mediamtx/README.md at ac10102329c7fe2e399756552008ecfdf3857bf0

mirror of https://github.com/bluenviron/mediamtx synced 2024-12-14 18:54:54 +00:00

automatically set 'record: yes' when not specified (#2366 )

2023-09-18 22:34:38 +02:00

58 KiB

Raw Blame History

MediaMTX (formerly rtsp-simple-server) is a ready-to-use and zero-dependency real-time media server and media proxy that allows to publish, read, proxy and record video and audio streams. It has been conceived as a "media router" that routes media streams from one end to the other.

Live streams can be published to the server with:

protocol	variants	video codecs	audio codecs
SRT clients		H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video	Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)
SRT cameras and servers		H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video	Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)
WebRTC clients	Browser-based, WHIP	AV1, VP9, VP8, H264	Opus, G722, G711
WebRTC servers	WHEP	AV1, VP9, VP8, H264	Opus, G722, G711
RTSP clients	UDP, TCP, RTSPS	AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec	Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), G726, G722, G711, LPCM and any RTP-compatible codec
RTSP cameras and servers	UDP, UDP-Multicast, TCP, RTSPS	AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec	Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), G726, G722, G711, LPCM and any RTP-compatible codec
RTMP clients	RTMP, RTMPS, Enhanced RTMP	AV1, VP9, H265, H264	MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)
RTMP cameras and servers	RTMP, RTMPS, Enhanced RTMP	H264	MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)
HLS cameras and servers	Low-Latency HLS, MP4-based HLS, legacy HLS	AV1, VP9, H265, H264	Opus, MPEG-4 Audio (AAC)
UDP/MPEG-TS	Unicast, broadcast, multicast	H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video	Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)
Raspberry Pi Cameras		H264

And can be read from the server with:

protocol	variants	video codecs	audio codecs
SRT		H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video	Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)
WebRTC	Browser-based, WHEP	AV1, VP9, VP8, H264	Opus, G722, G711
RTSP	UDP, UDP-Multicast, TCP, RTSPS	AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec	Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), G726, G722, G711, LPCM and any RTP-compatible codec
RTMP	RTMP, RTMPS, Enhanced RTMP	H264	MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)
HLS	Low-Latency HLS, MP4-based HLS, legacy HLS	AV1, VP9, H265, H264	Opus, MPEG-4 Audio (AAC)

Features

Publish live streams to the server
Read live streams from the server
Streams are automatically converted from a protocol to another
Serve multiple streams at once in separate paths
Record streams to disk
Authenticate users; use internal or external authentication
Redirect readers to other RTSP servers (load balancing)
Query and control the server through the API
Reload the configuration without disconnecting existing clients (hot reloading)
Read Prometheus-compatible metrics
Run external commands (hooks) when clients connect, disconnect, read or publish streams
Compatible with Linux, Windows and macOS, does not require any dependency or interpreter, it's a single executable

Note about rtsp-simple-server

rtsp-simple-server has been rebranded as MediaMTX. The reason is pretty obvious: this project started as a RTSP server but has evolved into a much more versatile product that is not tied to the RTSP protocol anymore. Nothing will change regarding license, features and backward compatibility.

Installation
Basic usage
Publish to the server
Read from the server
- By software
  - FFmpeg
  - GStreamer
  - VLC
  - Web browsers
- By protocol
  - SRT
  - WebRTC
  - RTSP
  - RTMP
  - HLS
Other features
Compile from source
Specifications
Related projects

Installation

There are several installation methods available: standalone binary, Docker image, Arch Linux package and OpenWRT package.

Standalone binary

Download and extract a standalone binary from the release page.
Start the server:
```
./mediamtx
```

Docker image

Download and launch the image:

docker run --rm -it --network=host bluenviron/mediamtx:latest

Available images:

name	FFmpeg included	RPI Camera support
bluenviron/mediamtx:latest	❌	❌
bluenviron/mediamtx:latest-ffmpeg	✔️	❌
bluenviron/mediamtx:latest-rpi	❌	✔️
bluenviron/mediamtx:latest-ffmpeg-rpi	✔️	✔️

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

docker run --rm -it \
-e MTX_PROTOCOLS=tcp \
-p 8554:8554 \
-p 1935:1935 \
-p 8888:8888 \
-p 8889:8889 \
-p 8890:8890/udp \
bluenviron/mediamtx

Arch Linux package

If you are running the Arch Linux distribution, run:

git clone https://aur.archlinux.org/mediamtx.git
cd mediamtx
makepkg -si

OpenWRT package

In a x86 Linux system, download the OpenWRT SDK corresponding to the wanted OpenWRT version and target from the OpenWRT website and extract it.

