Compute down track RTT.

conn.go

@@ -281,6 +281,9 @@ type rtpDownTrack struct {
 	maxREMBBitrate *bitrate
 	rate           *estimator.Estimator
 	stats          *receiverStats
+	srTime         uint64
+	srNTPTime      uint64
+	rtt            uint64
 }
 
 func (down *rtpDownTrack) WriteRTP(packet *rtp.Packet) error {

group.go

@@ -518,6 +518,7 @@ type trackStats struct {
 	bitrate    uint64
 	maxBitrate uint64
 	loss       uint8
+	rtt        time.Duration
 	jitter     time.Duration
 }
 
@@ -589,6 +590,8 @@ func getClientStats(c *webClient) clientStats {
 		conns := connStats{id: down.id}
 		for _, t := range down.tracks {
 			jiffies := rtptime.Jiffies()
+			rtt := rtptime.ToDuration(atomic.LoadUint64(&t.rtt),
+				rtptime.JiffiesPerSec)
 			loss, jitter := t.stats.Get(jiffies)
 			j := time.Duration(jitter) * time.Second /
 				time.Duration(t.track.Codec().ClockRate)
@@ -596,6 +599,7 @@ func getClientStats(c *webClient) clientStats {
 				bitrate:    uint64(t.rate.Estimate()) * 8,
 				maxBitrate: t.GetMaxBitrate(jiffies),
 				loss:       uint8(uint32(loss) * 100 / 256),
+				rtt:        rtt,
 				jitter:     j,
 			})
 		}

rtptime/rtptime.go

@@ -10,7 +10,7 @@ func FromDuration(d time.Duration, hz uint32) uint64 {
 	return uint64(d) * uint64(hz) / uint64(time.Second)
 }
 
-func toDuration(tm uint64, hz uint32) time.Duration {
+func ToDuration(tm uint64, hz uint32) time.Duration {
 	return time.Duration(tm * uint64(time.Second) / uint64(hz))
 }
 
@@ -29,6 +29,10 @@ func Jiffies() uint64 {
 	return Now(JiffiesPerSec)
 }
 
+func TimeToJiffies(tm time.Time) uint64 {
+	return FromDuration(tm.Sub(epoch), JiffiesPerSec)
+}
+
 var ntpEpoch = time.Date(1900, 1, 1, 0, 0, 0, 0, time.UTC)
 
 func NTPToTime(ntp uint64) time.Time {

rtptime/rtptime_test.go

@@ -11,7 +11,7 @@ func TestDuration(t *testing.T) {
 		t.Errorf("Expected 48000, got %v", a)
 	}
 
-	b := toDuration(48000, 48000)
+	b := ToDuration(48000, 48000)
 	if b != time.Second {
 		t.Errorf("Expected %v, got %v", time.Second, b)
 	}

webclient.go

@@ -20,8 +20,8 @@ import (
 
 	"sfu/estimator"
 	"sfu/jitter"
-	"sfu/rtptime"
 	"sfu/packetcache"
+	"sfu/rtptime"
 
 	"github.com/gorilla/websocket"
 	"github.com/pion/rtcp"
@@ -633,6 +633,7 @@ func sendSR(conn *rtpDownConnection) error {
 
 	now := time.Now()
 	nowNTP := rtptime.TimeToNTP(now)
+	jiffies := rtptime.TimeToJiffies(now)
 
 	for _, t := range conn.tracks {
 		clockrate := t.track.Codec().ClockRate
@@ -661,6 +662,8 @@ func sendSR(conn *rtpDownConnection) error {
 				PacketCount: p,
 				OctetCount:  b,
 			})
+		atomic.StoreUint64(&t.srTime, jiffies)
+		atomic.StoreUint64(&t.srNTPTime, nowNTP)
 	}
 
