ready-to-use RTSP / RTMP server and proxy that allows to read, publish and proxy video and audio streams

Overview

rtsp-simple-server

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rtsp-simple-server is a simple, ready-to-use and zero-dependency RTSP / RTMP server and proxy, a software that allows users to publish, read and proxy live video and audio streams. RTSP is a specification that describes how to perform these operations with the help of a server, that is contacted by both publishers and readers and relays the publisher's streams to the readers.

Features:

  • Publish live streams with RTSP (UDP or TCP mode) or RTMP
  • Read live streams with RTSP or RTMP
  • Pull and serve streams from other RTSP / RTMP servers or cameras, always or on-demand (RTSP proxy)
  • Each stream can have multiple video and audio tracks, encoded with any codec (including H264, H265, VP8, VP9, MPEG2, MP3, AAC, Opus, PCM, JPEG)
  • Serve multiple streams at once in separate paths
  • Encrypt streams with TLS (RTSPS)
  • Authenticate readers and publishers
  • Redirect readers to other RTSP servers (load balancing)
  • Run custom commands when clients connect, disconnect, read or publish streams
  • Reload the configuration without disconnecting existing clients (hot reloading)
  • Compatible with Linux, Windows and macOS, does not require any dependency or interpreter, it's a single executable

Table of contents

Installation

Standard

  1. Download and extract a precompiled binary from the release page.

  2. Start the server:

    ./rtsp-simple-server
    

Docker

Download and launch the image:

docker run --rm -it --network=host aler9/rtsp-simple-server

The --network=host flag is mandatory since Docker can change the source port of UDP packets for routing reasons, and this doesn't allow to find out the publisher of the packets. This issue can be avoided by disabling UDP and exposing the RTSP port:

docker run --rm -it -e RTSP_PROTOCOLS=tcp -p 8554:8554 -p 1935:1935 aler9/rtsp-simple-server

Basic usage

  1. Publish a stream. For instance, you can publish a video/audio file with FFmpeg:

    ffmpeg -re -stream_loop -1 -i file.ts -c copy -f rtsp rtsp://localhost:8554/mystream
    

    or GStreamer:

    gst-launch-1.0 rtspclientsink name=s location=rtsp://localhost:8554/mystream filesrc location=file.mp4 ! qtdemux name=d d.video_0 ! queue ! s.sink_0 d.audio_0 ! queue ! s.sink_1
    
  2. Open the stream. For instance, you can open the stream with VLC:

    vlc rtsp://localhost:8554/mystream
    

    or GStreamer:

    gst-launch-1.0 rtspsrc location=rtsp://localhost:8554/mystream name=s s. ! application/x-rtp,media=video ! decodebin ! autovideosink s. ! application/x-rtp,media=audio ! decodebin ! audioconvert ! audioresample ! autoaudiosink
    

    or FFmpeg:

    ffmpeg -i rtsp://localhost:8554/mystream -c copy output.mp4
    

Advanced usage and FAQs

Configuration

All the configuration parameters are listed and commented in the configuration file.

There are two ways to change the configuration:

  • By editing the rtsp-simple-server.yml file, that is

    • included into the release bundle

    • available in the root folder of the Docker image (/rtsp-simple-server.yml); it can be overridden in this way:

      docker run --rm -it --network=host -v $PWD/rtsp-simple-server.yml:/rtsp-simple-server.yml aler9/rtsp-simple-server
      
  • By overriding configuration parameters with environment variables, in the format RTSP_PARAMNAME, where PARAMNAME is the uppercase name of a parameter. For instance, the rtspPort parameter can be overridden in the following way:

    RTSP_RTSPPORT=8555 ./rtsp-simple-server
    

    Parameters in maps can be overridden by using underscores, in the following way:

    RTSP_PATHS_TEST_SOURCE=rtsp://myurl ./rtsp-simple-server
    

    This method is particularly useful when using Docker; any configuration parameter can be changed by passing environment variables with the -e flag:

    docker run --rm -it --network=host -e RTSP_PATHS_TEST_SOURCE=rtsp://myurl aler9/rtsp-simple-server
    

The configuration can be changed dinamically when the server is running (hot reloading) by writing to the configuration file. Changes are detected and applied without disconnecting existing clients, whenever it's possible.

