bluenviron / mediamtx

Ready-to-use SRT / WebRTC / RTSP / RTMP / LL-HLS media server and media proxy that allows to read, publish, proxy, record and playback video and audio streams.
MIT License
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RTSP does not extract user and password when using external auth server #2700 #2701

Closed jonra1993 closed 11 months ago

jonra1993 commented 11 months ago

Which version are you using?

v1.3.0

Which operating system are you using?

Describe the issue

RTSP does not extract user and password when using an external auth server. I have set up the external server and mediamtx does not send in the payload the user and passwords. I test with rtmp and it works.

Describe how to replicate the issue

  1. start the server in a container way and setup an external auth server
version: '3'
services:
  mediamtx:
    image: bluenviron/mediamtx:latest-ffmpeg
    ports:
      - "8554:8554"
      - "8002:8000"
      - "8001:8001"
      - "1935:1935"
      - "8888:8888"
      - "8889:8889"
      - "8890:8890/udp"
    environment:
      #- MTX_PROTOCOLS="udp, multicast, tcp"
      - MTX_API=yes      

Default config file

###############################################
# Global settings

# Settings in this section are applied anywhere.

###############################################
# Global settings -> General

# Verbosity of the program; available values are "error", "warn", "info", "debug".
logLevel: info
# Destinations of log messages; available values are "stdout", "file" and "syslog".
logDestinations: [stdout]
# If "file" is in logDestinations, this is the file which will receive the logs.
logFile: mediamtx.log

# 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

# HTTP URL to perform external authentication.
# Every time a user wants to authenticate, the server calls this URL
# with the POST method and a body containing:
# {
#   "ip": "ip",
#   "user": "user",
#   "password": "password",
#   "path": "path",
#   "protocol": "rtsp|rtmp|hls|webrtc",
#   "id": "id",
#   "action": "read|publish",
#   "query": "query"
# }
# If the response code is 20x, authentication is accepted, otherwise
# it is discarded.
externalAuthenticationURL: http://kong:8000/functions/v1/auth

# Enable the HTTP API.
api: yes
# Address of the API listener.
apiAddress: 127.0.0.1:9997

# Enable Prometheus-compatible metrics.
metrics: no
# Address of the metrics listener.
metricsAddress: 127.0.0.1:9998

# Enable pprof-compatible endpoint to monitor performances.
pprof: no
# Address of the pprof listener.
pprofAddress: 127.0.0.1:9999

# Command to run when a client connects to the server.
# This is terminated with SIGINT when a client disconnects from the server.
# The following environment variables are available:
# * RTSP_PORT: RTSP server port
# * MTX_CONN_TYPE: connection type
# * MTX_CONN_ID: connection ID
runOnConnect:
# Restart the command if it exits.
runOnConnectRestart: no
# Command to run when a client disconnects from the server.
# Environment variables are the same of runOnConnect.
runOnDisconnect:

###############################################
# Global settings -> RTSP

# Allow publishing and reading streams with the RTSP protocol.
rtsp: yes
# List of enabled RTSP transport protocols.
# UDP is the most performant, but doesn't work when there's a NAT/firewall between
# server and clients, and doesn't support encryption.
# UDP-multicast allows to save bandwidth when clients are all in the same LAN.
# TCP is the most versatile, and does support encryption.
# The handshake is always performed with TCP.
protocols: [udp, multicast, tcp]
# Encrypt handshakes and TCP streams with TLS (RTSPS).
# Available values are "no", "strict", "optional".
encryption: "no"
# Address of the TCP/RTSP listener. This is needed only when encryption is "no" or "optional".
rtspAddress: :8554
# Address of the TCP/TLS/RTSPS listener. This is needed only when encryption is "strict" or "optional".
rtspsAddress: :8322
# Address of the UDP/RTP listener. This is needed only when "udp" is in protocols.
rtpAddress: :8000
# Address of the UDP/RTCP listener. This is needed only when "udp" is in protocols.
rtcpAddress: :8001
# IP range of all UDP-multicast listeners. This is needed only when "multicast" is in protocols.
multicastIPRange: 224.1.0.0/16
# Port of all UDP-multicast/RTP listeners. This is needed only when "multicast" is in protocols.
multicastRTPPort: 8002
# Port of all UDP-multicast/RTCP listeners. This is needed only when "multicast" is in protocols.
multicastRTCPPort: 8003
# Path to the server key. This is needed only when encryption is "strict" or "optional".
# This can be generated with:
# openssl genrsa -out server.key 2048
# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650
serverKey: server.key
# Path to the server certificate. This is needed only when encryption is "strict" or "optional".
serverCert: server.crt
# Authentication methods. Available are "basic" and "digest".
# "digest" doesn't provide any additional security and is available for compatibility reasons only.
authMethods: [basic]

