Simulcast.

README.PROTOCOL

@@ -135,8 +135,17 @@ A peer must explicitly request the streams that it wants to receive.
 ```
 
 The field `request` is a dictionary that maps the labels of requested
-streams to a list containing either 'audio', 'video' or both.  An entry
-with an empty key `''` serves as default.
+streams to a list containing either 'audio', or one of 'video' or
+'video-low'.  The empty key `''` serves as default.  For example:
+
+```javascript
+{
+    type: 'request',
+    request: {
+        camera: ['audio', 'video-low'],
+        '': ['audio', 'video']
+    }
+}
 
 ## Pushing streams
 
@@ -157,16 +166,22 @@ A stream is created by the sender with the `offer` message:
 If a stream with the same id exists, then this is a renegotation;
 otherwise this message creates a new stream.  If the field `replace` is
 not empty, then this request additionally requests that an existing stream
-with the given id should be closed, and the new stream should replace it.
+with the given id should be closed, and the new stream should replace it;
+this is used most notably when changing the simulcast envelope.
 
-The field `label` is one of `camera`, `screenshare` or `video`, as in the
-`request` message.
+The field `label` is one of `camera`, `screenshare` or `video`, and will
+be matched against the keys sent by the receiver in their `request` message.
 
 The field `sdp` contains the raw SDP string (i.e. the `sdp` field of
 a JSEP session description).  Galène will interpret the `nack`,
 `nack pli`, `ccm fir` and `goog-remb` RTCP feedback types, and act
 accordingly.
 
+The sender may either send a single stream per media section in the SDP,
+or use rid-based simulcasting.  In the latter case, it should send two
+video streams, one with rid 'h' and high throughput, and one with rid 'l'
+and throughput limited to roughly 100kbit/s.
+
 The receiver may either abort the stream immediately (see below), or send
 an answer.
 

conn/conn.go

@@ -25,6 +25,7 @@ type UpTrack interface {
 	AddLocal(DownTrack) error
 	DelLocal(DownTrack) bool
 	Kind() webrtc.RTPCodecType
+	Label() string
 	Codec() webrtc.RTPCodecCapability
 	// get a recent packet.  Returns 0 if the packet is not in cache.
 	GetRTP(seqno uint16, result []byte) uint16
@@ -33,7 +34,6 @@ type UpTrack interface {
 
 // Type Down represents a connection in the server to client direction.
 type Down interface {
-	GetMaxBitrate(now uint64) uint64
 }
 
 // Type DownTrack represents a track in the server to client direction.
@@ -42,4 +42,5 @@ type DownTrack interface {
 	Accumulate(bytes uint32)
 	SetTimeOffset(ntp uint64, rtp uint32)
 	SetCname(string)
+	GetMaxBitrate() uint64
 }

diskwriter/diskwriter.go

@@ -600,7 +600,7 @@ func (conn *diskConn) initWriter(width, height uint32) error {
 	return nil
 }
 
-func (down *diskConn) GetMaxBitrate(now uint64) uint64 {
+func (t *diskTrack) GetMaxBitrate() uint64 {
 	return ^uint64(0)
 }
 

group/group.go

@@ -13,6 +13,7 @@ import (
 	"time"
 
 	"github.com/pion/ice/v2"
+	"github.com/pion/sdp/v3"
 	"github.com/pion/webrtc/v3"
 )
 
@@ -60,7 +61,8 @@ type ChatHistoryEntry struct {
 }
 
 const (
-	MinBitrate = 200000
+	LowBitrate = 100000
+	MinBitrate = 2 * LowBitrate
 )
 
 type Group struct {
@@ -252,6 +254,12 @@ func APIFromCodecs(codecs []webrtc.RTPCodecCapability) (*webrtc.API, error) {
 	if UDPMin > 0 && UDPMax > 0 {
 		s.SetEphemeralUDPPortRange(UDPMin, UDPMax)
 	}