Encryption

Incoming and outgoing streams can be encrypted with TLS (obtaining the RTSPS protocol). A self-signed TLS certificate is needed and can be generated with openSSL:

openssl genrsa -out server.key 2048
openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650

Edit rtsp-simple-server.yml, and set the protocols, encrypt, serverKey and serverCert parameters:

protocols: [tcp]
encryption: optional
serverKey: server.key
serverCert: server.crt

Streams can then be published and read with the rtsps scheme and the 8555 port:

ffmpeg -i rtsps://ip:8555/...

If the client is GStreamer, disable the certificate validation:

gst-launch-1.0 rtspsrc location=rtsps://ip:8555/... tls-validation-flags=0

If the client is VLC, encryption can't be deployed, since VLC doesn't support it.

Authentication

Edit rtsp-simple-server.yml and replace everything inside section paths with the following content:

paths:
  all:
    publishUser: myuser
    publishPass: mypass

Only publishers that provide both username and password will be able to proceed:

ffmpeg -re -stream_loop -1 -i file.ts -c copy -f rtsp rtsp://myuser:[email protected]:8554/mystream

It's possible to setup authentication for readers too:

paths:
  all:
    publishUser: myuser
    publishPass: mypass

    readUser: user
    readPass: userpass

If storing plain credentials in the configuration file is a security problem, username and passwords can be stored as sha256-hashed strings; a string must be hashed with sha256 and encoded with base64:

echo -n "userpass" | openssl dgst -binary -sha256 | openssl base64

Then stored with the sha256: prefix:

paths:
  all:
    readUser: sha256:j1tsRqDEw9xvq/D7/9tMx6Jh/jMhk3UfjwIB2f1zgMo=
    readPass: sha256:BdSWkrdV+ZxFBLUQQY7+7uv9RmiSVA8nrPmjGjJtZQQ=

WARNING: enable encryption or use a VPN to ensure that no one is intercepting the credentials.

Encrypt the configuration

The configuration file can be entirely encrypted for security purposes.

An online encryption tool is available here.

The encryption procedure is the following:

  1. NaCL's crypto_secretbox function is applied to the content of the configuration. NaCL is a cryptographic library available for C/C++, Go, C# and many other languages;

  2. The string is prefixed with the nonce;

  3. The string is encoded with base64.

After performing the encryption, it's enough to put the base64-encoded result into the configuration file, and launch the server with the RTSP_CONFKEY variable:

RTSP_CONFKEY=mykey ./rtsp-simple-server

Proxy mode

rtsp-simple-server is also a RTSP and RTMP proxy, that is usually deployed in one of these scenarios:

  • when there are multiple users that are receiving a stream and the bandwidth is limited; the proxy is used to receive the stream once. Users can then connect to the proxy instead of the original source.
  • when there's a NAT / firewall between a stream and the users; the proxy is installed on the NAT and makes the stream available to the outside world.

Edit rtsp-simple-server.yml and replace everything inside section paths with the following content:

paths:
  proxied:
    # url of the source stream, in the format rtsp://user:[email protected]:port/path
    source: rtsp://original-url

After starting the server, users can connect to rtsp://localhost:8554/proxied, instead of connecting to the original url. The server supports any number of source streams, it's enough to add additional entries to the paths section:

paths:
  proxied1:
    source: rtsp://url1

  proxied2:
    source: rtsp://url1

It's possible to save bandwidth by enabling the on-demand mode: the stream will be pulled only when at least a client is connected:

paths:
  proxied:
    source: rtsp://original-url
    sourceOnDemand: yes

RTMP protocol

RTMP is a protocol that is used to read and publish streams, but is less versatile and less efficient than RTSP (doesn't support UDP, encryption, doesn't support most RTSP codecs, doesn't support feedback mechanism). It is used when there's need of publishing or reading streams from a software that supports only RTMP (for instance, OBS Studio and DJI drones).

