add udpReadBufferSize parameter (#5129)

this allows to set a global UDP read buffer, applied to every UDP socket.

api/openapi.yaml

@@ -77,6 +77,9 @@ components:
         udpMaxPayloadSize:
           type: integer
           format: int64
+        udpReadBufferSize:
+          type: integer
+          format: int64
         runOnConnect:
           type: string
         runOnConnectRestart:
@@ -225,9 +228,6 @@ components:
           type: array
           items:
             type: string
-        rtspUDPReadBufferSize:
-          type: integer
-          format: int64
 
         # RTMP server
         rtmp:
@@ -393,21 +393,10 @@ components:
           type: string
         rtspRangeStart:
           type: string
-        rtspUDPReadBufferSize:
-          type: integer
-          format: int64
-
-        # MPEG-TS source
-        mpegtsUDPReadBufferSize:
-          type: integer
-          format: int64
 
         # RTP source
         rtpSDP:
           type: string
-        rtpUDPReadBufferSize:
-          type: integer
-          format: int64
 
         # Redirect source
         sourceRedirect:

docs/2-usage/26-decrease-packet-loss.md

@@ -1,47 +1,20 @@
 # Decrease packet loss
 
-MediaMTX is meant for routing live streams, and makes use of a series of protocol which try to minimize latency at cost of losing packets in transmit. In particular, most protocols are built on UDP, which is an "unrealiable transport", specifically picked because it allows to drop late packets in case of network congestions, preserving the real-time aspect of streams.
+MediaMTX is meant for routing live streams, and makes use of a series of protocol which try to preserve the real-time aspect of streams and minimize latency at cost of losing packets in transmit. In particular, most protocols are built on UDP, which is an "unrealiable transport", specifically picked because it allows to drop late packets in case of network congestions.
 
 These packet losses are usually detected and printed in MediaMTX logs.
 
-If you need to improve the stream reliability and decrease packet losses, the first thing to do is to check whether the network between the MediaMTX instance and the intended publishers and readers has sufficient bandwidth for transmitting the media stream. Most of the times, packet losses are caused by a network which is not fit for this scope. This limitation can be overcome by either recompressing the stream with a lower bitrate, or by changing network.
+If you need to improve the stream reliability and decrease packet losses, the first thing to do is to check whether the network between the MediaMTX instance and the intended publishers and readers has sufficient bandwidth for transmitting the media stream. Most of the times, packet losses are caused by a network which is not fit for this scope. This limitation can be overcome by either recompressing the stream with a lower bitrate, or by upgrading the network physical infrastructure (routers, cables, Wi-Fi, firewalls, topology, etc).
 
 There are however some parameters that can be tuned to improve the situation, at cost of increasing RAM consumption:
 
-- When publishing a stream with RTSP, MPEG-TS or RTP, packets might get discarded by the server because the UDP read buffer size is too small. Try increasing the UDP read buffer size:
+- When publishing a stream with a UDP-based protocol (currently RTSP, MPEG-TS, RTP, SRT, WebRTC), packets might get discarded by the server because the read buffer size of UDP sockets is too small. It can be increased with this parameter:
 
   ```yml
-  rtspUDPReadBufferSize: 1000000
+  udpReadBufferSize: 1000000
   ```
 
-  If the source of the stream is a camera:
+  The `udpReadBufferSize` parameter requires the `net.core.rmem_max` system parameter to be equal or greater than it. It can be set with this command:
-
-  ```yml
-  paths:
-    test:
-      source: rtsp://..
-      rtspUDPReadBufferSize: 1000000
-  ```
-
-  There are similar options for the MPEG-TS and RTP protocol:
-
-  ```yml
-  paths:
-    test:
-      source: udp+mpegts://..
-      mpegtsUDPReadBufferSize: 1000000
-  ```
-
-  or:
-
-  ```yml
-  paths:
-    test:
-      source: udp+rtp://..
-      rtpUDPReadBufferSize: 1000000
-  ```
-
-  All these options require the `net.core.rmem_max` system parameter to be equal or greater than `rtspUDPReadBufferSize`:
 
