mirror of
https://github.com/bluenviron/mediamtx
synced 2024-12-16 11:44:50 +00:00
674 lines
14 KiB
Go
674 lines
14 KiB
Go
package core
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import (
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"context"
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"encoding/hex"
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"fmt"
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"net"
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"net/http"
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"strings"
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"sync"
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"time"
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"github.com/bluenviron/gortsplib/v4/pkg/description"
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"github.com/bluenviron/gortsplib/v4/pkg/rtptime"
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"github.com/google/uuid"
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"github.com/pion/sdp/v3"
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"github.com/pion/webrtc/v3"
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"github.com/bluenviron/mediamtx/internal/asyncwriter"
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"github.com/bluenviron/mediamtx/internal/externalcmd"
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"github.com/bluenviron/mediamtx/internal/logger"
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"github.com/bluenviron/mediamtx/internal/webrtcpc"
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)
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type trackRecvPair struct {
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track *webrtc.TrackRemote
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receiver *webrtc.RTPReceiver
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}
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func webrtcMediasOfOutgoingTracks(tracks []*webRTCOutgoingTrack) []*description.Media {
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ret := make([]*description.Media, len(tracks))
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for i, track := range tracks {
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ret[i] = track.media
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}
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return ret
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}
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func webrtcMediasOfIncomingTracks(tracks []*webRTCIncomingTrack) []*description.Media {
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ret := make([]*description.Media, len(tracks))
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for i, track := range tracks {
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ret[i] = track.media
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}
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return ret
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}
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func whipOffer(body []byte) *webrtc.SessionDescription {
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return &webrtc.SessionDescription{
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Type: webrtc.SDPTypeOffer,
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SDP: string(body),
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}
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}
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func webrtcWaitUntilConnected(
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ctx context.Context,
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pc *webrtcpc.PeerConnection,
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) error {
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t := time.NewTimer(webrtcHandshakeTimeout)
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defer t.Stop()
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outer:
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for {
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select {
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case <-t.C:
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return fmt.Errorf("deadline exceeded while waiting connection")
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case <-pc.Connected():
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break outer
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case <-ctx.Done():
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return fmt.Errorf("terminated")
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}
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}
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return nil
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}
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func webrtcGatherOutgoingTracks(desc *description.Session) ([]*webRTCOutgoingTrack, error) {
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var tracks []*webRTCOutgoingTrack
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videoTrack, err := newWebRTCOutgoingTrackVideo(desc)
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if err != nil {
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return nil, err
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}
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if videoTrack != nil {
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tracks = append(tracks, videoTrack)
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}
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audioTrack, err := newWebRTCOutgoingTrackAudio(desc)
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if err != nil {
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return nil, err
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}
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if audioTrack != nil {
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tracks = append(tracks, audioTrack)
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}
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if tracks == nil {
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return nil, fmt.Errorf(
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"the stream doesn't contain any supported codec, which are currently AV1, VP9, VP8, H264, Opus, G722, G711")
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}
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return tracks, nil
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}
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func webrtcTrackCount(medias []*sdp.MediaDescription) (int, error) {
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videoTrack := false
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audioTrack := false
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trackCount := 0
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for _, media := range medias {
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switch media.MediaName.Media {
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case "video":
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if videoTrack {
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return 0, fmt.Errorf("only a single video and a single audio track are supported")
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}
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videoTrack = true
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case "audio":
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if audioTrack {
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return 0, fmt.Errorf("only a single video and a single audio track are supported")
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}
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audioTrack = true
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default:
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return 0, fmt.Errorf("unsupported media '%s'", media.MediaName.Media)
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}
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trackCount++
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}
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return trackCount, nil
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}
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func webrtcGatherIncomingTracks(
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ctx context.Context,
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pc *webrtcpc.PeerConnection,
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trackRecv chan trackRecvPair,
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trackCount int,
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) ([]*webRTCIncomingTrack, error) {
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var tracks []*webRTCIncomingTrack
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t := time.NewTimer(webrtcTrackGatherTimeout)
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defer t.Stop()
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for {
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select {
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case <-t.C:
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if trackCount == 0 {
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return tracks, nil
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}
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return nil, fmt.Errorf("deadline exceeded while waiting tracks")
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case pair := <-trackRecv:
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track, err := newWebRTCIncomingTrack(pair.track, pair.receiver, pc.WriteRTCP)
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if err != nil {