+	m.RegisterHeaderExtension(
+		webrtc.RTPHeaderExtensionCapability{sdp.SDESMidURI},
+		webrtc.RTPCodecTypeVideo)
+	m.RegisterHeaderExtension(
+		webrtc.RTPHeaderExtensionCapability{sdp.SDESRTPStreamIDURI},
+		webrtc.RTPCodecTypeVideo)
 
 	return webrtc.NewAPI(
 		webrtc.WithSettingEngine(s),

rtpconn/rtpconn.go

@@ -82,6 +82,7 @@ type rtpDownTrack struct {
 	remote         conn.UpTrack
 	ssrc           webrtc.SSRC
 	maxBitrate     *bitrate
+	maxREMBBitrate *bitrate
 	rate           *estimator.Estimator
 	stats          *receiverStats
 	atomics        *downTrackAtomics
@@ -140,7 +141,6 @@ type rtpDownConnection struct {
 	id                string
 	pc                *webrtc.PeerConnection
 	remote            conn.Up
-	maxREMBBitrate    *bitrate
 	iceCandidates     []*webrtc.ICECandidateInit
 	negotiationNeeded int
 
@@ -174,31 +174,22 @@ func newDownConn(c group.Client, id string, remote conn.Up) (*rtpDownConnection,
 		id:             id,
 		pc:             pc,
 		remote:         remote,
-		maxREMBBitrate: new(bitrate),
 	}
 
 	return conn, nil
 }
 
-func (down *rtpDownConnection) GetMaxBitrate(now uint64) uint64 {
-	rate := down.maxREMBBitrate.Get(now)
-	var trackRate uint64
-	tracks := down.getTracks()
-	for _, t := range tracks {
+func (t *rtpDownTrack) GetMaxBitrate() uint64 {
+	now := rtptime.Jiffies()
 	r := t.maxBitrate.Get(now)
 	if r == ^uint64(0) {
-			if t.track.Kind() == webrtc.RTPCodecTypeAudio {
-				r = 128 * 1024
-			} else {
 		r = 512 * 1024
 	}
+	rr := t.maxREMBBitrate.Get(now)
+	if rr == 0 || r < rr {
+		return r
 	}
-		trackRate += r
-	}
-	if trackRate < rate {
-		return trackRate
-	}
-	return rate
+	return rr
 }
 
 func (down *rtpDownConnection) addICECandidate(candidate *webrtc.ICECandidateInit) error {
@@ -311,6 +302,10 @@ func (up *rtpUpTrack) GetRTP(seqno uint16, result []byte) uint16 {
 	return up.cache.Get(seqno, result)
 }
 
+func (up *rtpUpTrack) Label() string {
+	return up.track.RID()
+}
+
 func (up *rtpUpTrack) Kind() webrtc.RTPCodecType {
 	return up.track.Kind()
 }
@@ -687,7 +682,7 @@ func rtcpUpListener(conn *rtpUpConnection, track *rtpUpTrack, r *webrtc.RTPRecei
 
 	for {
 		firstSR := false
-		n, _, err := r.Read(buf)
+		n, _, err := r.ReadSimulcast(buf, track.track.RID())
 		if err != nil {
 			if err != io.EOF && err != io.ErrClosedPipe {
 				log.Printf("Read RTCP: %v", err)
@@ -752,11 +747,11 @@ func rtcpUpListener(conn *rtpUpConnection, track *rtpUpTrack, r *webrtc.RTPRecei
 	}
 }
 
-func sendUpRTCP(conn *rtpUpConnection) error {
-	tracks := conn.getTracks()
+func sendUpRTCP(up *rtpUpConnection) error {
+	tracks := up.getTracks()
 
-	if len(conn.tracks) == 0 {
-		state := conn.pc.ConnectionState()
+	if len(up.tracks) == 0 {
+		state := up.pc.ConnectionState()
 		if state == webrtc.PeerConnectionStateClosed {
 			return io.ErrClosedPipe
 		}
@@ -765,7 +760,7 @@ func sendUpRTCP(conn *rtpUpConnection) error {
 
 	now := rtptime.Jiffies()
 
-	reports := make([]rtcp.ReceptionReport, 0, len(conn.tracks))
+	reports := make([]rtcp.ReceptionReport, 0, len(up.tracks))
 	for _, t := range tracks {
 		updateUpTrack(t)
 		stats := t.cache.GetStats(true)
@@ -810,29 +805,38 @@ func sendUpRTCP(conn *rtpUpConnection) error {
 		},
 	}
 
-	rate := ^uint64(0)
-
-	local := conn.getLocal()
-	for _, l := range local {
-		r := l.GetMaxBitrate(now)
-		if r < rate {
-			rate = r
-		}
-	}
-
-	if rate < group.MinBitrate {
-		rate = group.MinBitrate
-	}
-