###############################################
# Global settings -> RTMP

# Allow publishing and reading streams with the RTMP protocol.
rtmp: yes
# Address of the RTMP listener. This is needed only when encryption is "no" or "optional".
rtmpAddress: :1935
# Encrypt connections with TLS (RTMPS).
# Available values are "no", "strict", "optional".
rtmpEncryption: "no"
# Address of the RTMPS listener. This is needed only when encryption is "strict" or "optional".
rtmpsAddress: :1936
# Path to the server key. This is needed only when encryption is "strict" or "optional".
# This can be generated with:
# openssl genrsa -out server.key 2048
# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650
rtmpServerKey: server.key
# Path to the server certificate. This is needed only when encryption is "strict" or "optional".
rtmpServerCert: server.crt

###############################################
# Global settings -> HLS

# Allow reading streams with the HLS protocol.
hls: yes
# Address of the HLS listener.
hlsAddress: :8888
# Enable TLS/HTTPS on the HLS server.
# This is required for Low-Latency HLS.
hlsEncryption: no
# Path to the server key. This is needed only when encryption is yes.
# This can be generated with:
# openssl genrsa -out server.key 2048
# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650
hlsServerKey: server.key
# Path to the server certificate.
hlsServerCert: server.crt
# By default, HLS is generated only when requested by a user.
# This option allows to generate it always, avoiding the delay between request and generation.
hlsAlwaysRemux: no
# Variant of the HLS protocol to use. Available options are:
# * mpegts - uses MPEG-TS segments, for maximum compatibility.
# * fmp4 - uses fragmented MP4 segments, more efficient.
# * lowLatency - uses Low-Latency HLS.
hlsVariant: lowLatency
# Number of HLS segments to keep on the server.
# Segments allow to seek through the stream.
# Their number doesn't influence latency.
hlsSegmentCount: 7
# Minimum duration of each segment.
# A player usually puts 3 segments in a buffer before reproducing the stream.
# The final segment duration is also influenced by the interval between IDR frames,
# since the server changes the duration in order to include at least one IDR frame
# in each segment.
hlsSegmentDuration: 1s
# Minimum duration of each part.
# A player usually puts 3 parts in a buffer before reproducing the stream.
# Parts are used in Low-Latency HLS in place of segments.
# Part duration is influenced by the distance between video/audio samples
# and is adjusted in order to produce segments with a similar duration.
hlsPartDuration: 200ms
# Maximum size of each segment.
# This prevents RAM exhaustion.
hlsSegmentMaxSize: 50M
# Value of the Access-Control-Allow-Origin header provided in every HTTP response.
# This allows to play the HLS stream from an external website.
hlsAllowOrigin: '*'
# List of IPs or CIDRs of proxies placed before the HLS server.
# If the server receives a request from one of these entries, IP in logs
# will be taken from the X-Forwarded-For header.
hlsTrustedProxies: []
# Directory in which to save segments, instead of keeping them in the RAM.
# This decreases performance, since reading from disk is less performant than
# reading from RAM, but allows to save RAM.
hlsDirectory: ''