At the moment, only the H264 and AAC codecs can be used with the RTMP protocol.

Streams can be published or read with the RTMP protocol, for instance with FFmpeg:

ffmpeg -re -stream_loop -1 -i file.ts -c copy -f flv rtmp://localhost/mystream

or GStreamer:

gst-launch-1.0 -v flvmux name=s ! rtmpsink location=rtmp://localhost/mystream filesrc location=file.mp4 ! qtdemux name=d d.video_0 ! queue ! s.video d.audio_0 ! queue ! s.audio

Credentials can be provided by appending to the URL the user and pass parameters:

ffmpeg -re -stream_loop -1 -i file.ts -c copy -f flv rtmp://localhost:8554/mystream?user=myuser&pass=mypass

Publish from OBS Studio

In Settings -> Stream (or in the Auto-configuration Wizard), use the following parameters:

  • Service: Custom...
  • Server: rtmp://localhost
  • Stream key: mystream

If credentials are in use, use the following parameters:

  • Service: Custom...
  • Server: rtmp://localhost
  • Stream key: mystream?user=myuser&pass=mypass

Publish a webcam

Edit rtsp-simple-server.yml and replace everything inside section paths with the following content:

paths:
  cam:
    runOnInit: ffmpeg -f v4l2 -i /dev/video0 -f rtsp rtsp://localhost:$RTSP_PORT/$RTSP_PATH
    runOnInitRestart: yes

If the platform is Windows:

paths:
  cam:
    runOnInit: ffmpeg -f dshow -i video="USB2.0 HD UVC WebCam" -f rtsp rtsp://localhost:$RTSP_PORT/$RTSP_PATH
    runOnInitRestart: yes

Where USB2.0 HD UVC WebCam is the name of your webcam, that can be obtained with:

ffmpeg -list_devices true -f dshow -i dummy

After starting the server, the webcam can be reached on rtsp://localhost:8554/cam.

Publish a Raspberry Pi Camera

Install dependencies:

  1. Gstreamer

    sudo apt install -y gstreamer1.0-tools gstreamer1.0-rtsp
    
  2. gst-rpicamsrc, by following instruction here

Then edit rtsp-simple-server.yml and replace everything inside section paths with the following content:

paths:
  cam:
    runOnInit: gst-launch-1.0 rpicamsrc preview=false bitrate=2000000 keyframe-interval=50 ! video/x-h264,width=1920,height=1080,framerate=25/1 ! h264parse ! rtspclientsink location=rtsp://localhost:$RTSP_PORT/$RTSP_PATH
    runOnInitRestart: yes

After starting the server, the camera is available on rtsp://localhost:8554/cam.

Convert streams to HLS

HLS is a media format that allows to embed live streams into web pages, inside standard HTML tags. To generate HLS whenever someone publishes a stream, edit rtsp-simple-server.yml and replace everything inside section paths with the following content:

paths:
  all:
    runOnPublish: ffmpeg -i rtsp://localhost:$RTSP_PORT/$RTSP_PATH -c copy -f hls -hls_time 1 -hls_list_size 3 -hls_flags delete_segments -hls_allow_cache 0 stream.m3u8
    runOnPublishRestart: yes

The resulting files (stream.m3u8 and a lot of .ts segments) can be served by a web server.

The example above makes the assumption that published streams are encoded with H264 and AAC, since they are the only codecs supported by HLS; if streams make use of different codecs, they must be converted:

paths:
  all:
    runOnPublish: ffmpeg -i rtsp://localhost:$RTSP_PORT/$RTSP_PATH -c:a aac -b:a 64k -c:v libx264 -preset ultrafast -b:v 500k -f hls -hls_time 1 -hls_list_size 3 -hls_flags delete_segments -hls_allow_cache 0 stream.m3u8
    runOnPublishRestart: yes

