External/mediamtx
1
0
Fork 1
mirror of https://github.com/bluenviron/mediamtx.git synced 2025-12-20 02:00:05 -08:00
Code Issues Projects Releases Packages Wiki Activity Actions 2

add udpReadBufferSize parameter (#5129)

Browse source 
this allows to set a global UDP read buffer, applied to every UDP socket.
This commit is contained in:
Alessandro Ros 2025-10-29 11:28:22 +01:00 • committed by GitHub
parent 3ae5262510
commit adc4a6ceb6
No known key found for this signature in database
GPG key ID: B5690EEEBB952194
12 changed files with 78 additions and 91 deletions
Download patch file Download diff file Expand all files Collapse all files

17
api/openapi.yaml
View file

@ -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:

37
docs/2-usage/26-decrease-packet-loss.md
View file

@ -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:
```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`:
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:
```sh
sudo sysctl net.core.rmem_max=100000000

3
internal/conf/conf.go
View file

@ -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"`

6
internal/conf/path.go
View file

@ -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"`

18
internal/core/core.go
View file

@ -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 ||

2
internal/core/path.go
View file

@ -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,

2
internal/core/path_manager.go
View file

@ -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,

20
internal/staticsources/handler.go
View file

@ -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,
WriteTimeout: s.WriteTimeout,
WriteQueueSize: s.WriteQueueSize,
Parent: s,
ReadTimeout: s.ReadTimeout,
WriteTimeout: s.WriteTimeout,
WriteQueueSize: s.WriteQueueSize,
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,
Parent: s,
ReadTimeout: s.ReadTimeout,
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,
Parent: s,
ReadTimeout: s.ReadTimeout,
UDPReadBufferSize: s.UDPReadBufferSize,
Parent: s,
}
case s.Conf.Source == "rpiCamera":

12
internal/staticsources/mpegts/source.go
View file

@ -27,8 +27,9 @@ type parent interface {
// Source is a MPEG-TS static source.
type Source struct {
ReadTimeout conf.Duration
Parent parent
ReadTimeout conf.Duration
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
}

12
internal/staticsources/rtp/source.go
View file

@ -29,8 +29,9 @@ type parent interface {
// Source is a RTP static source.
type Source struct {
ReadTimeout conf.Duration
Parent parent
ReadTimeout conf.Duration
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
}

16
internal/staticsources/rtsp/source.go
View file

@ -71,10 +71,11 @@ type parent interface {
// Source is a RTSP static source.
type Source struct {
ReadTimeout conf.Duration
WriteTimeout conf.Duration
WriteQueueSize int
Parent parent
ReadTimeout conf.Duration
WriteTimeout conf.Duration
WriteQueueSize int
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)

24
mediamtx.yml
View file

@ -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")