   ```sh
   sudo sysctl net.core.rmem_max=100000000

internal/conf/conf.go

@@ -159,6 +159,7 @@ type Conf struct {
 	ReadBufferCount     *int            `json:"readBufferCount,omitempty"` // deprecated
 	WriteQueueSize      int             `json:"writeQueueSize"`
 	UDPMaxPayloadSize   int             `json:"udpMaxPayloadSize"`
+	UDPReadBufferSize   uint            `json:"udpReadBufferSize"`
 	RunOnConnect        string          `json:"runOnConnect"`
 	RunOnConnectRestart bool            `json:"runOnConnectRestart"`
 	RunOnDisconnect     string          `json:"runOnDisconnect"`
@@ -235,7 +236,7 @@ type Conf struct {
 	RTSPServerCert        string           `json:"rtspServerCert"`
 	AuthMethods           *RTSPAuthMethods `json:"authMethods,omitempty"` // deprecated
 	RTSPAuthMethods       RTSPAuthMethods  `json:"rtspAuthMethods"`
-	RTSPUDPReadBufferSize uint             `json:"rtspUDPReadBufferSize"`
+	RTSPUDPReadBufferSize *uint            `json:"rtspUDPReadBufferSize,omitempty"` // deprecated
 
 	// RTMP server
 	RTMP           bool       `json:"rtmp"`

internal/conf/path.go

@@ -152,14 +152,14 @@ type Path struct {
 	SourceAnyPortEnable   *bool          `json:"sourceAnyPortEnable,omitempty"` // deprecated
 	RTSPRangeType         RTSPRangeType  `json:"rtspRangeType"`
 	RTSPRangeStart        string         `json:"rtspRangeStart"`
-	RTSPUDPReadBufferSize uint           `json:"rtspUDPReadBufferSize"`
+	RTSPUDPReadBufferSize *uint          `json:"rtspUDPReadBufferSize,omitempty"` // deprecated
 
 	// MPEG-TS source
-	MPEGTSUDPReadBufferSize uint `json:"mpegtsUDPReadBufferSize"`
+	MPEGTSUDPReadBufferSize *uint `json:"mpegtsUDPReadBufferSize,omitempty"` // deprecated
 
 	// RTP source
 	RTPSDP               string `json:"rtpSDP"`
-	RTPUDPReadBufferSize uint   `json:"rtpUDPReadBufferSize"`
+	RTPUDPReadBufferSize *uint  `json:"rtpUDPReadBufferSize,omitempty"` // deprecated
 
 	// Redirect source
 	SourceRedirect string `json:"sourceRedirect"`

internal/core/core.go

@@ -399,6 +399,7 @@ func (p *Core) createResources(initial bool) error {
 			readTimeout:       p.conf.ReadTimeout,
 			writeTimeout:      p.conf.WriteTimeout,
 			writeQueueSize:    p.conf.WriteQueueSize,
+			udpReadBufferSize: p.conf.UDPReadBufferSize,
 			rtpMaxPayloadSize: rtpMaxPayloadSize,
 			pathConfs:         p.conf.Paths,
 			externalCmdPool:   p.externalCmdPool,
@@ -415,10 +416,15 @@ func (p *Core) createResources(initial bool) error {
 		_, useUDP := p.conf.RTSPTransports[gortsplib.ProtocolUDP]
 		_, useMulticast := p.conf.RTSPTransports[gortsplib.ProtocolUDPMulticast]
 
+		udpReadBufferSize := p.conf.UDPReadBufferSize
+		if p.conf.RTSPUDPReadBufferSize != nil {
+			udpReadBufferSize = *p.conf.RTSPUDPReadBufferSize
+		}
+
 		i := &rtsp.Server{
 			Address:             p.conf.RTSPAddress,
 			AuthMethods:         p.conf.RTSPAuthMethods,
-			UDPReadBufferSize:   p.conf.RTSPUDPReadBufferSize,
+			UDPReadBufferSize:   udpReadBufferSize,
 			ReadTimeout:         p.conf.ReadTimeout,
 			WriteTimeout:        p.conf.WriteTimeout,
 			WriteQueueSize:      p.conf.WriteQueueSize,
@@ -456,10 +462,15 @@ func (p *Core) createResources(initial bool) error {
 		_, useUDP := p.conf.RTSPTransports[gortsplib.ProtocolUDP]
 		_, useMulticast := p.conf.RTSPTransports[gortsplib.ProtocolUDPMulticast]
 