Remuxing, re-encoding, compression

To change the format, codec or compression of a stream, use FFmpeg or Gstreamer together with rtsp-simple-server. For instance, to re-encode an existing stream, that is available in the /original path, and publish the resulting stream in the /compressed path, edit rtsp-simple-server.yml and replace everything inside section paths with the following content:

paths:
  all:
  original:
    runOnPublish: ffmpeg -i rtsp://localhost:$RTSP_PORT/$RTSP_PATH -c:v libx264 -preset ultrafast -b:v 500k -max_muxing_queue_size 1024 -f rtsp rtsp://localhost:$RTSP_PORT/compressed
    runOnPublishRestart: yes

On-demand publishing

Edit rtsp-simple-server.yml and replace everything inside section paths with the following content:

paths:
  ondemand:
    runOnDemand: ffmpeg -re -stream_loop -1 -i file.ts -c copy -f rtsp rtsp://localhost:$RTSP_PORT/$RTSP_PATH
    runOnDemandRestart: yes

The command inserted into runOnDemand will start only when a client requests the path ondemand, therefore the file will start streaming only when requested.

Redirect to another server

To redirect to another server, use the redirect source:

paths:
  redirected:
    source: redirect
    sourceRedirect: rtsp://otherurl/otherpath

Fallback stream

If no one is publishing to the server, readers can be redirected to a fallback path or URL that is serving a fallback stream:

paths:
  withfallback:
    fallback: /otherpath

Start on boot with systemd

Systemd is the service manager used by Ubuntu, Debian and many other Linux distributions, and allows to launch rtsp-simple-server on boot.

Download a release bundle from the release page, unzip it, and move the executable and configuration in the system:

sudo mv rtsp-simple-server /usr/local/bin/
sudo mv rtsp-simple-server.yml /usr/local/etc/

Create the service:

sudo tee /etc/systemd/system/rtsp-simple-server.service >/dev/null << EOF
[Unit]
After=network.target
[Service]
ExecStart=/usr/local/bin/rtsp-simple-server /usr/local/etc/rtsp-simple-server.yml
[Install]
WantedBy=multi-user.target
EOF

Enable and start the service:

sudo systemctl enable rtsp-simple-server
sudo systemctl start rtsp-simple-server

Monitoring

There are multiple ways to monitor the server usage over time:

  • The current number of clients, publishers and readers is printed in each log line; for instance, the line:

    2020/01/01 00:00:00 [2/1/1] [client 127.0.0.1:44428] OPTION
    

    means that there are 2 clients, 1 publisher and 1 reader.

  • A metrics exporter, compatible with Prometheus, can be enabled with the parameter metrics: yes; then the server can be queried for metrics with Prometheus or with a simple HTTP request:

    wget -qO- localhost:9998/metrics
    

    Obtaining:

    rtsp_clients{state="idle"} 2 1596122687740
    rtsp_clients{state="publishing"} 15 1596122687740
    rtsp_clients{state="reading"} 8 1596122687740
    rtsp_sources{type="rtsp",state="idle"} 3 1596122687740
    rtsp_sources{type="rtsp",state="running"} 2 1596122687740
    rtsp_sources{type="rtmp",state="idle"} 1 1596122687740
    rtsp_sources{type="rtmp",state="running"} 0 1596122687740
    

    where:

    • rtsp_clients{state="idle"} is the count of clients that are neither publishing nor reading
    • rtsp_clients{state="publishing"} is the count of clients that are publishing
    • rtsp_clients{state="reading"} is the count of clients that are reading
    • rtsp_sources{type="rtsp",state="idle"} is the count of rtsp sources that are not running
    • rtsp_sources{type="rtsp",state="running"} is the count of rtsp sources that are running
    • rtsp_sources{type="rtmp",state="idle"} is the count of rtmp sources that are not running
    • rtsp_sources{type="rtmp",state="running"} is the count of rtmp sources that are running
  • A performance monitor, compatible with pprof, can be enabled with the parameter pprof: yes; then the server can be queried for metrics with pprof-compatible tools, like:

    go tool pprof -text http://localhost:9999/debug/pprof/goroutine
    go tool pprof -text http://localhost:9999/debug/pprof/heap
    go tool pprof -text http://localhost:9999/debug/pprof/profile?seconds=30
    

Command-line usage

usage: rtsp-simple-server []

rtsp-simple-server v0.0.0

RTSP server.