 	var ssrcs []uint32
+	var rate uint64
 	for _, t := range tracks {
 		if !t.hasRtcpFb("goog-remb", "") {
 			continue
 		}
 		ssrcs = append(ssrcs, uint32(t.track.SSRC()))
+		var r uint64
+		if t.Kind() == webrtc.RTPCodecTypeAudio {
+			r = 100 * 1024
+		} else if t.Label() == "l" {
+			r = group.LowBitrate
+		} else {
+			local := t.getLocal()
+			r = ^uint64(0)
+			for _, down := range local {
+				rr := down.GetMaxBitrate()
+				if rr < group.MinBitrate {
+					rr = group.MinBitrate
+				}
+				if r > rr {
+					r = rr
+				}
+			}
+			if r == ^uint64(0) {
+				r = 512 * 1024
+			}
+		}
+		rate += r
 	}
 
-	if len(ssrcs) > 0 {
+	if rate < ^uint64(0) && len(ssrcs) > 0 {
 		packets = append(packets,
 			&rtcp.ReceiverEstimatedMaximumBitrate{
 				Bitrate: rate,
@@ -840,7 +844,7 @@ func sendUpRTCP(conn *rtpUpConnection) error {
 			},
 		)
 	}
-	return conn.pc.WriteRTCP(packets)
+	return up.pc.WriteRTCP(packets)
 }
 
 func rtcpUpSender(conn *rtpUpConnection) {
@@ -1049,7 +1053,7 @@ func rtcpDownListener(conn *rtpDownConnection, track *rtpDownTrack, s *webrtc.RT
 					log.Printf("sendFIR: %v", err)
 				}
 			case *rtcp.ReceiverEstimatedMaximumBitrate:
-				conn.maxREMBBitrate.Set(p.Bitrate, jiffies)
+				track.maxREMBBitrate.Set(p.Bitrate, jiffies)
 			case *rtcp.ReceiverReport:
 				for _, r := range p.Reports {
 					if r.SSRC == uint32(track.ssrc) {

rtpconn/rtpreader.go

@@ -149,6 +149,16 @@ func readLoop(conn *rtpUpConnection, track *rtpUpTrack) {
 
 		kf, _ := isKeyframe(codec.MimeType, &packet)
 
+		if packet.Extension {
+			packet.Extension = false
+			packet.Extensions = nil
+			bytes, err = packet.MarshalTo(buf)
+			if err != nil {
+				log.Printf("%v", err)
+				continue
+			}
+		}
+
 		first, index := track.cache.Store(
 			packet.SequenceNumber, packet.Timestamp,
 			kf, packet.Marker, buf[:bytes],

rtpconn/rtpstats.go

@@ -47,7 +47,6 @@ func (c *webClient) GetStats() *stats.Client {
 	for _, down := range c.down {
 		conns := stats.Conn{
 			Id:         down.id,
-			MaxBitrate: down.GetMaxBitrate(jiffies),
 		}
 		for _, t := range down.tracks {
 			rate, _ := t.rate.Estimate()

rtpconn/webclient.go

@@ -385,6 +385,7 @@ func addDownTrackUnlocked(conn *rtpDownConnection, remoteTrack *rtpUpTrack, remo
 		ssrc:           parms.Encodings[0].SSRC,
 		remote:         remoteTrack,
 		maxBitrate:     new(bitrate),
+		maxREMBBitrate: new(bitrate),
 		stats:          new(receiverStats),
 		rate:           estimator.New(time.Second),
 		atomics:        &downTrackAtomics{},
@@ -646,33 +647,60 @@ func requestedTracks(c *webClient, up conn.Up, tracks []conn.UpTrack) []conn.UpT
 		return nil
 	}
 
-	var audio, video bool
+	var audio, video, videoLow bool
 	for _, s := range r {
 		switch s {
 		case "audio":
 			audio = true
 		case "video":
 			video = true
+		case "video-low":
+			videoLow = true
 		default:
 			log.Printf("client requested unknown value %v", s)
 		}
 	}
 
+	find := func(kind webrtc.RTPCodecType, labels ...string) conn.UpTrack {
+		for _, l := range labels {
+			for _, t := range tracks {
+				if t.Kind() != kind {
+					continue
+				}
+				if t.Label() == l {
+					return t
+				}
+			}
+		}
+		for _, t := range tracks {
+			if t.Kind() != kind {
+				continue
+			}
+			return t
+		}
+		return nil
+	}
+
 	var ts []conn.UpTrack
 	if audio {
-		for _, t := range tracks {
-			if t.Kind() == webrtc.RTPCodecTypeAudio {
+		t := find(webrtc.RTPCodecTypeAudio)