+		udpReadBufferSize := p.conf.UDPReadBufferSize
+		if p.conf.RTSPUDPReadBufferSize != nil {
+			udpReadBufferSize = *p.conf.RTSPUDPReadBufferSize
+		}
+
 		i := &rtsp.Server{
 			Address:             p.conf.RTSPSAddress,
 			AuthMethods:         p.conf.RTSPAuthMethods,
-			UDPReadBufferSize:   p.conf.RTSPUDPReadBufferSize,
+			UDPReadBufferSize:   udpReadBufferSize,
 			ReadTimeout:         p.conf.ReadTimeout,
 			WriteTimeout:        p.conf.WriteTimeout,
 			WriteQueueSize:      p.conf.WriteQueueSize,
@@ -749,6 +760,7 @@ func (p *Core) closeResources(newConf *conf.Conf, calledByAPI bool) {
 		newConf.ReadTimeout != p.conf.ReadTimeout ||
 		newConf.WriteTimeout != p.conf.WriteTimeout ||
 		newConf.WriteQueueSize != p.conf.WriteQueueSize ||
+		newConf.UDPReadBufferSize != p.conf.UDPReadBufferSize ||
 		newConf.UDPMaxPayloadSize != p.conf.UDPMaxPayloadSize ||
 		newConf.RTSPEncryption != p.conf.RTSPEncryption ||
 		closeMetrics ||
@@ -764,6 +776,7 @@ func (p *Core) closeResources(newConf *conf.Conf, calledByAPI bool) {
 		newConf.RTSPAddress != p.conf.RTSPAddress ||
 		!reflect.DeepEqual(newConf.RTSPAuthMethods, p.conf.RTSPAuthMethods) ||
 		newConf.RTSPUDPReadBufferSize != p.conf.RTSPUDPReadBufferSize ||
+		newConf.UDPReadBufferSize != p.conf.UDPReadBufferSize ||
 		newConf.ReadTimeout != p.conf.ReadTimeout ||
 		newConf.WriteTimeout != p.conf.WriteTimeout ||
 		newConf.WriteQueueSize != p.conf.WriteQueueSize ||
@@ -787,6 +800,7 @@ func (p *Core) closeResources(newConf *conf.Conf, calledByAPI bool) {
 		newConf.RTSPSAddress != p.conf.RTSPSAddress ||
 		!reflect.DeepEqual(newConf.RTSPAuthMethods, p.conf.RTSPAuthMethods) ||
 		newConf.RTSPUDPReadBufferSize != p.conf.RTSPUDPReadBufferSize ||
+		newConf.UDPReadBufferSize != p.conf.UDPReadBufferSize ||
 		newConf.ReadTimeout != p.conf.ReadTimeout ||
 		newConf.WriteTimeout != p.conf.WriteTimeout ||
 		newConf.WriteQueueSize != p.conf.WriteQueueSize ||

internal/core/path.go

@@ -70,6 +70,7 @@ type path struct {
 	readTimeout       conf.Duration
 	writeTimeout      conf.Duration
 	writeQueueSize    int
+	udpReadBufferSize uint
 	rtpMaxPayloadSize int
 	conf              *conf.Path
 	name              string
@@ -174,6 +175,7 @@ func (pa *path) run() {
 			ReadTimeout:       pa.readTimeout,
 			WriteTimeout:      pa.writeTimeout,
 			WriteQueueSize:    pa.writeQueueSize,
+			UDPReadBufferSize: pa.udpReadBufferSize,
 			RTPMaxPayloadSize: pa.rtpMaxPayloadSize,
 			Matches:           pa.matches,
 			PathManager:       pa.parent,

internal/core/path_manager.go

@@ -76,6 +76,7 @@ type pathManager struct {
 	readTimeout       conf.Duration
 	writeTimeout      conf.Duration
 	writeQueueSize    int
+	udpReadBufferSize uint
 	rtpMaxPayloadSize int
 	pathConfs         map[string]*conf.Path
 	externalCmdPool   *externalcmd.Pool
@@ -439,6 +440,7 @@ func (pm *pathManager) createPath(
 		readTimeout:       pm.readTimeout,
 		writeTimeout:      pm.writeTimeout,
 		writeQueueSize:    pm.writeQueueSize,
+		udpReadBufferSize: pm.udpReadBufferSize,
 		rtpMaxPayloadSize: pm.rtpMaxPayloadSize,
 		conf:              pathConf,
 		name:              name,

internal/staticsources/handler.go

@@ -67,6 +67,7 @@ type Handler struct {
 	ReadTimeout       conf.Duration
 	WriteTimeout      conf.Duration