Flags:
  --help     Show context-sensitive help (also try --help-long and --help-man).
  --version  print version

Args:
  []  path to a config file. The default is rtsp-simple-server.yml.

Compile and run from source

Install Go 1.15, download the repository, open a terminal in it and run:

go run .

You can perform the entire operation inside Docker:

make run

Links

Related projects

IETF Standards

Conventions

Comments
  • Video playback issues with specific MP4 files

    Video playback issues with specific MP4 files

    Which version are you using?

    v0.20.2

    Which operating system are you using?

    • [X] Linux amd64 standard
    • [X] Linux amd64 Docker
    • [X] Linux arm64 standard
    • [X] Linux arm64 Docker
    • [X] Linux arm7 standard
    • [X] Linux arm7 Docker

    Describe the issue

    First, thanks for this terrific project. I use rtsp-simple-server as part of the ring-mqtt project which allows users of Ring camera to access these devices and event records via standard RTSP, leverage run-on-demand and it works great for the most part.

    For event recordings Ring offers two different types of MP4 files, one is the raw recording exactly as the camera took it, and a second one, referred to as the transcoded file, includes additional data such as a timestamp overlay and, based on the camera capabilities, potentially some pre-roll video to show the seconds leading up to the triggered event.

    Currently the code can play back the recorded raw video just fine, but I had quite a few users that asked about playing back the transcoded video, but, when adding this capability I ran into an issue, for some reason the transcoded video, when streamed through rtsp-simple-server, many times the first 5-10 seconds of the video portion is black, then the video finally starts to play (audio starts immediately). Weirdly, every now and then it will play correctly from the start, and this seemed to improve with 0.20.2 release, but it's still not quite working perfectly.

    At first I thought this was some issue with ffmpeg, but I decided to try streaming the file with go2rtc instead of rtsp-simple-server and there the file always plays back perfectly. Since the ffmpeg command used by go2rtc and rtsp-simple-server are nearly identical, the only significant variable is rtsp-simple-server.

    To summarize:

    test_event.mp4 -> ffmpeg -> rtsp-simple-server -> vlc = Playback via VLC stutters at points and/or black screen to start test_event.mp4 -> ffmpeg -> go2rtc (RTSP module) -> vlc = Playback works perfectly, no different than directly opening mp4 file

    Describe how to replicate the issue

    1. start the server with this config:
    paths:
      file1:
        runOnDemand: ffmpeg -re -stream_loop -1 -i /media/test_event.mp4 -c copy -rtsp_transport tcp -f rtsp rtsp://localhost:$RTSP_PORT/$RTSP_PATH
        runOnDemandRestart: yes
    
    1. read with vlc from rtsp://localhost:8554/file1

    Did you attach the server logs?

    Yes, with debug enabled. rtsp-simple-server.log

    Did you attach a network dump?

    No, but I can do this if it would help somehow. I will also put a link to the video to use to reproduce the issue.

    opened by tsightler 1
  • Possible combatibility issue with RTMP and with Streamlabs Mobile app Android version

    Possible combatibility issue with RTMP and with Streamlabs Mobile app Android version

    Which version are you using?

    v0.20.2 (have tried 0.19.x too)

    Which operating system are you using?

    • [X ] Linux amd64 standard

    Describe the issue

    Streamlabs mobile Android version app ( https://play.google.com/store/apps/details?id=com.streamlabs )doesn't work with RTMP. Basically based on logs it tries to connect twice and does i/o timeout. Current Android version of it is 3.6.12.171. This issue has been at least since summer.

    What works: Streamlabs Mobile and their iOS version of it. Also Camerafi Live on Android and Larix Android works.

    I am using default settings out-of-the-box from RTSP-simple server or my custom. It doesn't matter, it acts like the same way. I am using the same network LAN and there is no firewall or network issue and their iOS version works and other similar Android apps works fine too.