Open a terminal in the SDK folder and setup the SDK:

./scripts/feeds update -a
./scripts/feeds install -a
make defconfig

Download the server Makefile and set the server version inside the file:

mkdir package/mediamtx
wget -O package/mediamtx/Makefile https://raw.githubusercontent.com/bluenviron/mediamtx/main/openwrt.mk
sed -i "s/v0.0.0/$(git ls-remote --tags --sort=v:refname https://github.com/bluenviron/mediamtx | tail -n1 | sed 's/.*\///; s/\^{}//')/" package/mediamtx/Makefile

Compile the server:

make package/mediamtx/compile -j$(nproc)

Transfer the .ipk file from bin/packages/*/base to the OpenWRT system
Install it with:
```
opkg install [ipk-file-name].ipk
```

Basic usage

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

Open the stream. For instance, you can open the stream with VLC:

vlc --network-caching=50 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

Publish to the server

By software

FFmpeg

FFmpeg can publish a stream to the server in multiple ways (SRT client, SRT server, RTSP client, RTMP client, UDP/MPEG-TS, WebRTC with WHIP). The recommended one consists in publishing as a RTSP client:

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

The RTSP protocol supports multiple underlying transport protocols, each with its own characteristics (see RTSP-specific features). You can set the transport protocol by using the rtsp_transport flag, for instance, in order to use TCP:

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

The resulting stream will be available in path /mystream.

GStreamer

GStreamer can publish a stream to the server in multiple ways (SRT client, SRT server, RTSP client, RTMP client, UDP/MPEG-TS, WebRTC with WHIP). The recommended one consists in publishing as a RTSP client:

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

If the stream is video only:

gst-launch-1.0 filesrc location=file.mp4 ! qtdemux name=d \
d.video_0 ! rtspclientsink location=rtsp://localhost:8554/mystream

The RTSP protocol supports multiple underlying transport protocols, each with its own characteristics (see RTSP-specific features). You can set the transport protocol by using the protocols flag:

gst-launch-1.0 filesrc location=file.mp4 ! qtdemux name=d \
d.video_0 ! rtspclientsink protocols=tcp name=s location=rtsp://localhost:8554/mystream

The resulting stream will be available in path /mystream.

OBS Studio

OBS Studio can publish to the server in multiple ways (SRT client, RTMP client, WebRTC client). The recommended one consists in publishing as a RTMP client. 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

Save the configuration and click Start streaming.

If you want to generate a stream that can be read with WebRTC, open Settings -> Output -> Recording and use the following parameters:

FFmpeg output type: Output to URL
File path or URL: rtsp://localhost:8554/mystream
Container format: rtsp
Check show all codecs (even if potentically incompatible
Video encoder: h264_nvenc (libx264)
Video encoder settings (if any): bf=0
Audio track: 1
Audio encoder: libopus

Then use the button Start Recording (instead of Start Streaming) to start streaming.

Latest versions of OBS Studio can publish to the server with the WebRTC / WHIP protocol. Use the following parameters:

Service: WHIP
Server: http://localhost:8889/mystream/whip

Save the configuration and click Start streaming.

The resulting stream will be available in path /mystream.

OpenCV

OpenCV can publish to the server through its GStreamer plugin, as a RTSP client. It must be compiled with GStreamer support, by following this procedure:

sudo apt install -y libgstreamer1.0-dev libgstreamer-plugins-base1.0-dev gstreamer1.0-plugins-ugly gstreamer1.0-rtsp python3-dev python3-numpy
git clone --depth=1 -b 4.5.4 https://github.com/opencv/opencv
cd opencv
mkdir build && cd build
cmake -D CMAKE_INSTALL_PREFIX=/usr -D WITH_GSTREAMER=ON ..
make -j$(nproc)
sudo make install

You can check that OpenCV has been installed correctly by running:

python3 -c 'import cv2; print(cv2.getBuildInformation())'

Check that the output contains GStreamer: YES.

Videos can be published with VideoWriter:

from datetime import datetime
from time import sleep, time

import cv2
import numpy as np

fps = 15
width = 800
height = 600
colors = [
    (0, 0, 255),
    (255, 0, 0),
    (0, 255, 0),
]

out = cv2.VideoWriter('appsrc ! videoconvert' + \
    ' ! video/x-raw,format=I420' + \
    ' ! x264enc speed-preset=ultrafast bitrate=600 key-int-max=' + str(fps * 2) + \
    ' ! video/x-h264,profile=baseline' + \
    ' ! rtspclientsink location=rtsp://localhost:8554/mystream',
    cv2.CAP_GSTREAMER, 0, fps, (width, height), True)
if not out.isOpened():
    raise Exception("can't open video writer")

curcolor = 0
start = time()

while True:
    frame = np.zeros((height, width, 3), np.uint8)

    # create a rectangle
    color = colors[curcolor]
    curcolor += 1
    curcolor %= len(colors)
    for y in range(0, int(frame.shape[0] / 2)):
        for x in range(0, int(frame.shape[1] / 2)):
            frame[y][x] = color

    out.write(frame)
    print("%s frame written to the server" % datetime.now())

    now = time()
    diff = (1 / fps) - now - start
    if diff > 0:
        sleep(diff)
    start = now

The resulting stream will be available in path /mystream.

Web browsers

Web browsers can publish a stream to the server by using the WebRTC protocol. Start the server and open the web page:

http://localhost:8889/mystream/publish

The resulting stream will be available in path /mystream.

This web page can be embedded into another web page by using an iframe:

<iframe src="http://mediamtx-ip:8889/mystream/publish" scrolling="no"></iframe>

For more advanced setups, you can create and serve a custom web page by starting from the source code of the publish page.