 	WriteQueueSize    int
+	UDPReadBufferSize uint
 	RTPMaxPayloadSize int
 	Matches           []string
 	PathManager       handlerPathManager
@@ -101,10 +102,11 @@ func (s *Handler) Initialize() {
 		strings.HasPrefix(s.Conf.Source, "rtsp+ws://") ||
 		strings.HasPrefix(s.Conf.Source, "rtsps+ws://"):
 		s.instance = &ssrtsp.Source{
-			ReadTimeout:    s.ReadTimeout,
+			ReadTimeout:       s.ReadTimeout,
-			WriteTimeout:   s.WriteTimeout,
+			WriteTimeout:      s.WriteTimeout,
-			WriteQueueSize: s.WriteQueueSize,
+			WriteQueueSize:    s.WriteQueueSize,
-			Parent:         s,
+			UDPReadBufferSize: s.UDPReadBufferSize,
+			Parent:            s,
 		}
 
 	case strings.HasPrefix(s.Conf.Source, "rtmp://") ||
@@ -126,8 +128,9 @@ func (s *Handler) Initialize() {
 		strings.HasPrefix(s.Conf.Source, "udp+mpegts://") ||
 		strings.HasPrefix(s.Conf.Source, "unix+mpegts://"):
 		s.instance = &ssmpegts.Source{
-			ReadTimeout: s.ReadTimeout,
+			ReadTimeout:       s.ReadTimeout,
-			Parent:      s,
+			UDPReadBufferSize: s.UDPReadBufferSize,
+			Parent:            s,
 		}
 
 	case strings.HasPrefix(s.Conf.Source, "srt://"):
@@ -146,8 +149,9 @@ func (s *Handler) Initialize() {
 	case strings.HasPrefix(s.Conf.Source, "udp+rtp://") ||
 		strings.HasPrefix(s.Conf.Source, "unix+rtp://"):
 		s.instance = &ssrtp.Source{
-			ReadTimeout: s.ReadTimeout,
+			ReadTimeout:       s.ReadTimeout,
-			Parent:      s,
+			UDPReadBufferSize: s.UDPReadBufferSize,
+			Parent:            s,
 		}
 
 	case s.Conf.Source == "rpiCamera":

internal/staticsources/mpegts/source.go

@@ -27,8 +27,9 @@ type parent interface {
 
 // Source is a MPEG-TS static source.
 type Source struct {
-	ReadTimeout conf.Duration
+	ReadTimeout       conf.Duration
-	Parent      parent
+	UDPReadBufferSize uint
+	Parent            parent
 }
 
 // Log implements logger.Writer.
@@ -55,7 +56,12 @@ func (s *Source) Run(params defs.StaticSourceRunParams) error {
 		}
 
 	default:
-		nc, err = udp.CreateConn(u, int(params.Conf.MPEGTSUDPReadBufferSize))
+		udpReadBufferSize := s.UDPReadBufferSize
+		if params.Conf.MPEGTSUDPReadBufferSize != nil {
+			udpReadBufferSize = *params.Conf.MPEGTSUDPReadBufferSize
+		}
+
+		nc, err = udp.CreateConn(u, int(udpReadBufferSize))
 		if err != nil {
 			return err
 		}

internal/staticsources/rtp/source.go

@@ -29,8 +29,9 @@ type parent interface {
 
 // Source is a RTP static source.
 type Source struct {
-	ReadTimeout conf.Duration
+	ReadTimeout       conf.Duration
-	Parent      parent
+	UDPReadBufferSize uint
+	Parent            parent
 }
 
 // Log implements logger.Writer.
@@ -69,7 +70,12 @@ func (s *Source) Run(params defs.StaticSourceRunParams) error {
 		}
 
 	default:
-		nc, err = udp.CreateConn(u, int(params.Conf.RTPUDPReadBufferSize))
+		udpReadBufferSize := s.UDPReadBufferSize
+		if params.Conf.RTPUDPReadBufferSize != nil {
+			udpReadBufferSize = *params.Conf.RTPUDPReadBufferSize
+		}
+
+		nc, err = udp.CreateConn(u, int(udpReadBufferSize))
 		if err != nil {
 			return err
 		}

internal/staticsources/rtsp/source.go

@@ -71,10 +71,11 @@ type parent interface {
 
 // Source is a RTSP static source.
 type Source struct {
-	ReadTimeout    conf.Duration
+	ReadTimeout       conf.Duration
-	WriteTimeout   conf.Duration
+	WriteTimeout      conf.Duration
-	WriteQueueSize int