    Describe how to replicate the issue

    1. start the server
    2. Try to Stream with Streamlabs Mobile Android app with RTMP

    Did you attach the server logs?

    yes

    Did you attach a network dump?

    no

    opened by notwildsystem 0
  • Support reading with WebRTC

    Support reading with WebRTC

    Fixes #566

    Usage

    1. download a nightly release or build from source:

      rtsp-simple-server_v0.20.2-9-g9399f6a_darwin_amd64.tar.gz rtsp-simple-server_v0.20.2-9-g9399f6a_linux_amd64.tar.gz rtsp-simple-server_v0.20.2-9-g9399f6a_linux_arm64v8.tar.gz rtsp-simple-server_v0.20.2-9-g9399f6a_linux_armv6.tar.gz rtsp-simple-server_v0.20.2-9-g9399f6a_linux_armv7.tar.gz rtsp-simple-server_v0.20.2-9-g9399f6a_windows_amd64.zip

    2. get or generate a TLS certificate:

      openssl genrsa -out server.key 2048
      openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650
      
    3. start the server and publish something

    4. visit `https://server-host:8889/path-of-the-stream

    Configuration

    # Disable support for the WebRTC protocol.
    webrtcDisable: no
    # Address of the WebRTC listener.
    webrtcAddress: :8889
    # Path to the server key. This is mandatory since HTTPS is mandatory in order to use WebRTC.
    # 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 STUN servers. These are used in WebRTC to get the public IP of both server and clients.
    webrtcStunServers: [stun.l.google.com:19302]
    

    Roadmap

    • [x] read H264 tracks
    • [ ] read Opus tracks
    • [ ] read PCMA tracks
    • [ ] read PCMU tracks
    • [ ] read VP8 tracks
    • [ ] read VP9 tracks
    • [ ] authentication
    • [ ] API
    • [ ] metrics
    • [ ] TURN servers
    • [ ] publish video/audio from a browser
    • [ ] find a way to generate and store a TLS certificate if the default one is missing
    opened by aler9 0
  • externalAuthenticationURL set and called, but POST body is empty

    externalAuthenticationURL set and called, but POST body is empty

    Which version are you using?

    v0.20.2

    Which operating system are you using?

    • [ ] Linux amd64 standard
    • [x] Linux amd64 Docker
    • [ ] Linux arm64 standard
    • [ ] Linux arm64 Docker
    • [ ] Linux arm7 standard
    • [ ] Linux arm7 Docker
    • [ ] Linux arm6 standard
    • [ ] Linux arm6 Docker
    • [ ] Windows amd64 standard
    • [ ] Windows amd64 Docker (WSL backend)
    • [ ] macOS amd64 standard
    • [ ] macOS amd64 Docker
    • [ ] Other (please describe)

    Describe the issue

    First of all, thanks for this great project. Really helpful!

    I think I found a bug in the external Authentication. I am using the latest docker version of rtsp-simple-server (using version 0.20.2)

    My config looks like this:

    externalAuthenticationURL: http://10.20.20.75/testauth
    
    paths:
      proxied1:
        # url of the source stream, in the format rtsp://user:[email protected]:port/path
        source: rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mp4
        sourceOnDemand: yes
    
      proxied2:
        source: rtsp://rtsp.stream/pattern
        sourceOnDemand: yes
    

    The testauth endpoint is a simple nodejs express server. When opening one of the streams using HLS with Chrome, the server will call the auth api endpoint, but the POST request will be empty.

    Can you confirm this behaviour? Or am I missing something?

    Describe how to replicate the issue

    1. Start docker container docker run --rm -it --network=host -v $PWD/rtsp-test.yml:/rtsp-simple-server.yml:ro aler9/rtsp-simple-server

    2. Open the resulting HLS stream in Chrome http://myip/proxied2

    3. The authentication endpoint will receive a POST request from the server, but with an empty body

    4. If the auth server will return 200, the stream will load

    5. If the auth server will return 401, the basic auth dialog will be shown in the browser. When accepted, the auth server will receive another empty POST request. The stream will not be loaded in the end. The server will log [HLS] [muxer proxied2] authentication error: external authentication failed: bad status code: 401

    Did you attach the server logs?

    no

    Did you attach a network dump?

    no

    opened by j-schraeder 1
  • Crashes after a few seconds

    Crashes after a few seconds

    Which version are you using?

    v0.20.2

    Which operating system are you using?

