MediaMTX Бесконечно перезапускается

я пытаюсь сделать трансляцию на сервер Mediamtx по SRT, однако изменение в paths в mediamtx.yml приводят к бесконечному перезапуску контейнера.

Я пытаюсь соделать трансляцию на сервер MediaMTX через OBS Studio с помощью данной ссылки:

srt://192.168.140.36:25590?streamid=#!::r=home_camera

В логах MediaMTX следующая ошибка:

2025/08/25 12:29:32 INF [SRT] [conn 10.7.0.41:61321] opened
2025/08/25 12:29:32 DEB [path home_camera] created
2025/08/25 12:29:32 INF [SRT] [conn 10.7.0.41:61321] closed: no stream is available on path 'home_camera'
2025/08/25 12:29:32 DEB [path home_camera] destroyed: not in use

Насколько я понял он ругается что home_camera не прописан в paths в mediamtx.yml. Я добавил его, однако это приводит к бесконечному перезапуску контейнера. Вот код mediamtx.yml:

# Включаем HLS
# hls: Включает поддержку HLS.
# hlsAddress: Адрес и порт, на котором будет доступен HLS (например, :8888 означает порт 8888 на всех интерфейсах).
# hlsSegmentDuration: Продолжительность каждого сегмента HLS.
# hlsPartDuration: Продолжительность каждой части сегмента (для Low-Latency HLS).
# hlsSegmentCount: Количество сегментов, хранящихся в плейлисте.
hls: no
hlsAddress: :8888
hlsSegmentDuration: 1s
hlsPartDuration: 500ms
hlsSegmentCount: 10

# Timeout of read operations.
readTimeout: 10s
# Timeout of write operations.
writeTimeout: 10s
# Size of the queue of outgoing packets.
# A higher value allows to increase throughput, a lower value allows to save RAM.
writeQueueSize: 512
# Maximum size of outgoing UDP packets.
# This can be decreased to avoid fragmentation on networks with a low UDP MTU.
udpMaxPayloadSize: 1472

# Enable controlling the server through the Control API.
api: no
# Address of the Control API listener.
apiAddress: :9997

pathDefaults:
  record: yes
  # Path of recording segments.
  # Extension is added automatically.
  # Available variables are %path (path name), %Y %m %d (year, month, day),
  # %H %M %S (hours, minutes, seconds), %f (microseconds), %s (unix epoch).
  recordPath: ./recordings/%path/%Y-%m-%d_%H-%M-%S-%f
  # Format of recorded segments.
  # Available formats are "fmp4" (fragmented MP4) and "mpegts" (MPEG-TS).
  recordFormat: fmp4
  # fMP4 segments are concatenation of small MP4 files (parts), each with this duration.
  # MPEG-TS segments are concatenation of 188-bytes packets, flushed to disk with this period.
  # When a system failure occurs, the last part gets lost.
  # Therefore, the part duration is equal to the RPO (recovery point objective).
  recordPartDuration: 1s
  # Minimum duration of each segment.
  recordSegmentDuration: 15m
  # Delete segments after this timespan.
  # Set to 0s to disable automatic deletion.
  recordDeleteAfter: 20m


##############################################
#Настройка RTSP
rtsp: yes
rtspAddress: :25554

###############################################
# Global settings -> SRT server

# Enable publishing and reading streams with the SRT protocol.
srt: yes
# Address of the SRT listener.
srtAddress: :25590


###############################################
# Global settings -> WebRTC server

# Enable publishing and reading streams with the WebRTC protocol.
webrtc: yes
# Важный HTTP порт
webrtcAddress: :25889
# Важный UDP порт
webrtcLocalUDPAddress: :25189
# Value of the Access-Control-Allow-Origin header provided in every HTTP response.
# This allows to play the WebRTC stream from an external website.
webrtcAllowOrigin: '*'
# List of IPs or CIDRs of proxies placed before the WebRTC server.
# If the server receives a request from one of these entries, IP in logs
# will be taken from the X-Forwarded-For header.
webrtcTrustedProxies: []
# Address of a local TCP listener that will receive connections.
# This is disabled by default since TCP is less efficient than UDP and
# introduces a progressive delay when network is congested.
webrtcLocalTCPAddress: ''
# WebRTC clients need to know the IP of the server.
# Gather IPs from interfaces and send them to clients.
webrtcIPsFromInterfaces: yes
# List of interfaces whose IPs will be sent to clients.
# An empty value means to use all available interfaces.
webrtcIPsFromInterfacesList: []
# List of additional hosts or IPs to send to clients.
webrtcAdditionalHosts: [192.168.140.36]
# ICE servers. Needed only when local listeners can't be reached by clients.
# STUN servers allows to obtain and share the public IP of the server.
# TURN/TURNS servers forces all traffic through them.
webrtcICEServers2: []
  # - url: stun:stun.l.google.com:19302
  # if user is "AUTH_SECRET", then authentication is secret based.
  # the secret must be inserted into the password field.
  # username: ''
  # password: ''
# clientOnly: false
# Time to wait for the WebRTC handshake to complete.
webrtcHandshakeTimeout: 10s
# Maximum time to gather video tracks.
webrtcTrackGatherTimeout: 2s
# The maximum time to gather STUN candidates.
webrtcSTUNGatherTimeout: 5s

# Включаем HTTP серверы
playback: no
playbackAddress: :9996

# Уровень логирования pathDefaults.record
#logLevel: warn
logLevel: debug

paths:
  home_camera:
    srtSource: enable
    source: publisher
  all:
    source: publisher

Помогите пожалуйста, при необходимости могу скинуть все необходимые конфиги. Заранее спасибо


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