+	WriteQueueSize    int
-	Parent         parent
+	UDPReadBufferSize uint
+	Parent            parent
 }
 
 // Log implements logger.Writer.
@@ -145,6 +146,11 @@ func (s *Source) Run(params defs.StaticSourceRunParams) error {
 		return err
 	}
 
+	udpReadBufferSize := s.UDPReadBufferSize
+	if params.Conf.RTSPUDPReadBufferSize != nil {
+		udpReadBufferSize = *params.Conf.RTSPUDPReadBufferSize
+	}
+
 	c := &gortsplib.Client{
 		Scheme:            scheme,
 		Host:              u.Host,
@@ -154,7 +160,7 @@ func (s *Source) Run(params defs.StaticSourceRunParams) error {
 		ReadTimeout:       time.Duration(s.ReadTimeout),
 		WriteTimeout:      time.Duration(s.WriteTimeout),
 		WriteQueueSize:    s.WriteQueueSize,
-		UDPReadBufferSize: int(params.Conf.RTSPUDPReadBufferSize),
+		UDPReadBufferSize: int(udpReadBufferSize),
 		AnyPortEnable:     params.Conf.RTSPAnyPort,
 		OnRequest: func(req *base.Request) {
 			s.Log(logger.Debug, "[c->s] %v", req)

mediamtx.yml

@@ -25,6 +25,10 @@ writeQueueSize: 512
 # Maximum size of outgoing UDP packets.
 # This can be decreased to avoid fragmentation on networks with a low UDP MTU.
 udpMaxPayloadSize: 1472
+# Size of the read buffer of every UDP socket.
+# This can be increased to decrease packet losses.
+# It defaults to the default value of the operating system.
+udpReadBufferSize: 0
 
 # Command to run when a client connects to the server.
 # This is terminated with SIGINT when a client disconnects from the server.
@@ -277,10 +281,6 @@ rtspServerCert: server.crt
 # Authentication methods. Available are "basic" and "digest".
 # "digest" doesn't provide any additional security and is available for compatibility only.
 rtspAuthMethods: [basic]
-# Size of the UDP buffer of the RTSP server.
-# This can be increased to mitigate packet losses.
-# It defaults to the default value of the operating system.
-rtspUDPReadBufferSize: 0
 
 ###############################################
 # Global settings -> RTMP server
@@ -538,28 +538,12 @@ pathDefaults:
   # * npt: duration such as "300ms", "1.5m" or "2h45m", valid time units are "ns", "us" (or "µs"), "ms", "s", "m", "h"
   # * smpte: duration such as "300ms", "1.5m" or "2h45m", valid time units are "ns", "us" (or "µs"), "ms", "s", "m", "h"
   rtspRangeStart:
-  # Size of the UDP buffer of the RTSP client.
-  # This can be increased to mitigate packet losses.
-  # It defaults to the default value of the operating system.
-  rtspUDPReadBufferSize: 0
-
-  ###############################################
-  # Default path settings -> MPEG-TS source (when source is MPEG-TS)
-
-  # Size of the UDP buffer of the MPEG-TS client.
-  # This can be increased to mitigate packet losses.
-  # It defaults to the default value of the operating system.
-  mpegtsUDPReadBufferSize: 0
 
   ###############################################
   # Default path settings -> RTP source (when source is RTP)
 
   # session description protocol (SDP) of the RTP stream.
   rtpSDP:
-  # Size of the UDP buffer of the RTP client.
-  # This can be increased to mitigate packet losses.
-  # It defaults to the default value of the operating system.
-  rtpUDPReadBufferSize: 0
 
   ###############################################
   # Default path settings -> Redirect source (when source is "redirect")