    Raspberry Pi OS Linux raspberrypi4 5.15.61-v8+

    • [ ] Linux amd64 standard
    • [ ] Linux amd64 Docker
    • [x] Linux arm64 standard
    • [ ] Linux arm64 Docker
    • [ ] Linux arm7 standard
    • [ ] Linux arm7 Docker
    • [ ] Linux arm6 standard
    • [ ] Linux arm6 Docker
    • [ ] Windows amd64 standard
    • [ ] Windows amd64 Docker (WSL backend)
    • [ ] macOS amd64 standard
    • [ ] macOS amd64 Docker
    • [ ] Other (please describe)

    Describe the issue

    I'm converting an mjpeg stream to h264 (h264_v4l2m2m encoder) as a Raspberry Pi 4 using ffmpeg (ffmpeg version 4.3.4-0+deb11u1+rpt3) and publishing to rtsp-simple-server using the command:

    INPUT_OPTS=" -loglevel debug -reconnect_delay_max 10 -hide_banner -avoid_negative_ts make_zero -fflags nobuffer -flags low_delay -strict experimental -fflags +genpts+discardcorrupt -use_wallclock_as_timestamps 1 -f mjpeg -re"
    OUTPUT_OPTS="-an -c:v h264_v4l2m2m -g 20 -pix_fmt yuv420p -vsync 2 -b:v 900K -maxrate 1M  -f rtsp -rtsp_transport tcp -force_key_frames 'expr:gte(t,n_forced*2)' -r 10"
    
    ffmpeg $INPUT_OPTS -i $URL $OUTPUT_OPTS  $RTSP_SERVER"
    

    It seems to run for a few seconds but then rtsp-simple-server crashes with the log below.

    Describe how to replicate the issue

    Run command above

    Did you attach the server logs?

    yes

    panic: runtime error: index out of range [0] with length 0
    
    goroutine 89 [running]:
    github.com/aler9/rtsp-simple-server/internal/hls.(*muxerVariantMPEGTSSegmenter).writeH264(0x400046f698?, {0x99510?, 0xc79c39bbfb7?, 0xcb8260?}, 0x116a9bed0068dc80?, {0x40004e2d70?, 0x400046f6a8?, 0x5bd15c?})
            /s/internal/hls/muxer_variant_mpegts_segmenter.go:64 +0x47c
    github.com/aler9/rtsp-simple-server/internal/hls.(*muxerVariantMPEGTS).writeH264(0x4000122b00?, {0x0?, 0x0?, 0xcb8260?}, 0x0?, {0x40004e2d70?, 0x0?, 0x0?})
            /s/internal/hls/muxer_variant_mpegts.go:43 +0x2c
    github.com/aler9/rtsp-simple-server/internal/hls.(*Muxer).WriteH264(...)
            /s/internal/hls/muxer.go:81
    github.com/aler9/rtsp-simple-server/internal/core.(*hlsMuxer).runWriter(0x40003f9680, 0x4000248780, 0x0, 0x0, 0xffffffffffffffff)
            /s/internal/core/hls_muxer.go:419 +0x180
    github.com/aler9/rtsp-simple-server/internal/core.(*hlsMuxer).runInner.func2()
            /s/internal/core/hls_muxer.go:356 +0x3c
    created by github.com/aler9/rtsp-simple-server/internal/core.(*hlsMuxer).runInner
            /s/internal/core/hls_muxer.go:355 +0x5dc
    

    Did you attach a network dump?

    no

    EDIT: this issue actually only happens with the stock ffmpeg 4.3.4, using a new build of ffmpeg 5 from here (e.g. ffmpeg version N-60837-ge81242bb13-static), things seems to be fine.

    opened by roger- 0
Releases(v0.20.2)
Owner
Alessandro Ros
Software and robotics engineer, i deal with ML-based services, unmanned vehicles and anything that can be modeled and controlled. MSc @ PoliMi
Alessandro Ros
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