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

# Allow publishing and reading streams with the WebRTC protocol.
webrtc: yes
# Address of the WebRTC listener.
webrtcAddress: :8889
# Enable TLS/HTTPS on the WebRTC server.
webrtcEncryption: no
# Path to the server key.
# This can be generated with:
# openssl genrsa -out server.key 2048
# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650
webrtcServerKey: server.key
# Path to the server certificate.
webrtcServerCert: server.crt
# 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: []
# List of ICE servers.
webrtcICEServers2:
  # URL can point to a STUN, TURN or TURNS server.
  # STUN servers are used to obtain the public IP of server and clients. They are
  # needed when server and clients are on different LANs.
  # TURN/TURNS servers are needed when a direct connection between server and
  # clients is not possible. All traffic is routed through them.
- 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: ''
# List of interfaces that will be used to gather IPs to send
# to the counterpart to establish a connection.
webrtcICEInterfaces: []
# List of public IP addresses that are to be used as a host.
# This is used typically for servers that are behind 1:1 D-NAT.
webrtcICEHostNAT1To1IPs: []
# Address of a ICE UDP listener in format host:port.
# If filled, ICE traffic will pass through a single UDP port,
# allowing the deployment of the server inside a container or behind a NAT.
webrtcICEUDPMuxAddress:
# Address of a ICE TCP listener in format host:port.
# If filled, ICE traffic will pass through a single TCP port,
# allowing the deployment of the server inside a container or behind a NAT.
# Using this setting forces usage of the TCP protocol, which is not
# optimal for WebRTC.
webrtcICETCPMuxAddress:

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

# Allow publishing and reading streams with the SRT protocol.
srt: yes
# Address of the SRT listener.
srtAddress: :8890

###############################################
# Default path settings

# Settings in "pathDefaults" are applied anywhere,
# unless they are overridden in "paths".
pathDefaults:

  ###############################################
  # Default path settings -> General

  # Source of the stream. This can be:
  # * publisher -> the stream is provided by a RTSP, RTMP, WebRTC or SRT client
  # * rtsp://existing-url -> the stream is pulled from another RTSP server / camera
  # * rtsps://existing-url -> the stream is pulled from another RTSP server / camera with RTSPS
  # * rtmp://existing-url -> the stream is pulled from another RTMP server / camera
  # * rtmps://existing-url -> the stream is pulled from another RTMP server / camera with RTMPS
  # * http://existing-url/stream.m3u8 -> the stream is pulled from another HLS server / camera
  # * https://existing-url/stream.m3u8 -> the stream is pulled from another HLS server / camera with HTTPS
  # * udp://ip:port -> the stream is pulled with UDP, by listening on the specified IP and port
  # * srt://existing-url -> the stream is pulled from another SRT server / camera
  # * whep://existing-url -> the stream is pulled from another WebRTC server / camera
  # * wheps://existing-url -> the stream is pulled from another WebRTC server / camera with HTTPS
  # * redirect -> the stream is provided by another path or server
  # * rpiCamera -> the stream is provided by a Raspberry Pi Camera
  source: publisher
  # If the source is a URL, and the source certificate is self-signed
  # or invalid, you can provide the fingerprint of the certificate in order to
  # validate it anyway. It can be obtained by running:
  # openssl s_client -connect source_ip:source_port </dev/null 2>/dev/null | sed -n '/BEGIN/,/END/p' > server.crt
  # openssl x509 -in server.crt -noout -fingerprint -sha256 | cut -d "=" -f2 | tr -d ':'
  sourceFingerprint:
  # If the source is a URL, it will be pulled only when at least
  # one reader is connected, saving bandwidth.
  sourceOnDemand: no
  # If sourceOnDemand is "yes", readers will be put on hold until the source is
  # ready or until this amount of time has passed.
  sourceOnDemandStartTimeout: 10s
  # If sourceOnDemand is "yes", the source will be closed when there are no
  # readers connected and this amount of time has passed.
  sourceOnDemandCloseAfter: 10s
  # Maximum number of readers. Zero means no limit.
  maxReaders: 0
  # SRT encryption passphrase require to read from this path
  srtReadPassphrase:

  ###############################################
  # Default path settings -> Recording

  # Record streams to disk.
  record: no
  # Path of recording segments.
  # Extension is added automatically.
  # Available variables are %path (path name), %Y %m %d %H %M %S %f (time in strftime format)
  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: 100ms
  # Minimum duration of each segment.
  recordSegmentDuration: 1h
  # Delete segments after this timespan.
  # Set to 0s to disable automatic deletion.
  recordDeleteAfter: 24h