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return nil, err
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}
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tracks = append(tracks, track)
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if len(tracks) == trackCount {
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return tracks, nil
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}
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case <-pc.Disconnected():
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return nil, fmt.Errorf("peer connection closed")
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case <-ctx.Done():
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return nil, fmt.Errorf("terminated")
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}
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}
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}
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type webRTCSessionPathManager interface {
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addPublisher(req pathAddPublisherReq) pathAddPublisherRes
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addReader(req pathAddReaderReq) pathAddReaderRes
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}
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type webRTCSession struct {
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writeQueueSize int
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api *webrtc.API
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req webRTCNewSessionReq
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wg *sync.WaitGroup
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externalCmdPool *externalcmd.Pool
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pathManager webRTCSessionPathManager
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parent *webRTCManager
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ctx context.Context
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ctxCancel func()
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created time.Time
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uuid uuid.UUID
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secret uuid.UUID
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mutex sync.RWMutex
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pc *webrtcpc.PeerConnection
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chNew chan webRTCNewSessionReq
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chAddCandidates chan webRTCAddSessionCandidatesReq
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}
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func newWebRTCSession(
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parentCtx context.Context,
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writeQueueSize int,
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api *webrtc.API,
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req webRTCNewSessionReq,
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wg *sync.WaitGroup,
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externalCmdPool *externalcmd.Pool,
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pathManager webRTCSessionPathManager,
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parent *webRTCManager,
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) *webRTCSession {
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ctx, ctxCancel := context.WithCancel(parentCtx)
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s := &webRTCSession{
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writeQueueSize: writeQueueSize,
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api: api,
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req: req,
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wg: wg,
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externalCmdPool: externalCmdPool,
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pathManager: pathManager,
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parent: parent,
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ctx: ctx,
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ctxCancel: ctxCancel,
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created: time.Now(),
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uuid: uuid.New(),
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secret: uuid.New(),
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chNew: make(chan webRTCNewSessionReq),
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chAddCandidates: make(chan webRTCAddSessionCandidatesReq),
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}
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s.Log(logger.Info, "created by %s", req.remoteAddr)
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wg.Add(1)
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go s.run()
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return s
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}
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func (s *webRTCSession) Log(level logger.Level, format string, args ...interface{}) {
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id := hex.EncodeToString(s.uuid[:4])
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s.parent.Log(level, "[session %v] "+format, append([]interface{}{id}, args...)...)
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}
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func (s *webRTCSession) close() {
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s.ctxCancel()
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}
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func (s *webRTCSession) run() {
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defer s.wg.Done()
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err := s.runInner()
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s.ctxCancel()
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s.parent.closeSession(s)
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s.Log(logger.Info, "closed: %v", err)
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}
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func (s *webRTCSession) runInner() error {
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select {
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case <-s.chNew:
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case <-s.ctx.Done():
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return fmt.Errorf("terminated")
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}
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errStatusCode, err := s.runInner2()
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if errStatusCode != 0 {
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s.req.res <- webRTCNewSessionRes{
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err: err,
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errStatusCode: errStatusCode,
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}
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}
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return err
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}
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func (s *webRTCSession) runInner2() (int, error) {
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if s.req.publish {
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return s.runPublish()
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}
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return s.runRead()
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}
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func (s *webRTCSession) runPublish() (int, error) {
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ip, _, _ := net.SplitHostPort(s.req.remoteAddr)
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res := s.pathManager.addPublisher(pathAddPublisherReq{
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author: s,
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pathName: s.req.pathName,
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credentials: authCredentials{
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query: s.req.query,
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ip: net.ParseIP(ip),
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user: s.req.user,
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pass: s.req.pass,
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proto: authProtocolWebRTC,
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id: &s.uuid,
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},
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})
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if res.err != nil {
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if _, ok := res.err.(*errAuthentication); ok {
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// wait some seconds to stop brute force attacks
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<-time.After(webrtcPauseAfterAuthError)
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return http.StatusUnauthorized, res.err
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}
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return http.StatusBadRequest, res.err
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}
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defer res.path.removePublisher(pathRemovePublisherReq{author: s})
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servers, err := s.parent.generateICEServers()
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if err != nil {
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return http.StatusInternalServerError, err
