webrtc: print lost packets (#2468)

internal/core/webrtc_incoming_track.go

@@ -7,11 +7,14 @@ import (
 
 	"github.com/bluenviron/gortsplib/v4/pkg/description"
 	"github.com/bluenviron/gortsplib/v4/pkg/format"
+	"github.com/bluenviron/gortsplib/v4/pkg/liberrors"
+	"github.com/bluenviron/gortsplib/v4/pkg/rtplossdetector"
 	"github.com/bluenviron/gortsplib/v4/pkg/rtptime"
 	"github.com/pion/rtcp"
 	"github.com/pion/rtp"
 	"github.com/pion/webrtc/v3"
 
+	"github.com/bluenviron/mediamtx/internal/logger"
 	"github.com/bluenviron/mediamtx/internal/stream"
 )
 
@@ -112,7 +115,8 @@ func (webrtcTrackWrapper) PTSEqualsDTS(*rtp.Packet) bool {
 	return true
 }
 
-func (t *webRTCIncomingTrack) start(stream *stream.Stream, timeDecoder *rtptime.GlobalDecoder) {
+func (t *webRTCIncomingTrack) start(stream *stream.Stream, timeDecoder *rtptime.GlobalDecoder, log logger.Writer) {
+	lossDetector := rtplossdetector.New()
 	trackWrapper := &webrtcTrackWrapper{clockRate: int(t.track.Codec().ClockRate)}
 
 	go func() {
@@ -122,6 +126,12 @@ func (t *webRTCIncomingTrack) start(stream *stream.Stream, timeDecoder *rtptime.
 				return
 			}
 
+			lost := lossDetector.Process(pkt)
+			if lost != 0 {
+				log.Log(logger.Warn, (liberrors.ErrClientRTPPacketsLost{Lost: lost}).Error())
+				// do not return
+			}
+
 			// sometimes Chrome sends empty RTP packets. ignore them.
 			if len(pkt.Payload) == 0 {
 				continue

internal/core/webrtc_session.go

@@ -410,7 +410,7 @@ func (s *webRTCSession) runPublish() (int, error) {
 	timeDecoder := rtptime.NewGlobalDecoder()
 
 	for _, track := range tracks {
-		track.start(rres.stream, timeDecoder)
+		track.start(rres.stream, timeDecoder, s)
 	}
 
 	select {

internal/core/webrtc_source.go

@@ -158,7 +158,7 @@ func (s *webRTCSource) run(ctx context.Context, cnf *conf.PathConf, _ chan *conf
 	timeDecoder := rtptime.NewGlobalDecoder()
 
 	for _, track := range tracks {
-		track.start(rres.stream, timeDecoder)
+		track.start(rres.stream, timeDecoder, s)
 	}
 
 	select {