+		if t != nil {
 			ts = append(ts, t)
-				break
-			}
 		}
 	}
 	if video {
-		for _, t := range tracks {
-			if t.Kind() == webrtc.RTPCodecTypeVideo {
+		t := find(
+			webrtc.RTPCodecTypeVideo, "h", "m", "video",
+		)
+		if t != nil {
 			ts = append(ts, t)
-				break
 		}
+	} else if videoLow {
+		t := find(
+			webrtc.RTPCodecTypeVideo, "l", "m", "video",
+		)
+		if t != nil {
+			ts = append(ts, t)
 		}
 	}
 

static/galene.html

@@ -213,7 +213,9 @@
             <select id="requestselect" class="select select-inline">
               <option value="">nothing</option>
               <option value="audio">audio only</option>
+              <option value="screenshare-low">screen share (low)</option>
               <option value="screenshare">screen share</option>
+              <option value="everything-low">everything (low)</option>
               <option value="everything" selected>everything</option>
             </select>
           </form>

static/galene.js

@@ -78,6 +78,7 @@ function getUserPass() {
  * @property {boolean} [localMute]
  * @property {string} [video]
  * @property {string} [audio]
+ * @property {boolean} [simulcast]
  * @property {string} [send]
  * @property {string} [request]
  * @property {boolean} [activityDetection]
@@ -550,9 +551,15 @@ function mapRequest(what) {
     case 'audio':
         return {'': ['audio']};
         break;
+    case 'screenshare-low':
+        return {screenshare: ['audio','video-low'], '': ['audio']};
+        break;
     case 'screenshare':
         return {screenshare: ['audio','video'], '': ['audio']};
         break;
+    case 'everything-low':
+        return {'': ['audio','video-low']};
+        break;
     case 'everything':
         return {'': ['audio','video']}
         break;
@@ -611,20 +618,25 @@ getInputElement('fileinput').onchange = function(e) {
 function gotUpStats(stats) {
     let c = this;
 
-    let text = '';
+    let values = [];
 
-    c.pc.getSenders().forEach(s => {
-        let tid = s.track && s.track.id;
-        let stats = tid && c.stats[tid];
-        let rate = stats && stats['outbound-rtp'] && stats['outbound-rtp'].rate;
+    for(let id in stats) {
+        if(stats[id] && stats[id]['outbound-rtp']) {
+            let rate = stats[id]['outbound-rtp'].rate;
             if(typeof rate === 'number') {
-            if(text)
-                text = text + ' + ';
-            text = text + Math.round(rate / 1000) + 'kbps';
+                values.push(rate);
+            }
+        }
     }
-    });
 
-    setLabel(c, text);
+    if(values.length === 0) {
+        setLabel(c, '');
+    } else {
+        values.sort((x,y) => x - y);
+        setLabel(c, values
+                 .map(x => Math.round(x / 1000).toString())
+                 .reduce((x, y) => x + '+' + y));
+    }
 }
 
 /**
@@ -800,6 +812,7 @@ function newUpStream(localId) {
  * @param {number} [bps]
  */
 async function setMaxVideoThroughput(c, bps) {
+    let simulcast = doSimulcast();
     let senders = c.pc.getSenders();
     for(let i = 0; i < senders.length; i++) {
         let s = senders[i];
@@ -808,17 +821,17 @@ async function setMaxVideoThroughput(c, bps) {
         let p = s.getParameters();
         if(!p.encodings)
             p.encodings = [{}];
+        if((!simulcast && p.encodings.length != 1) ||
+           (simulcast && p.encodings.length != 2)) {
+            // change the simulcast envelope