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}
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pc, err := webrtcpc.New(
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servers,
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s.api,
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s)
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if err != nil {
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return http.StatusBadRequest, err
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}
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defer pc.Close()
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offer := whipOffer(s.req.offer)
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var sdp sdp.SessionDescription
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err = sdp.Unmarshal([]byte(offer.SDP))
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if err != nil {
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return http.StatusBadRequest, err
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}
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trackCount, err := webrtcTrackCount(sdp.MediaDescriptions)
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if err != nil {
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return http.StatusBadRequest, err
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}
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_, err = pc.AddTransceiverFromKind(webrtc.RTPCodecTypeVideo, webrtc.RtpTransceiverInit{
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Direction: webrtc.RTPTransceiverDirectionRecvonly,
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})
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if err != nil {
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return http.StatusBadRequest, err
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}
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_, err = pc.AddTransceiverFromKind(webrtc.RTPCodecTypeAudio, webrtc.RtpTransceiverInit{
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Direction: webrtc.RTPTransceiverDirectionRecvonly,
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})
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if err != nil {
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return http.StatusBadRequest, err
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}
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trackRecv := make(chan trackRecvPair)
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pc.OnTrack(func(track *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) {
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select {
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case trackRecv <- trackRecvPair{track, receiver}:
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case <-s.ctx.Done():
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}
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})
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err = pc.SetRemoteDescription(*offer)
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if err != nil {
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return http.StatusBadRequest, err
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}
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answer, err := pc.CreateAnswer(nil)
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if err != nil {
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return http.StatusBadRequest, err
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}
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err = pc.SetLocalDescription(answer)
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if err != nil {
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return http.StatusBadRequest, err
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}
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err = pc.WaitGatheringDone(s.ctx)
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if err != nil {
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return http.StatusBadRequest, err
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}
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s.writeAnswer(pc.LocalDescription())
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go s.readRemoteCandidates(pc)
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err = webrtcWaitUntilConnected(s.ctx, pc)
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if err != nil {
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return 0, err
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}
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s.mutex.Lock()
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s.pc = pc
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s.mutex.Unlock()
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tracks, err := webrtcGatherIncomingTracks(s.ctx, pc, trackRecv, trackCount)
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if err != nil {
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return 0, err
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}
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medias := webrtcMediasOfIncomingTracks(tracks)
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rres := res.path.startPublisher(pathStartPublisherReq{
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author: s,
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desc: &description.Session{Medias: medias},
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generateRTPPackets: false,
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})
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if rres.err != nil {
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return 0, rres.err
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}
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timeDecoder := rtptime.NewGlobalDecoder()
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for _, track := range tracks {
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track.start(rres.stream, timeDecoder)
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}
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select {
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case <-pc.Disconnected():
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return 0, fmt.Errorf("peer connection closed")
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case <-s.ctx.Done():
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return 0, fmt.Errorf("terminated")
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}
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}
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func (s *webRTCSession) runRead() (int, error) {
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ip, _, _ := net.SplitHostPort(s.req.remoteAddr)
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res := s.pathManager.addReader(pathAddReaderReq{
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author: s,
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pathName: s.req.pathName,
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credentials: authCredentials{
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query: s.req.query,
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ip: net.ParseIP(ip),
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user: s.req.user,
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pass: s.req.pass,
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proto: authProtocolWebRTC,
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id: &s.uuid,
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},
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})
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if res.err != nil {
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if _, ok := res.err.(*errAuthentication); ok {
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// wait some seconds to stop brute force attacks
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<-time.After(webrtcPauseAfterAuthError)
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return http.StatusUnauthorized, res.err
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}
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if strings.HasPrefix(res.err.Error(), "no one is publishing") {
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return http.StatusNotFound, res.err
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}
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return http.StatusBadRequest, res.err
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}
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defer res.path.removeReader(pathRemoveReaderReq{author: s})
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tracks, err := webrtcGatherOutgoingTracks(res.stream.Desc())
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if err != nil {
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return http.StatusBadRequest, err
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}
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servers, err := s.parent.generateICEServers()
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if err != nil {
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return http.StatusInternalServerError, err
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}
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pc, err := webrtcpc.New(
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servers,
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s.api,
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s)
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if err != nil {
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return http.StatusBadRequest, err
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}