  ###############################################
  # Default path settings -> Authentication

  # Username required to publish.
  # SHA256-hashed values can be inserted with the "sha256:" prefix.
  publishUser:
  # Password required to publish.
  # SHA256-hashed values can be inserted with the "sha256:" prefix.
  publishPass:
  # IPs or networks (x.x.x.x/24) allowed to publish.
  publishIPs: []

  # Username required to read.
  # SHA256-hashed values can be inserted with the "sha256:" prefix.
  readUser:
  # password required to read.
  # SHA256-hashed values can be inserted with the "sha256:" prefix.
  readPass:
  # IPs or networks (x.x.x.x/24) allowed to read.
  readIPs: []

  ###############################################
  # Default path settings -> Publisher source (when source is "publisher")

  # allow another client to disconnect the current publisher and publish in its place.
  overridePublisher: yes
  # if no one is publishing, redirect readers to this path.
  # It can be can be a relative path (i.e. /otherstream) or an absolute RTSP URL.
  fallback:
  # SRT encryption passphrase required to publish to this path
  srtPublishPassphrase:

  ###############################################
  # Default path settings -> RTSP source (when source is a RTSP or a RTSPS URL)

  # protocol used to pull the stream. available values are "automatic", "udp", "multicast", "tcp".
  sourceProtocol: automatic
  # support sources that don't provide server ports or use random server ports. This is a security issue
  # and must be used only when interacting with sources that require it.
  sourceAnyPortEnable: no
  # range header to send to the source, in order to start streaming from the specified offset.
  # available values:
  # * clock: Absolute time
  # * npt: Normal Play Time
  # * smpte: SMPTE timestamps relative to the start of the recording
  rtspRangeType:
  # available values:
  # * clock: UTC ISO 8601 combined date and time string, e.g. 20230812T120000Z
  # * 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:

  ###############################################
  # Default path settings -> Redirect source (when source is "redirect")

  # RTSP URL which clients will be redirected to.
  sourceRedirect:

  ###############################################
  # Default path settings -> Raspberry Pi Camera source (when source is "rpiCamera")

  # ID of the camera
  rpiCameraCamID: 0
  # width of frames
  rpiCameraWidth: 1920
  # height of frames
  rpiCameraHeight: 1080
  # flip horizontally
  rpiCameraHFlip: false
  # flip vertically
  rpiCameraVFlip: false
  # brightness [-1, 1]
  rpiCameraBrightness: 0
  # contrast [0, 16]
  rpiCameraContrast: 1
  # saturation [0, 16]
  rpiCameraSaturation: 1
  # sharpness [0, 16]
  rpiCameraSharpness: 1
  # exposure mode.
  # values: normal, short, long, custom
  rpiCameraExposure: normal
  # auto-white-balance mode.
  # values: auto, incandescent, tungsten, fluorescent, indoor, daylight, cloudy, custom
  rpiCameraAWB: auto
  # denoise operating mode.
  # values: off, cdn_off, cdn_fast, cdn_hq
  rpiCameraDenoise: "off"
  # fixed shutter speed, in microseconds.
  rpiCameraShutter: 0
  # metering mode of the AEC/AGC algorithm.
  # values: centre, spot, matrix, custom
  rpiCameraMetering: centre
  # fixed gain
  rpiCameraGain: 0
  # EV compensation of the image [-10, 10]
  rpiCameraEV: 0
  # Region of interest, in format x,y,width,height
  rpiCameraROI:
  # whether to enable HDR on Raspberry Camera 3.
  rpiCameraHDR: false
  # tuning file
  rpiCameraTuningFile:
  # sensor mode, in format [width]:[height]:[bit-depth]:[packing]
  # bit-depth and packing are optional.
  rpiCameraMode:
  # frames per second
  rpiCameraFPS: 30
  # period between IDR frames
  rpiCameraIDRPeriod: 60
  # bitrate
  rpiCameraBitrate: 1000000
  # H264 profile
  rpiCameraProfile: main
  # H264 level
  rpiCameraLevel: '4.1'
  # Autofocus mode
  # values: auto, manual, continuous
  rpiCameraAfMode: continuous
  # Autofocus range
  # values: normal, macro, full
  rpiCameraAfRange: normal
  # Autofocus speed
  # values: normal, fast
  rpiCameraAfSpeed: normal
  # Lens position (for manual autofocus only), will be set to focus to a specific distance
  # calculated by the following formula: d = 1 / value
  # Examples: 0 moves the lens to infinity.
  #           0.5 moves the lens to focus on objects 2m away.
  #           2 moves the lens to focus on objects 50cm away.
  rpiCameraLensPosition: 0.0
  # Specifies the autofocus window, in the form x,y,width,height where the coordinates
  # are given as a proportion of the entire image.
  rpiCameraAfWindow:
  # enables printing text on each frame.
  rpiCameraTextOverlayEnable: false
  # text that is printed on each frame.
  # format is the one of the strftime() function.
  rpiCameraTextOverlay: '%Y-%m-%d %H:%M:%S - MediaMTX'

