.Configuration{
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ICEServers: []webrtc.ICEServer{{URLs: cfg.STUNServers}},
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})
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if err != nil {
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log.Println("Failed to initiate peer connection", err)
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return webrtc.SessionDescription{}
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}
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2020-09-25 13:12:28 +00:00
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// Create video track
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codec, payloadType := getPayloadType(mediaEngine, webrtc.RTPCodecTypeVideo, "VP8")
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videoTrack, err := webrtc.NewTrack(payloadType, rand.Uint32(), "video", "pion", codec)
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if err != nil {
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log.Println("Failed to create new video track", err)
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2020-09-24 17:01:26 +00:00
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return webrtc.SessionDescription{}
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}
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if _, err = peerConnection.AddTrack(videoTrack); err != nil {
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log.Println("Failed to add video track", err)
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return webrtc.SessionDescription{}
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}
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2020-09-25 13:12:28 +00:00
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// Create audio track
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codec, payloadType = getPayloadType(mediaEngine, webrtc.RTPCodecTypeAudio, "opus")
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audioTrack, err := webrtc.NewTrack(payloadType, rand.Uint32(), "audio", "pion", codec)
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if err != nil {
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log.Println("Failed to create new audio track", err)
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return webrtc.SessionDescription{}
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}
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if _, err = peerConnection.AddTrack(audioTrack); err != nil {
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log.Println("Failed to add audio track", err)
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return webrtc.SessionDescription{}
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}
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2020-09-24 17:01:26 +00:00
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// Set the remote SessionDescription
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if err = peerConnection.SetRemoteDescription(remoteSdp.RemoteDescription); err != nil {
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2020-09-24 17:01:26 +00:00
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log.Println("Failed to set remote description", err)
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return webrtc.SessionDescription{}
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}
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// Create answer
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answer, err := peerConnection.CreateAnswer(nil)
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if err != nil {
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log.Println("Failed to create answer", err)
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return webrtc.SessionDescription{}
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}
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// Create channel that is blocked until ICE Gathering is complete
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gatherComplete := webrtc.GatheringCompletePromise(peerConnection)
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// Sets the LocalDescription, and starts our UDP listeners
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if err = peerConnection.SetLocalDescription(answer); err != nil {
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log.Println("Failed to set local description", err)
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return webrtc.SessionDescription{}
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}
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2020-10-05 20:00:08 +00:00
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streamID := remoteSdp.StreamID
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2020-09-25 13:12:28 +00:00
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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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peerConnection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
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log.Printf("Connection State has changed %s \n", connectionState.String())
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2020-10-05 20:00:08 +00:00
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if videoTracks[streamID] == nil {
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videoTracks[streamID] = make([]*webrtc.Track, 0, 1)
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}
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if audioTracks[streamID] == nil {
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audioTracks[streamID] = make([]*webrtc.Track, 0, 1)
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}
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if connectionState == webrtc.ICEConnectionStateConnected {
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// Register tracks
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videoTracks[streamID] = append(videoTracks[streamID], videoTrack)
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audioTracks[streamID] = append(audioTracks[streamID], audioTrack)
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2020-09-29 16:03:28 +00:00
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monitoring.WebRTCConnectedSessions.Inc()
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2020-09-25 13:12:28 +00:00
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} else if connectionState == webrtc.ICEConnectionStateDisconnected {
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// Unregister tracks
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2020-10-05 20:00:08 +00:00
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videoTracks[streamID] = removeTrack(videoTracks[streamID], videoTrack)
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audioTracks[streamID] = removeTrack(audioTracks[streamID], audioTrack)
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2020-09-29 16:03:28 +00:00
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monitoring.WebRTCConnectedSessions.Dec()
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2020-09-25 13:12:28 +00:00
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}
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})
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2020-09-24 17:01:26 +00:00
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// Block until ICE Gathering is complete, disabling trickle ICE
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// we do this because we only can exchange one signaling message
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// in a production application you should exchange ICE Candidates via OnICECandidate
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<-gatherComplete
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// Output the local description and send it to browser
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return *peerConnection.LocalDescription()
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}
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2020-09-24 09:24:13 +00:00
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// Search for Codec PayloadType
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//
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// Since we are answering we need to match the remote PayloadType
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2020-09-25 10:03:40 +00:00
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func getPayloadType(m webrtc.MediaEngine, codecType webrtc.RTPCodecType, codecName string) (*webrtc.RTPCodec, uint8) {
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2020-09-24 09:24:13 +00:00
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for _, codec := range m.GetCodecsByKind(codecType) {
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if codec.Name == codecName {
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2020-09-25 10:03:40 +00:00
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return codec, codec.PayloadType
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2020-09-24 09:24:13 +00:00
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}
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}
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panic(fmt.Sprintf("Remote peer does not support %s", codecName))
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}
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2020-09-25 13:12:28 +00:00
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// Serve WebRTC media streaming server
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2020-10-05 20:00:08 +00:00
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func Serve(remoteSdpChan chan struct {
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StreamID string
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RemoteDescription webrtc.SessionDescription
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}, localSdpChan chan webrtc.SessionDescription, inputChannel chan srt.Packet, cfg *Options) {
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2020-10-05 08:11:30 +00:00
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log.Printf("WebRTC server using UDP from port %d to %d", cfg.MinPortUDP, cfg.MaxPortUDP)
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2020-09-29 15:04:23 +00:00
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2020-10-05 20:00:08 +00:00
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// Allocate memory
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videoTracks = make(map[string][]*webrtc.Track)
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audioTracks = make(map[string][]*webrtc.Track)
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2020-10-05 08:11:30 +00:00
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// Ingest data from SRT
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go ingestFrom(inputChannel)
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2020-09-25 13:12:28 +00:00
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// Handle new connections
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for {
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// Wait for incoming session description
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// then send the local description to browser
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2020-10-04 18:16:29 +00:00
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localSdpChan <- newPeerHandler(<-remoteSdpChan, cfg)
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2020-09-25 13:12:28 +00:00
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}
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}
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