 	return conn.pc.WriteRTCP(packets)
@@ -876,6 +879,7 @@ func rtcpDownListener(conn *rtpDownConnection, track *rtpDownTrack, s *webrtc.RT
 			}
 			return
 		}
+		jiffies := rtptime.Jiffies()
 
 		for _, p := range ps {
 			switch p := p.(type) {
@@ -918,25 +922,21 @@ func rtcpDownListener(conn *rtpDownConnection, track *rtpDownTrack, s *webrtc.RT
 					log.Printf("sendFIR: %v", err)
 				}
 			case *rtcp.ReceiverEstimatedMaximumBitrate:
-				track.maxREMBBitrate.Set(
-					p.Bitrate, rtptime.Jiffies(),
-				)
+				track.maxREMBBitrate.Set(p.Bitrate, jiffies)
 			case *rtcp.ReceiverReport:
 				for _, r := range p.Reports {
 					if r.SSRC == track.track.SSRC() {
-						handleReport(track, r)
+						handleReport(track, r, jiffies)
 					}
 				}
 			case *rtcp.SenderReport:
 				for _, r := range p.Reports {
 					if r.SSRC == track.track.SSRC() {
-						handleReport(track, r)
+						handleReport(track, r, jiffies)
 					}
 				}
 			case *rtcp.TransportLayerNack:
-				maxBitrate := track.GetMaxBitrate(
-					rtptime.Jiffies(),
-				)
+				maxBitrate := track.GetMaxBitrate(jiffies)
 				bitrate := track.rate.Estimate()
 				if uint64(bitrate)*7/8 < maxBitrate {
 					sendRecovery(p, track)
@@ -946,10 +946,32 @@ func rtcpDownListener(conn *rtpDownConnection, track *rtpDownTrack, s *webrtc.RT
 	}
 }
 
-func handleReport(track *rtpDownTrack, report rtcp.ReceptionReport) {
-	jiffies := rtptime.Jiffies()
+func handleReport(track *rtpDownTrack, report rtcp.ReceptionReport, jiffies uint64) {
 	track.stats.Set(report.FractionLost, report.Jitter, jiffies)
 	track.updateRate(report.FractionLost, jiffies)
+
+	if report.LastSenderReport != 0 {
+		jiffies := rtptime.Jiffies()
+		srTime := atomic.LoadUint64(&track.srTime)
+		if jiffies < srTime || jiffies-srTime > 8*rtptime.JiffiesPerSec {
+			return
+		}
+		srNTPTime := atomic.LoadUint64(&track.srNTPTime)
+		if report.LastSenderReport == uint32(srNTPTime>>16) {
+			delay := uint64(report.Delay) *
+				(rtptime.JiffiesPerSec / 0x10000)
+			if delay > jiffies-srTime {
+				return
+			}
+			rtt := (jiffies - srTime) - delay
+			oldrtt := atomic.LoadUint64(&track.rtt)
+			newrtt := rtt
+			if oldrtt > 0 {
+				newrtt = (3*oldrtt + rtt) / 4
+			}
+			atomic.StoreUint64(&track.rtt, newrtt)
+		}
+	}
 }
 
 func trackKinds(down *rtpDownConnection) (audio bool, video bool) {

webserver.go

@@ -154,11 +154,14 @@ func statsHandler(w http.ResponseWriter, r *http.Request) {
 		fmt.Fprintf(w, "<td>%d%%</td>",
 			t.loss,
 		)
-		if t.jitter > 0 {
-			fmt.Fprintf(w, "<td>%v</td>", t.jitter)
-		} else {
-			fmt.Fprintf(w, "<td></td>")
+		fmt.Fprintf(w, "<td>")
+		if t.rtt > 0 {
+			fmt.Fprintf(w, "%v", t.rtt)
 		}
+		if t.jitter > 0 {
+			fmt.Fprintf(w, "&#177;%v", t.jitter)
+		}
+		fmt.Fprintf(w, "</td>")
 		fmt.Fprintf(w, "</tr>")
 	}