  ###############################################
  # Default path settings -> Hooks

  # Command to run when this path is initialized.
  # This can be used to publish a stream when the server is launched.
  # This is terminated with SIGINT when the program closes.
  # The following environment variables are available:
  # * MTX_PATH: path name
  # * RTSP_PORT: RTSP server port
  # * G1, G2, ...: regular expression groups, if path name is
  #   a regular expression.
  runOnInit:
  # Restart the command if it exits.
  runOnInitRestart: no

  # Command to run when this path is requested by a reader.
  # This can be used to publish a stream on demand.
  # This is terminated with SIGINT when the path is not requested anymore.
  # The following environment variables are available:
  # * MTX_PATH: path name
  # * MTX_QUERY: query parameters (passed by first reader)
  # * RTSP_PORT: RTSP server port
  # * G1, G2, ...: regular expression groups, if path name is
  #   a regular expression.
  runOnDemand:
  # Restart the command if it exits.
  runOnDemandRestart: no
  # Readers will be put on hold until the runOnDemand command starts publishing
  # or until this amount of time has passed.
  runOnDemandStartTimeout: 10s
  # The command will be closed when there are no
  # readers connected and this amount of time has passed.
  runOnDemandCloseAfter: 10s

  # Command to run when the stream is ready to be read, whenever it is
  # published by a client or pulled from a server / camera.
  # This is terminated with SIGINT when the stream is not ready anymore.
  # The following environment variables are available:
  # * MTX_PATH: path name
  # * MTX_QUERY: query parameters (passed by publisher)
  # * RTSP_PORT: RTSP server port
  # * G1, G2, ...: regular expression groups, if path name is
  #   a regular expression.
  # * MTX_SOURCE_TYPE: source type
  # * MTX_SOURCE_ID: source ID
  runOnReady:
  # Restart the command if it exits.
  runOnReadyRestart: no
  # Command to run when the stream is not available anymore.
  # Environment variables are the same of runOnReady.
  runOnNotReady:

  # Command to run when a client starts reading.
  # This is terminated with SIGINT when a client stops reading.
  # The following environment variables are available:
  # * MTX_PATH: path name
  # * MTX_QUERY: query parameters (passed by reader)
  # * RTSP_PORT: RTSP server port
  # * G1, G2, ...: regular expression groups, if path name is
  #   a regular expression.
  # * MTX_READER_TYPE: reader type
  # * MTX_READER_ID: reader ID
  runOnRead:
  # Restart the command if it exits.
  runOnReadRestart: no
  # Command to run when a client stops reading.
  # Environment variables are the same of runOnRead.
  runOnUnread:

  # Command to run when a recording segment is created.
  # The following environment variables are available:
  # * MTX_PATH: path name
  # * RTSP_PORT: RTSP server port
  # * G1, G2, ...: regular expression groups, if path name is
  #   a regular expression.
  # * MTX_SEGMENT_PATH: segment file path
  runOnRecordSegmentCreate:

