rtspss

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Published: Mar 30, 2021 License: MIT Imports: 16 Imported by: 0

README

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 TLS certificate must be installed on the server; if the server is installed on a machine that is publicly accessible from the internet, a certificate can be requested from a Certificate authority by using tools like Certbot; otherwise, a self-signed certificate 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 and the server certificate is self signed, remember to 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:mypass@localhost: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:pass@host: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, use the following parameters:

  • Server: rtmp://localhost/mystream
  • Stream key: (leave empty)

If credentials are in use, use the following parameters:

  • Server: rtmp://localhost/mystream?user=myuser&pass=mypass
  • Stream key: (leave empty)

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 ! rtspclientsink location=rtsp://localhost:$RTSP_PORT/$RTSP_PATH
    runOnInitRestart: yes

After starting the server, the webcam 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 <video> 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 -re -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 -re -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
    

    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 [<flags>]

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:
  [<confpath>]  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

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