_rtsp-simple-server_ is a ready-to-use and zero-dependency server and proxy that allows users to publish, read and proxy live video and audio streams through various protocols: |protocol|description|publish|read|proxy| |--------|-----------|-------|----|-----| |RTSP|fastest way to publish and read streams|:heavy_check_mark:|:heavy_check_mark:|:heavy_check_mark:| |RTMP|allows to interact with legacy software|:heavy_check_mark:|:heavy_check_mark:|:heavy_check_mark:| |HLS|allows to embed streams into a web page|:x:|:heavy_check_mark:|:heavy_check_mark:| Features: * Publish live streams to the server * Read live streams from the server * Act as a proxy and serve streams from other servers or cameras, always or on-demand * Each stream can have multiple video and audio tracks, encoded with any codec, including H264, H265, VP8, VP9, MPEG2, MP3, AAC, Opus, PCM, JPEG * Streams are automatically converted from a protocol to another. For instance, it's possible to publish a stream with RTSP and read it with HLS Plus: * Serve multiple streams at once in separate paths * Authenticate readers and publishers * Query and control the server through an HTTP API * 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 [](https://github.com/aler9/rtsp-simple-server/actions?query=workflow:test) [](https://github.com/aler9/rtsp-simple-server/actions?query=workflow:lint) [](https://codecov.io/gh/aler9/rtsp-simple-server/branch/main) [](https://github.com/aler9/rtsp-simple-server/releases) [](https://hub.docker.com/r/aler9/rtsp-simple-server) [](https://aler9.github.io/rtsp-simple-server) ## Table of contents * [Installation](#installation) * [Standard](#standard) * [Docker](#docker) * [Basic usage](#basic-usage) * [Advanced usage and FAQs](#advanced-usage-and-faqs) * [Configuration](#configuration) * [Encryption](#encryption) * [Authentication](#authentication) * [Encrypt the configuration](#encrypt-the-configuration) * [Proxy mode](#proxy-mode) * [RTMP protocol](#rtmp-protocol) * [HLS protocol](#hls-protocol) * [Publish from OBS Studio](#publish-from-obs-studio) * [Publish a webcam](#publish-a-webcam) * [Publish a Raspberry Pi Camera](#publish-a-raspberry-pi-camera) * [Remuxing, re-encoding, compression](#remuxing-re-encoding-compression) * [Save published videos to disk](#save-published-videos-to-disk) * [On-demand publishing](#on-demand-publishing) * [Redirect to another server](#redirect-to-another-server) * [Fallback stream](#fallback-stream) * [Start on boot with systemd](#start-on-boot-with-systemd) * [Corrupted frames](#corrupted-frames) * [HTTP API](#http-api) * [Metrics](#metrics) * [pprof](#pprof) * [Command-line usage](#command-line-usage) * [Compile and run from source](#compile-and-run-from-source) * [Links](#links) ## Installation ### Standard 1. Download and extract a precompiled binary from the [release page](https://github.com/aler9/rtsp-simple-server/releases). 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 ``` Please keep in mind that the Docker image doesn't include _FFmpeg_. if you need to use _FFmpeg_ for a custom command or anything else, you need to build a Docker image that contains both _rtsp-simple-server_ and _FFmpeg_, by following instructions [here](https://github.com/aler9/rtsp-simple-server/issues/183#issuecomment-760856015). ## 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-play-1.0 rtsp://localhost:8554/mystream ``` 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](rtsp-simple-server.yml). There are 3 ways to change the configuration: 1. 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 ``` 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. 2. By overriding configuration parameters with environment variables, in the format `RTSP_PARAMNAME`, where `PARAMNAME` is the uppercase name of a parameter. For instance, the `rtspAddress` parameter can be overridden in the following way: ``` RTSP_RTSPADDRESS="127.0.0.1:8554" ./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 ``` 3. By using the [HTTP API](#http-api). ### 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: ```yml 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: ```yml 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: ```yml 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: ```yml 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](https://play.golang.org/p/rX29jwObNe4). 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++](https://nacl.cr.yp.to/secretbox.html), [Go](https://pkg.go.dev/golang.org/x/crypto/nacl/secretbox), [C#](https://github.com/somdoron/NaCl.net) 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 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: ```yml 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: ```yml 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: ```yml 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 ``` ### HLS protocol HLS is a media format that allows to embed live streams into web pages. Every stream published to the server can be accessed with a web browser by visiting: ``` http://localhost:8888/mystream ``` where `mystream` is the name of a stream that is being published. The direct HLS URL, that can be used to read the stream with players (VLC) or Javascript libraries (hls.js) can be obtained by appending `/index.m3u8`: ``` http://localhost:8888/mystream/index.m3u8 ``` Please note that most browsers don't support HLS directly (except Safari); a Javascript library, like [hls.js](https://github.com/video-dev/hls.js), must be used to load the stream. ### 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: ```yml paths: cam: runOnInit: ffmpeg -f v4l2 -i /dev/video0 -c:v libx264 -preset ultrafast -tune zerolatency -b:v 600k -f rtsp rtsp://localhost:$RTSP_PORT/$RTSP_PATH runOnInitRestart: yes ``` If the platform is Windows: ```yml paths: cam: runOnInit: ffmpeg -f dshow -i video="USB2.0 HD UVC WebCam" -c:v libx264 -preset ultrafast -tune zerolatency -b:v 600k -f rtsp rtsp://localhost:$RTSP_PORT/$RTSP_PATH runOnInitRestart: yes ``` Where `USB2.0 HD UVC WebCam` is the name of your webcam, that can be obtained with: ``` ffmpeg -list_devices true -f dshow -i dummy ``` After starting the server, the webcam can be reached on `rtsp://localhost:8554/cam`. ### Publish a Raspberry Pi Camera Install dependencies: 1. Gstreamer ``` sudo apt install -y gstreamer1.0-tools gstreamer1.0-rtsp ``` 2. gst-rpicamsrc, by following [instruction here](https://github.com/thaytan/gst-rpicamsrc) Then edit `rtsp-simple-server.yml` and replace everything inside section `paths` with the following content: ```yml paths: cam: runOnInit: gst-launch-1.0 rpicamsrc preview=false bitrate=2000000 keyframe-interval=50 ! video/x-h264,width=1920,height=1080,framerate=25/1 ! h264parse ! rtspclientsink location=rtsp://localhost:$RTSP_PORT/$RTSP_PATH runOnInitRestart: yes ``` After starting the server, the camera is available on `rtsp://localhost:8554/cam`. ### Remuxing, re-encoding, compression To change the format, codec or compression of a stream, use _FFmpeg_ or _Gstreamer_ together with _rtsp-simple-server_. For instance, to re-encode an existing stream, that is available in the `/original` path, and publish the resulting stream in the `/compressed` path, edit `rtsp-simple-server.yml` and replace everything inside section `paths` with the following content: ```yml paths: all: original: runOnPublish: ffmpeg -i rtsp://localhost:$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 ``` ### Save published videos to disk To Save published videos to disk, it's enough to put _FFmpeg_ inside `runOnPublish`: ```yml paths: all: original: runOnPublish: ffmpeg -i rtsp://localhost:$RTSP_PORT/$RTSP_PATH -c copy -f segment -strftime 1 -segment_time 60 -segment_format mp4 saved_%Y-%m-%d_%H-%M-%S.mp4 runOnPublishRestart: yes ``` ### On-demand publishing Edit `rtsp-simple-server.yml` and replace everything inside section `paths` with the following content: ```yml 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: ```yml 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: ```yml 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](https://github.com/aler9/rtsp-simple-server/releases), 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 ``` ### Corrupted frames In some scenarios, the server can send incomplete or corrupted frames. This can be caused by multiple reasons: * the packet buffer of the server is too small and can't handle the stream throughput. A solution consists in increasing its size: ```yml readBufferCount: 1024 ``` * The stream throughput is too big and the stream can't be sent correctly with the UDP stream protocol. UDP is more performant, faster and more efficient than TCP, but doesn't have a retransmission mechanism, that is needed in case of streams that need a large bandwidth. A solution consists in switching to TCP: ```yml protocols: [tcp] ``` In case the source is a camera: ```yml paths: test: source: rtsp://.. sourceProtocol: tcp ``` * the software that is generating the stream (a camera or FFmpeg) is generating non-conformant RTP packets, with a payload bigger than the maximum allowed (that is 1460 due to the UDP MTU). A solution consists in increasing the buffer size: ```yml readBufferSize: 8192 ``` ### HTTP API The server can be queried and controlled with an HTTP API, that must be enabled by setting the `api` parameter in the configuration: ```yml api: yes ``` The API listens on `apiAddress`, that by default is `127.0.0.1:9997`; for instance, to obtain a list of active paths, run: ``` curl http://127.0.0.1:9997/v1/paths/list ``` Full documentation of the API is available on the [dedicated site](https://aler9.github.io/rtsp-simple-server/). ### Metrics 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: ``` paths{state="ready"} 2 1628760831152 paths{state="notReady"} 0 1628760831152 rtsp_sessions{state="idle"} 0 1628760831152 rtsp_sessions{state="read"} 0 1628760831152 rtsp_sessions{state="publish"} 1 1628760831152 rtsps_sessions{state="idle"} 0 1628760831152 rtsps_sessions{state="read"} 0 1628760831152 rtsps_sessions{state="publish"} 0 1628760831152 rtmp_conns{state="idle"} 0 1628760831152 rtmp_conns{state="read"} 0 1628760831152 rtmp_conns{state="publish"} 1 1628760831152 ``` where: * `paths{state="ready"}` is the count of paths that are ready * `paths{state="notReady"}` is the count of paths that are not ready * `rtsp_sessions{state="idle"}` is the count of RTSP sessions that are idle * `rtsp_sessions{state="read"}` is the count of RTSP sessions that are reading * `rtsp_sessions{state="publish"}` is the counf ot RTSP sessions that are publishing * `rtsps_sessions{state="idle"}` is the count of RTSPS sessions that are idle * `rtsps_sessions{state="read"}` is the count of RTSPS sessions that are reading * `rtsps_sessions{state="publish"}` is the counf ot RTSPS sessions that are publishing * `rtmp_conns{state="idle"}` is the count of RTMP connections that are idle * `rtmp_conns{state="read"}` is the count of RTMP connections that are reading * `rtmp_conns{state="publish"}` is the count of RTMP connections that are publishing ### pprof 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 1.16, 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 * https://github.com/aler9/gortsplib (RTSP library used internally) * https://github.com/pion/sdp (SDP library used internally) * https://github.com/pion/rtcp (RTCP library used internally) * https://github.com/pion/rtp (RTP library used internally) * https://github.com/notedit/rtmp (RTMP library used internally) * https://github.com/flaviostutz/rtsp-relay IETF Standards * RTSP 1.0 https://tools.ietf.org/html/rfc2326 * RTSP 2.0 https://tools.ietf.org/html/rfc7826 * HTTP 1.1 https://tools.ietf.org/html/rfc2616 Conventions * https://github.com/golang-standards/project-layout