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defer pc.Close()
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for _, track := range tracks {
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var err error
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track.sender, err = pc.AddTrack(track.track)
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if err != nil {
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return http.StatusBadRequest, err
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}
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}
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offer := whipOffer(s.req.offer)
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err = pc.SetRemoteDescription(*offer)
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if err != nil {
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return http.StatusBadRequest, err
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}
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answer, err := pc.CreateAnswer(nil)
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if err != nil {
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return http.StatusBadRequest, err
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}
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err = pc.SetLocalDescription(answer)
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if err != nil {
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return http.StatusBadRequest, err
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}
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err = pc.WaitGatheringDone(s.ctx)
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if err != nil {
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return http.StatusBadRequest, err
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}
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s.writeAnswer(pc.LocalDescription())
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go s.readRemoteCandidates(pc)
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err = webrtcWaitUntilConnected(s.ctx, pc)
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if err != nil {
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return 0, err
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}
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s.mutex.Lock()
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s.pc = pc
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s.mutex.Unlock()
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writer := asyncwriter.New(s.writeQueueSize, s)
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defer res.stream.RemoveReader(writer)
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for _, track := range tracks {
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track.start(res.stream, writer)
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}
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s.Log(logger.Info, "is reading from path '%s', %s",
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res.path.name, sourceMediaInfo(webrtcMediasOfOutgoingTracks(tracks)))
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pathConf := res.path.safeConf()
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if pathConf.RunOnRead != "" {
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s.Log(logger.Info, "runOnRead command started")
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onReadCmd := externalcmd.NewCmd(
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s.externalCmdPool,
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pathConf.RunOnRead,
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pathConf.RunOnReadRestart,
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res.path.externalCmdEnv(),
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func(err error) {
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s.Log(logger.Info, "runOnRead command exited: %v", err)
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})
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defer func() {
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onReadCmd.Close()
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s.Log(logger.Info, "runOnRead command stopped")
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}()
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}
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if pathConf.RunOnUnread != "" {
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defer func() {
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s.Log(logger.Info, "runOnUnread command launched")
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externalcmd.NewCmd(
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s.externalCmdPool,
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pathConf.RunOnUnread,
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false,
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res.path.externalCmdEnv(),
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nil)
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}()
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}
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writer.Start()
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select {
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case <-pc.Disconnected():
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writer.Stop()
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return 0, fmt.Errorf("peer connection closed")
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|
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case err := <-writer.Error():
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return 0, err
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|
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case <-s.ctx.Done():
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writer.Stop()
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return 0, fmt.Errorf("terminated")
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}
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}
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|
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func (s *webRTCSession) writeAnswer(answer *webrtc.SessionDescription) {
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s.req.res <- webRTCNewSessionRes{
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sx: s,
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answer: []byte(answer.SDP),
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}
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}
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|
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func (s *webRTCSession) readRemoteCandidates(pc *webrtcpc.PeerConnection) {
|
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for {
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select {
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case req := <-s.chAddCandidates:
|
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for _, candidate := range req.candidates {
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err := pc.AddICECandidate(*candidate)
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if err != nil {
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req.res <- webRTCAddSessionCandidatesRes{err: err}
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}
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}
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req.res <- webRTCAddSessionCandidatesRes{}
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|
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case <-s.ctx.Done():
|
|
return
|
|
}
|
|
}
|
|
}
|
|
|
|
// new is called by webRTCHTTPServer through webRTCManager.
|
|
func (s *webRTCSession) new(req webRTCNewSessionReq) webRTCNewSessionRes {
|
|
select {
|
|
case s.chNew <- req:
|
|
return <-req.res
|
|
|
|
case <-s.ctx.Done():
|
|
return webRTCNewSessionRes{err: fmt.Errorf("terminated"), errStatusCode: http.StatusInternalServerError}
|
|
}
|
|
}
|
|
|
|
// addCandidates is called by webRTCHTTPServer through webRTCManager.
|
|
func (s *webRTCSession) addCandidates(
|
|
req webRTCAddSessionCandidatesReq,
|
|
) webRTCAddSessionCandidatesRes {
|
|
select {
|
|
case s.chAddCandidates <- req:
|
|
return <-req.res
|
|
|
|
case <-s.ctx.Done():
|
|
return webRTCAddSessionCandidatesRes{err: fmt.Errorf("terminated")}
|
|
}
|
|
}
|
|
|
|
// apiSourceDescribe implements sourceStaticImpl.
|
|
func (s *webRTCSession) apiSourceDescribe() pathAPISourceOrReader {
|
|
return pathAPISourceOrReader{
|
|
Type: "webRTCSession",
|
|
ID: s.uuid.String(),
|
|
}
|
|
}
|
|
|
|
// apiReaderDescribe implements reader.
|
|
func (s *webRTCSession) apiReaderDescribe() pathAPISourceOrReader {
|
|
return s.apiSourceDescribe()
|
|
}
|
|
|
|
func (s *webRTCSession) apiItem() *apiWebRTCSession {
|
|
s.mutex.RLock()
|
|
defer s.mutex.RUnlock()
|
|
|
|
peerConnectionEstablished := false
|
|
localCandidate := ""
|
|
remoteCandidate := ""
|
|
bytesReceived := uint64(0)
|
|
bytesSent := uint64(0)
|
|
|
|
if s.pc != nil {
|
|
peerConnectionEstablished = true
|
|
localCandidate = s.pc.LocalCandidate()
|
|
remoteCandidate = s.pc.RemoteCandidate()
|
|
bytesReceived = s.pc.BytesReceived()
|
|
bytesSent = s.pc.BytesSent()
|
|
}
|
|
|
|
return &apiWebRTCSession{
|
|
ID: s.uuid,
|
|
Created: s.created,
|
|
RemoteAddr: s.req.remoteAddr,
|
|
PeerConnectionEstablished: peerConnectionEstablished,
|
|
LocalCandidate: localCandidate,
|
|
RemoteCandidate: remoteCandidate,
|
|
State: func() apiWebRTCSessionState {
|
|
if s.req.publish {
|
|
return apiWebRTCSessionStatePublish
|
|
}
|
|
return apiWebRTCSessionStateRead
|
|
}(),
|
|
Path: s.req.pathName,
|
|
BytesReceived: bytesReceived,
|
|
BytesSent: bytesSent,
|
|
}
|
|
}
|