feat: add packet_buffer_init_video config parameter

pkg/config/config.go

@@ -89,8 +89,15 @@ type RTCConfig struct {
 
 	// Deprecated: use PacketBufferSizeVideo and PacketBufferSizeAudio
 	PacketBufferSize int `yaml:"packet_buffer_size,omitempty"`
-	// Number of packets to buffer for NACK - video
+	// Number of packets to buffer for NACK - video (max capacity)
 	PacketBufferSizeVideo int `yaml:"packet_buffer_size_video,omitempty"`
+	// Initial bucket size for video packets. If set, the bucket starts at
+	// this size instead of InitPacketBufferSizeVideo (300). Useful for
+	// high-PPS H.264 streams under network delay where burst patterns
+	// exceed the default initial size but the PPS-based growth logic
+	// doesn't detect the need because it uses average PPS, not peak burst.
+	// If not set or zero, falls back to InitPacketBufferSizeVideo.
+	PacketBufferInitVideo int `yaml:"packet_buffer_init_video,omitempty"`
 	// Number of packets to buffer for NACK - audio
 	PacketBufferSizeAudio int `yaml:"packet_buffer_size_audio,omitempty"`
 

pkg/rtc/config.go

@@ -40,6 +40,7 @@ type WebRTCConfig struct {
 
 type ReceiverConfig struct {
 	PacketBufferSizeVideo int
+	PacketBufferInitVideo int // Initial bucket size (0 = use InitPacketBufferSizeVideo)
 	PacketBufferSizeAudio int
 }
 
@@ -117,6 +118,7 @@ func NewWebRTCConfig(conf *config.Config) (*WebRTCConfig, error) {
 		WebRTCConfig: *webRTCConfig,
 		Receiver: ReceiverConfig{
 			PacketBufferSizeVideo: rtcConf.PacketBufferSizeVideo,
+			PacketBufferInitVideo: rtcConf.PacketBufferInitVideo,
 			PacketBufferSizeAudio: rtcConf.PacketBufferSizeAudio,
 		},
 		Publisher:  publisherConfig,

pkg/rtc/room.go

@@ -264,7 +264,7 @@ func NewRoom(
 		participantRequestSources:            make(map[livekit.ParticipantIdentity]routing.MessageSource),
 		hasPublished:                         make(map[livekit.ParticipantIdentity]bool),
 		agentParticpants:                     make(map[livekit.ParticipantIdentity]*agentJob),
-		bufferFactory:                        buffer.NewFactoryOfBufferFactory(config.Receiver.PacketBufferSizeVideo, config.Receiver.PacketBufferSizeAudio),
+		bufferFactory:                        buffer.NewFactoryOfBufferFactory(config.Receiver.PacketBufferSizeVideo, config.Receiver.PacketBufferInitVideo, config.Receiver.PacketBufferSizeAudio),
 		batchedUpdates:                       make(map[livekit.ParticipantIdentity]*participantUpdate),
 		closed:                               make(chan struct{}),
 		trailer:                              []byte(utils.RandomSecret()),

pkg/sfu/buffer/buffer.go

@@ -77,6 +77,7 @@ type Buffer struct {
 	bucket          *bucket.Bucket[uint64]
 	nacker          *nack.NackQueue
 	maxVideoPkts    int
+	initVideoPkts   int
 	maxAudioPkts    int
 	codecType       webrtc.RTPCodecType
 	payloadType     uint8
@@ -142,12 +143,13 @@ type Buffer struct {
 }
 
 // NewBuffer constructs a new Buffer
-func NewBuffer(ssrc uint32, maxVideoPkts, maxAudioPkts int) *Buffer {
+func NewBuffer(ssrc uint32, maxVideoPkts, initVideoPkts, maxAudioPkts int) *Buffer {
 	l := logger.GetLogger() // will be reset with correct context via SetLogger
 	b := &Buffer{
-		mediaSSRC:    ssrc,
-		maxVideoPkts: maxVideoPkts,
-		maxAudioPkts: maxAudioPkts,
+		mediaSSRC:     ssrc,
+		maxVideoPkts:  maxVideoPkts,
+		initVideoPkts: initVideoPkts,
+		maxAudioPkts:  maxAudioPkts,
 		snRangeMap:   utils.NewRangeMap[uint64, uint64](100),
 		pliThrottle:  int64(500 * time.Millisecond),
 		logger:       l.WithComponent(sutils.ComponentPub).WithComponent(sutils.ComponentSFU),
@@ -260,7 +262,12 @@ func (b *Buffer) Bind(params webrtc.RTPParameters, codec webrtc.RTPCodecCapabili
 
 	case strings.HasPrefix(b.mime, "video/"):
 		b.codecType = webrtc.RTPCodecTypeVideo
-		b.bucket = bucket.NewBucket[uint64](InitPacketBufferSizeVideo)
+		// Use initVideoPkts from config if set, otherwise fall back to default.
+		initSize := InitPacketBufferSizeVideo
+		if b.initVideoPkts > 0 {
+			initSize = b.initVideoPkts
+		}
+		b.bucket = bucket.NewBucket[uint64](initSize)
 		if b.frameRateCalculator[0] == nil {
 			if strings.EqualFold(codec.MimeType, webrtc.MimeTypeVP8) {
 				b.frameRateCalculator[0] = NewFrameRateCalculatorVP8(b.clockRate, b.logger)

pkg/sfu/buffer/factory.go

@@ -23,19 +23,22 @@ import (
 
 type FactoryOfBufferFactory struct {
 	trackingPacketsVideo int
+	initPacketsVideo     int
 	trackingPacketsAudio int
 }
 
-func NewFactoryOfBufferFactory(trackingPacketsVideo int, trackingPacketsAudio int) *FactoryOfBufferFactory {
+func NewFactoryOfBufferFactory(trackingPacketsVideo int, initPacketsVideo int, trackingPacketsAudio int) *FactoryOfBufferFactory {
 	return &FactoryOfBufferFactory{
 		trackingPacketsVideo: trackingPacketsVideo,
+		initPacketsVideo:     initPacketsVideo,
 		trackingPacketsAudio: trackingPacketsAudio,
 	}
 }
 
 func (f *FactoryOfBufferFactory) CreateBufferFactory() *Factory {
 	return &Factory{
 		trackingPacketsVideo: f.trackingPacketsVideo,
+		initPacketsVideo:     f.initPacketsVideo,
 		trackingPacketsAudio: f.trackingPacketsAudio,
 		rtpBuffers:           make(map[uint32]*Buffer),
 		rtcpReaders:          make(map[uint32]*RTCPReader),
@@ -46,6 +49,7 @@ func (f *FactoryOfBufferFactory) CreateBufferFactory() *Factory {
 type Factory struct {
 	sync.RWMutex
 	trackingPacketsVideo int
+	initPacketsVideo     int
 	trackingPacketsAudio int
 	rtpBuffers           map[uint32]*Buffer
 	rtcpReaders          map[uint32]*RTCPReader
@@ -72,7 +76,7 @@ func (f *Factory) GetOrNew(packetType packetio.BufferPacketType, ssrc uint32) io
 		if reader, ok := f.rtpBuffers[ssrc]; ok {
 			return reader
 		}
-		buffer := NewBuffer(ssrc, f.trackingPacketsVideo, f.trackingPacketsAudio)
+		buffer := NewBuffer(ssrc, f.trackingPacketsVideo, f.initPacketsVideo, f.trackingPacketsAudio)
 		f.rtpBuffers[ssrc] = buffer
 		for repair, base := range f.rtxPair {
 			if repair == ssrc {