+            await replaceUpStream(c);
+            return;
+        }
         p.encodings.forEach(e => {
-            if(bps > 0)
-                e.maxBitrate = bps;
-            else
-                delete e.maxBitrate;
+            if(!e.rid || e.rid === 'h')
+                e.maxBitrate = bps || unlimitedRate;
         });
-        try {
         await s.setParameters(p);
-        } catch(e) {
-            console.error(e);
-        }
     }
 }
 
@@ -1022,6 +1035,19 @@ function isSafari() {
     return ua.indexOf('safari') >= 0 && ua.indexOf('chrome') < 0;
 }
 
+const unlimitedRate = 1000000000;
+const simulcastRate = 100000;
+
+/**
+ * @returns {boolean}
+ */
+function doSimulcast() {
+    if(!getSettings().simulcast)
+        return false;
+    let bps = getMaxVideoThroughput();
+    return bps <= 0 || bps >= 2 * simulcastRate;
+}
+
 /**
  * Sets up c to send the given stream.  Some extra parameters are stored
  * in c.userdata.
@@ -1029,6 +1055,7 @@ function isSafari() {
  * @param {Stream} c
  * @param {MediaStream} stream
  */
+
 function setUpStream(c, stream) {
     if(c.stream != null)
         throw new Error("Setting nonempty stream");
@@ -1073,11 +1100,20 @@ function setUpStream(c, stream) {
             c.close();
         };
 
-        let encodings = [{}];
+        let encodings = [];
         if(t.kind === 'video') {
+            let simulcast = doSimulcast();
             let bps = getMaxVideoThroughput();
-            if(bps > 0)
-                encodings[0].maxBitrate = bps;
+            encodings.push({
+                rid: 'h',
+                maxBitrate: bps || unlimitedRate,
+            });
+            if(simulcast)
+                encodings.push({
+                    rid: 'l',
+                    scaleResolutionDownBy: 2,
+                    maxBitrate: simulcastRate,
+                });
         }
         c.pc.addTransceiver(t, {
             direction: 'sendonly',

static/protocol.js

@@ -1246,17 +1246,20 @@ Stream.prototype.updateStats = async function() {
         if(report) {
             for(let r of report.values()) {
                 if(stid && r.type === 'outbound-rtp') {
+                    let id = stid;
+                    if(r.rid)
+                        id = id + '-' + r.rid
                     if(!('bytesSent' in r))
                         continue;
-                    if(!stats[stid])
-                        stats[stid] = {};
-                    stats[stid][r.type] = {};
-                    stats[stid][r.type].timestamp = r.timestamp;
-                    stats[stid][r.type].bytesSent = r.bytesSent;
-                    if(old[stid] && old[stid][r.type])
-                        stats[stid][r.type].rate =
-                        ((r.bytesSent - old[stid][r.type].bytesSent) * 1000 /
-                         (r.timestamp - old[stid][r.type].timestamp)) * 8;
+                    if(!stats[id])
+                        stats[id] = {};
+                    stats[id][r.type] = {};
+                    stats[id][r.type].timestamp = r.timestamp;
+                    stats[id][r.type].bytesSent = r.bytesSent;
+                    if(old[id] && old[id][r.type])
+                        stats[id][r.type].rate =
+                        ((r.bytesSent - old[id][r.type].bytesSent) * 1000 /
+                         (r.timestamp - old[id][r.type].timestamp)) * 8;
                 }
             }
         }