package core import ( "context" "encoding/hex" "encoding/json" "fmt" "strconv" "sync" "time" "github.com/bluenviron/gortsplib/v3/pkg/media" "github.com/bluenviron/gortsplib/v3/pkg/ringbuffer" "github.com/google/uuid" "github.com/pion/ice/v2" "github.com/pion/sdp/v3" "github.com/pion/webrtc/v3" "github.com/aler9/mediamtx/internal/logger" ) const ( webrtcHandshakeTimeout = 10 * time.Second webrtcTrackGatherTimeout = 2 * time.Second webrtcPayloadMaxSize = 1188 // 1200 - 12 (RTP header) webrtcStreamID = "mediamtx" ) type trackRecvPair struct { track *webrtc.TrackRemote receiver *webrtc.RTPReceiver } func mediasOfOutgoingTracks(tracks []*webRTCOutgoingTrack) media.Medias { ret := make(media.Medias, len(tracks)) for i, track := range tracks { ret[i] = track.media } return ret } func mediasOfIncomingTracks(tracks []*webRTCIncomingTrack) media.Medias { ret := make(media.Medias, len(tracks)) for i, track := range tracks { ret[i] = track.media } return ret } func insertTias(offer *webrtc.SessionDescription, value uint64) { var sd sdp.SessionDescription err := sd.Unmarshal([]byte(offer.SDP)) if err != nil { return } for _, media := range sd.MediaDescriptions { if media.MediaName.Media == "video" { media.Bandwidth = append(media.Bandwidth, sdp.Bandwidth{ Type: "TIAS", Bandwidth: value, }) } } enc, err := sd.Marshal() if err != nil { return } offer.SDP = string(enc) } func gatherOutgoingTracks(medias media.Medias) ([]*webRTCOutgoingTrack, error) { var tracks []*webRTCOutgoingTrack videoTrack, err := newWebRTCOutgoingTrackVideo(medias) if err != nil { return nil, err } if videoTrack != nil { tracks = append(tracks, videoTrack) } audioTrack, err := newWebRTCOutgoingTrackAudio(medias) if err != nil { return nil, err } if audioTrack != nil { tracks = append(tracks, audioTrack) } if tracks == nil { return nil, fmt.Errorf( "the stream doesn't contain any supported codec, which are currently H264, VP8, VP9, G711, G722, Opus") } return tracks, nil } func gatherIncomingTracks( ctx context.Context, pc *peerConnection, trackRecv chan trackRecvPair, ) ([]*webRTCIncomingTrack, error) { var tracks []*webRTCIncomingTrack t := time.NewTimer(webrtcTrackGatherTimeout) defer t.Stop() for { select { case <-t.C: return tracks, nil case pair := <-trackRecv: track, err := newWebRTCIncomingTrack(pair.track, pair.receiver, pc.WriteRTCP) if err != nil { return nil, err } tracks = append(tracks, track) if len(tracks) == 2 { return tracks, nil } case <-pc.disconnected: return nil, fmt.Errorf("peer connection closed") case <-ctx.Done(): return nil, fmt.Errorf("terminated") } } } type webRTCSessionPathManager interface { publisherAdd(req pathPublisherAddReq) pathPublisherAnnounceRes readerAdd(req pathReaderAddReq) pathReaderSetupPlayRes } type webRTCSession struct { readBufferCount int req webRTCSessionNewReq wg *sync.WaitGroup iceHostNAT1To1IPs []string iceUDPMux ice.UDPMux iceTCPMux ice.TCPMux pathManager webRTCSessionPathManager parent *webRTCManager ctx context.Context ctxCancel func() created time.Time uuid uuid.UUID secret uuid.UUID answerSent bool pcMutex sync.RWMutex pc *peerConnection chAddRemoteCandidates chan webRTCSessionAddCandidatesReq } func newWebRTCSession( parentCtx context.Context, readBufferCount int, req webRTCSessionNewReq, wg *sync.WaitGroup, iceHostNAT1To1IPs []string, iceUDPMux ice.UDPMux, iceTCPMux ice.TCPMux, pathManager webRTCSessionPathManager, parent *webRTCManager, ) *webRTCSession { ctx, ctxCancel := context.WithCancel(parentCtx) s := &webRTCSession{ readBufferCount: readBufferCount, req: req, wg: wg, iceHostNAT1To1IPs: iceHostNAT1To1IPs, iceUDPMux: iceUDPMux, iceTCPMux: iceTCPMux, parent: parent, pathManager: pathManager, ctx: ctx, ctxCancel: ctxCancel, created: time.Now(), uuid: uuid.New(), secret: uuid.New(), chAddRemoteCandidates: make(chan webRTCSessionAddCandidatesReq), } s.Log(logger.Info, "created by %s", req.remoteAddr) wg.Add(1) go s.run() return s } func (s *webRTCSession) Log(level logger.Level, format string, args ...interface{}) { id := hex.EncodeToString(s.uuid[:4]) s.parent.Log(level, "[session %v] "+format, append([]interface{}{id}, args...)...) } func (s *webRTCSession) close() { s.ctxCancel() } func (s *webRTCSession) safePC() *peerConnection { s.pcMutex.RLock() defer s.pcMutex.RUnlock() return s.pc } func (s *webRTCSession) run() { defer s.wg.Done() err := s.runInner() if !s.answerSent { select { case s.req.res <- webRTCNewSessionRes{ err: err, }: case <-s.ctx.Done(): } } s.parent.sessionClose(s) s.Log(logger.Info, "closed (%v)", err) } func (s *webRTCSession) runInner() error { if s.req.publish { return s.runPublish() } return s.runRead() } func (s *webRTCSession) runPublish() error { res := s.pathManager.publisherAdd(pathPublisherAddReq{ author: s, pathName: s.req.pathName, skipAuth: true, }) if res.err != nil { return res.err } defer res.path.publisherRemove(pathPublisherRemoveReq{author: s}) offer, err := s.decodeOffer() if err != nil { return err } pc, err := newPeerConnection( s.req.videoCodec, s.req.audioCodec, s.parent.genICEServers(), s.iceHostNAT1To1IPs, s.iceUDPMux, s.iceTCPMux, s) if err != nil { return err } defer pc.close() _, err = pc.AddTransceiverFromKind(webrtc.RTPCodecTypeVideo, webrtc.RtpTransceiverInit{ Direction: webrtc.RTPTransceiverDirectionRecvonly, }) if err != nil { return err } _, err = pc.AddTransceiverFromKind(webrtc.RTPCodecTypeAudio, webrtc.RtpTransceiverInit{ Direction: webrtc.RTPTransceiverDirectionRecvonly, }) if err != nil { return err } trackRecv := make(chan trackRecvPair) pc.OnTrack(func(track *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) { select { case trackRecv <- trackRecvPair{track, receiver}: case <-pc.closed: } }) err = pc.SetRemoteDescription(*offer) if err != nil { return err } answer, err := pc.CreateAnswer(nil) if err != nil { return err } err = pc.SetLocalDescription(answer) if err != nil { return err } if s.req.videoBitrate != "" { tmp, err := strconv.ParseUint(s.req.videoBitrate, 10, 31) if err != nil { return err } insertTias(&answer, tmp*1024) } err = s.waitGatheringDone(pc) if err != nil { return err } err = s.writeAnswer(pc.LocalDescription()) if err != nil { return err } go s.readRemoteCandidates(pc) err = s.waitUntilConnected(pc) if err != nil { return err } tracks, err := gatherIncomingTracks(s.ctx, pc, trackRecv) if err != nil { return err } medias := mediasOfIncomingTracks(tracks) rres := res.path.publisherStart(pathPublisherStartReq{ author: s, medias: medias, generateRTPPackets: false, }) if rres.err != nil { return rres.err } s.Log(logger.Info, "is publishing to path '%s', %s", res.path.name, sourceMediaInfo(medias)) for _, track := range tracks { track.start(rres.stream) } select { case <-pc.disconnected: return fmt.Errorf("peer connection closed") case <-s.ctx.Done(): return fmt.Errorf("terminated") } } func (s *webRTCSession) runRead() error { res := s.pathManager.readerAdd(pathReaderAddReq{ author: s, pathName: s.req.pathName, skipAuth: true, }) if res.err != nil { return res.err } defer res.path.readerRemove(pathReaderRemoveReq{author: s}) tracks, err := gatherOutgoingTracks(res.stream.medias()) if err != nil { return err } offer, err := s.decodeOffer() if err != nil { return err } pc, err := newPeerConnection( "", "", s.parent.genICEServers(), s.iceHostNAT1To1IPs, s.iceUDPMux, s.iceTCPMux, s) if err != nil { return err } defer pc.close() for _, track := range tracks { var err error track.sender, err = pc.AddTrack(track.track) if err != nil { return err } } err = pc.SetRemoteDescription(*offer) if err != nil { return err } answer, err := pc.CreateAnswer(nil) if err != nil { return err } err = pc.SetLocalDescription(answer) if err != nil { return err } err = s.waitGatheringDone(pc) if err != nil { return err } err = s.writeAnswer(pc.LocalDescription()) if err != nil { return err } go s.readRemoteCandidates(pc) err = s.waitUntilConnected(pc) if err != nil { return err } ringBuffer, _ := ringbuffer.New(uint64(s.readBufferCount)) defer ringBuffer.Close() writeError := make(chan error) for _, track := range tracks { track.start(s.ctx, s, res.stream, ringBuffer, writeError) } defer res.stream.readerRemove(s) s.Log(logger.Info, "is reading from path '%s', %s", res.path.name, sourceMediaInfo(mediasOfOutgoingTracks(tracks))) go func() { for { item, ok := ringBuffer.Pull() if !ok { return } item.(func())() } }() select { case <-pc.disconnected: return fmt.Errorf("peer connection closed") case err := <-writeError: return err case <-s.ctx.Done(): return fmt.Errorf("terminated") } } func (s *webRTCSession) decodeOffer() (*webrtc.SessionDescription, error) { var offer webrtc.SessionDescription err := json.Unmarshal(s.req.offer, &offer) if err != nil { return nil, err } if offer.Type != webrtc.SDPTypeOffer { return nil, fmt.Errorf("received SDP is not an offer") } return &offer, nil } func (s *webRTCSession) waitGatheringDone(pc *peerConnection) error { for { select { case <-pc.localCandidateRecv: case <-pc.gatheringDone: return nil case <-s.ctx.Done(): return fmt.Errorf("terminated") } } } func (s *webRTCSession) writeAnswer(answer *webrtc.SessionDescription) error { enc, err := json.Marshal(answer) if err != nil { return err } select { case s.req.res <- webRTCNewSessionRes{ sx: s, answer: enc, }: s.answerSent = true case <-s.ctx.Done(): return fmt.Errorf("terminated") } return nil } func (s *webRTCSession) waitUntilConnected(pc *peerConnection) error { t := time.NewTimer(webrtcHandshakeTimeout) defer t.Stop() outer: for { select { case <-t.C: return fmt.Errorf("deadline exceeded") case <-pc.connected: break outer case <-s.ctx.Done(): return fmt.Errorf("terminated") } } s.pcMutex.Lock() s.pc = pc s.pcMutex.Unlock() return nil } func (s *webRTCSession) readRemoteCandidates(pc *peerConnection) { for { select { case req := <-s.chAddRemoteCandidates: for _, candidate := range req.candidates { err := pc.AddICECandidate(*candidate) if err != nil { req.res <- webRTCSessionAddCandidatesRes{err: err} } } req.res <- webRTCSessionAddCandidatesRes{} case <-s.ctx.Done(): return } } } func (s *webRTCSession) addRemoteCandidates( req webRTCSessionAddCandidatesReq, ) webRTCSessionAddCandidatesRes { select { case s.chAddRemoteCandidates <- req: return <-req.res case <-s.ctx.Done(): return webRTCSessionAddCandidatesRes{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() }