webrtc: speed up track detection (#2105)

internal/core/webrtc_manager.go

@@ -25,7 +25,7 @@ import (
 const (
 	webrtcPauseAfterAuthError  = 2 * time.Second
 	webrtcHandshakeTimeout     = 10 * time.Second
-	webrtcTrackGatherTimeout   = 2 * time.Second
+	webrtcTrackGatherTimeout   = 5 * time.Second
 	webrtcPayloadMaxSize       = 1188 // 1200 - 12 (RTP header)
 	webrtcStreamID             = "mediamtx"
 	webrtcTurnSecretExpiration = 24 * 3600 * time.Second

internal/core/webrtc_session.go

@@ -14,6 +14,7 @@ import (
 	"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/bluenviron/mediamtx/internal/logger"
@@ -40,6 +41,30 @@ func mediasOfIncomingTracks(tracks []*webRTCIncomingTrack) media.Medias {
 	return ret
 }
 
+func waitUntilConnected(
+	ctx context.Context,
+	pc *peerConnection,
+) error {
+	t := time.NewTimer(webrtcHandshakeTimeout)
+	defer t.Stop()
+
+outer:
+	for {
+		select {
+		case <-t.C:
+			return fmt.Errorf("deadline exceeded while waiting connection")
+
+		case <-pc.connected:
+			break outer
+
+		case <-ctx.Done():
+			return fmt.Errorf("terminated")
+		}
+	}
+
+	return nil
+}
+
 func gatherOutgoingTracks(medias media.Medias) ([]*webRTCOutgoingTrack, error) {
 	var tracks []*webRTCOutgoingTrack
 
@@ -73,6 +98,7 @@ func gatherIncomingTracks(
 	ctx context.Context,
 	pc *peerConnection,
 	trackRecv chan trackRecvPair,
+	trackCount int,
 ) ([]*webRTCIncomingTrack, error) {
 	var tracks []*webRTCIncomingTrack
 
@@ -82,10 +108,7 @@ func gatherIncomingTracks(
 	for {
 		select {
 		case <-t.C:
-			if len(tracks) == 0 {
+			return nil, fmt.Errorf("deadline exceeded while waiting tracks")
-				return nil, fmt.Errorf("no tracks found")
-			}
-			return tracks, nil
 
 		case pair := <-trackRecv:
 			track, err := newWebRTCIncomingTrack(pair.track, pair.receiver, pc.WriteRTCP)
@@ -94,7 +117,7 @@ func gatherIncomingTracks(
 			}
 			tracks = append(tracks, track)
 
-			if len(tracks) == 2 {
+			if len(tracks) == trackCount {
 				return tracks, nil
 			}
 
@@ -262,6 +285,39 @@ func (s *webRTCSession) runPublish() (int, error) {
 	}
 	defer pc.close()
 
+	offer := s.offer()
+
+	var sdp sdp.SessionDescription
+	err = sdp.Unmarshal([]byte(offer.SDP))
+	if err != nil {
+		return http.StatusBadRequest, err
+	}
+
+	videoTrack := false
+	audioTrack := false
+	trackCount := 0
+
+	for _, media := range sdp.MediaDescriptions {
+		switch media.MediaName.Media {
+		case "video":
+			if videoTrack {
+				return http.StatusBadRequest, fmt.Errorf("only a single video and a single audio track are supported")
+			}
+			videoTrack = true
+
+		case "audio":
+			if audioTrack {
+				return http.StatusBadRequest, fmt.Errorf("only a single video and a single audio track are supported")
+			}
+			audioTrack = true
+
+		default:
+			return http.StatusBadRequest, fmt.Errorf("unsupported media '%s'", media.MediaName.Media)
+		}
+
+		trackCount++
+	}
+
 	_, err = pc.AddTransceiverFromKind(webrtc.RTPCodecTypeVideo, webrtc.RtpTransceiverInit{
 		Direction: webrtc.RTPTransceiverDirectionRecvonly,
 	})
@@ -285,7 +341,6 @@ func (s *webRTCSession) runPublish() (int, error) {
 		}
 	})
 
-	offer := s.buildOffer()
 	err = pc.SetRemoteDescription(*offer)
 	if err != nil {
 		return http.StatusBadRequest, err
@@ -313,12 +368,16 @@ func (s *webRTCSession) runPublish() (int, error) {
 
 	go s.readRemoteCandidates(pc)
 
-	err = s.waitUntilConnected(pc)
+	err = waitUntilConnected(s.ctx, pc)
 	if err != nil {
 		return 0, err
 	}
 
-	tracks, err := gatherIncomingTracks(s.ctx, pc, trackRecv)
+	s.mutex.Lock()
+	s.pc = pc
+	s.mutex.Unlock()
+
+	tracks, err := gatherIncomingTracks(s.ctx, pc, trackRecv, trackCount)
 	if err != nil {
 		return 0, err
 	}
@@ -406,7 +465,8 @@ func (s *webRTCSession) runRead() (int, error) {
 		}
 	}
 
-	offer := s.buildOffer()
+	offer := s.offer()
+
 	err = pc.SetRemoteDescription(*offer)
 	if err != nil {
 		return http.StatusBadRequest, err
@@ -434,11 +494,15 @@ func (s *webRTCSession) runRead() (int, error) {
 
 	go s.readRemoteCandidates(pc)
 
-	err = s.waitUntilConnected(pc)
+	err = waitUntilConnected(s.ctx, pc)
 	if err != nil {
 		return 0, err
 	}
 
+	s.mutex.Lock()
+	s.pc = pc
+	s.mutex.Unlock()
+
 	ringBuffer, _ := ringbuffer.New(uint64(s.readBufferCount))
 	defer ringBuffer.Close()
 
@@ -475,7 +539,7 @@ func (s *webRTCSession) runRead() (int, error) {
 	}
 }
 
-func (s *webRTCSession) buildOffer() *webrtc.SessionDescription {
+func (s *webRTCSession) offer() *webrtc.SessionDescription {
 	return &webrtc.SessionDescription{
 		Type: webrtc.SDPTypeOffer,
 		SDP:  string(s.req.offer),
@@ -502,31 +566,6 @@ func (s *webRTCSession) writeAnswer(answer *webrtc.SessionDescription) {
 	s.answerSent = true
 }
 
-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.mutex.Lock()
-	s.pc = pc
-	s.mutex.Unlock()
-
-	return nil
-}
-
 func (s *webRTCSession) readRemoteCandidates(pc *peerConnection) {
 	for {
 		select {