  # Command to run when a recording segment is complete.
  # The following environment variables are available:
  # * MTX_PATH: path name
  # * RTSP_PORT: RTSP server port
  # * G1, G2, ...: regular expression groups, if path name is
  #   a regular expression.
  # * MTX_SEGMENT_PATH: segment file path
  runOnRecordSegmentComplete:

###############################################
# Path settings

# Settings in "paths" are applied to specific paths, and the map key
# is the name of the path.
# Any setting in "pathDefaults" can be overridden here.
# It's possible to use regular expressions by using a tilde as prefix,
# for example "~^(test1|test2)$" will match both "test1" and "test2",
# for example "~^prefix" will match all paths that start with "prefix".
paths:
  # example:
  # my_camera:
  #   source: rtsp://my_camera

  # Settings under path "all_others" are applied to all paths that
  # do not match another entry.
  all_others:
  1. publish with .
    ffmpeg -f avfoundation -video_size 640x480 -i "0" -c:v h264 -f rtsp -rtsp_transport tcp "rtsp://myuser:mypass@localhost:8554/mystream?user=myuser&pass=mypass"
  2. print payload in auth server

Did you attach the server logs?

yes

image

Did you attach a network dump?

no

krusadellc commented 11 months ago

Did you try sending a 4XX error response in your authentication API endpoint? We have seen some RTSP clients like VLC do not even attempt a basic auth unless an error response is sent back by the auth endpoint.

jonra1993 commented 11 months ago

Hello @krusadellc I have done that it works perfectly with rtmp when I send credentials it auth when they miss I send a 4xx as I mentioned in the issue Mediamtx is not sending user and pass to auth server. For now, I am send them from query params and extract from them. But I still have user and password as empty strings. The log from by auth server I am not suing VLC

aler9 commented 11 months ago

Hello, i tested again but in my case the external authentication mechanism worked without any issue. Make sure to return code 400 in case of empty credentials, otherwise the client won't send credentials. You didn't post the entire log therefore it's impossible to understand whether you're implementing the server correctly or not. Here's a sample code.

package main

import (
    "encoding/json"
    "fmt"
    "net"
    "net/http"
    "context"

    "github.com/gin-gonic/gin"
)

type server struct {
    s *http.Server
}

func newServer() (*server, error) {
    ln, err := net.Listen("tcp", ":9120")
    if err != nil {
        return nil, err
    }

    ts := &server{}

    router := gin.New()
    router.POST("/auth", ts.onAuth)

    ts.s = &http.Server{Handler: router}
    go ts.s.Serve(ln)

    return ts, nil
}

func (ts *server) close() {
    ts.s.Shutdown(context.Background())
}

func (ts *server) onAuth(ctx *gin.Context) {
    var in struct {
        IP       string `json:"ip"`
        User     string `json:"user"`
        Password string `json:"password"`
        Path     string `json:"path"`
        Action   string `json:"action"`
        Query    string `json:"query"`
    }
    err := json.NewDecoder(ctx.Request.Body).Decode(&in)
    if err != nil {
        ctx.AbortWithStatus(http.StatusBadRequest)
        return
    }

    fmt.Printf("%+v\n", in)

    if in.User == "" {
        ctx.AbortWithStatus(http.StatusUnauthorized)
        return
    }

    ctx.Status(http.StatusOK)
}

func main() {
    _, err := newServer()
    if err != nil {
        panic(err)
    }
    select {}
}
jonra1993 commented 11 months ago

Hello @aler9 thanks I solved it it was missing the 400 error when the first request user is empty I see Mediamtx sends another request.

image
saket424 commented 11 months ago

In releases earlier to 1.3, an error code of 401 used to work when username and password were empty. With 1.3 the error code must be 400, otherwise things don't authenticate successfully

aler9 commented 10 months ago

@saket424 i just tested external authentication against v1.4.0 and it works well when the code is ether 401 or 400, there's no difference at all. Currently, the server just checks whether status code is inside range 200-299 or not:

https://github.com/bluenviron/mediamtx/blob/94953f5d22ae31056cc96c04d23c807f02c0f29c/internal/core/auth.go#L69-L73

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