By device

Generic webcam

If the OS is Linux-based, edit mediamtx.yml and replace everything inside section paths with the following content:

paths:
  cam:
    runOnInit: ffmpeg -f v4l2 -i /dev/video0 -pix_fmt yuv420p -preset ultrafast -b:v 600k -f rtsp rtsp://localhost:$RTSP_PORT/$MTX_PATH
    runOnInitRestart: yes

If the OS is Windows:

paths:
  cam:
    runOnInit: ffmpeg -f dshow -i video="USB2.0 HD UVC WebCam" -pix_fmt yuv420p -c:v libx264 -preset ultrafast -b:v 600k -f rtsp rtsp://localhost:$RTSP_PORT/$MTX_PATH
    runOnInitRestart: yes

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

ffmpeg -list_devices true -f dshow -i dummy

The resulting stream will be available in path /cam.

Raspberry Pi Cameras

MediaMTX natively supports the Raspberry Pi Camera, enabling high-quality and low-latency video streaming from the camera to any user, for any purpose. There are a couple of requirements:

The server must run on a Raspberry Pi, with Raspberry Pi OS bullseye or newer as operative system. Both 32 bit and 64 bit operative systems are supported.
Make sure that the legacy camera stack is disabled. Type sudo raspi-config, then go to Interfacing options, enable/disable legacy camera support, choose no. Reboot the system.

If you want to run the standard (non-Docker) version of the server:

Make sure that the following packages are installed:
- libcamera0 (≥ 0.0.5)
- libfreetype6
download the server executable. If you're using 64-bit version of the operative system, make sure to pick the arm64 variant.
edit mediamtx.yml and replace everything inside section paths with the following content:
```
paths:
  cam:
    source: rpiCamera
```

The resulting stream will be available in path /cam.

If you want to run the server inside Docker, you need to use the latest-rpi image (that already contains required libraries) and launch the container with some additional flags:

docker run --rm -it \
--network=host \
--privileged \
--tmpfs /dev/shm:exec \
-v /run/udev:/run/udev:ro \
-e MTX_PATHS_CAM_SOURCE=rpiCamera \
bluenviron/mediamtx:latest-rpi

Camera settings can be changed by using the rpiCamera* parameters:

paths:
  cam:
    source: rpiCamera
    rpiCameraWidth: 1920
    rpiCameraHeight: 1080

All available parameters are listed in the sample configuration file.

In order to add audio from a USB microfone, install GStreamer and alsa-utils:

sudo apt install -y gstreamer1.0-tools gstreamer1.0-rtsp gstreamer1.0-alsa alsa-utils

list available audio cards with:

arecord -L

Sample output:

surround51:CARD=ICH5,DEV=0
    Intel ICH5, Intel ICH5
    5.1 Surround output to Front, Center, Rear and Subwoofer speakers
default:CARD=U0x46d0x809
    USB Device 0x46d:0x809, USB Audio
    Default Audio Device

Find the audio card of the microfone and take note of its name, for instance default:CARD=U0x46d0x809. Then use GStreamer inside runOnReady to read the video stream, add audio and publish the new stream to another path:

paths:
  cam:
    source: rpiCamera
    runOnReady: >
      gst-launch-1.0
      rtspclientsink name=s location=rtsp://localhost:$RTSP_PORT/cam_with_audio
      rtspsrc location=rtsp://127.0.0.1:$RTSP_PORT/$MTX_PATH latency=0 ! rtph264depay ! s.
      alsasrc device=default:CARD=U0x46d0x809 ! opusenc bitrate=16000 ! s.      
    runOnReadyRestart: yes
  cam_with_audio:

The resulting stream will be available in path /cam_with_audio.

By protocol

SRT clients

SRT is a protocol that allows to publish and read live data stream, providing encryption, integrity and a retransmission mechanism. It is usually used to transfer media streams encoded with MPEG-TS. In order to publish a stream to the server with the SRT protocol, use this URL:

srt://localhost:8890?streamid=publish:mystream&pkt_size=1316

Replace mystream with any name you want. The resulting stream will be available in path /mystream.

If credentials are enabled, append username and password to streamid;

srt://localhost:8890?streamid=publish:mystream:user:pass&pkt_size=1316

If you want to publish a stream by using a client in listening mode (i.e. with mode=listener appended to the URL), read the next section.

Known clients that can publish with SRT are FFmpeg, GStreamer, OBS Studio.

SRT cameras and servers

In order to ingest into the server a SRT stream from an existing server, camera or client in listening mode (i.e. with mode=listener appended to the URL), add the corresponding URL into the source parameter of a path:

paths:
  proxied:
    # url of the source stream, in the format srt://host:port?streamid=streamid&other_parameters
    source: srt://original-url

WebRTC clients

WebRTC is an API that makes use of a set of protocols and methods to connect two clients together and allow them to exchange real-time media or data streams. You can publish a stream with WebRTC and a web browser by visiting:

http://localhost:8889/mystream/publish

The resulting stream will be available in path /mystream.

WHIP is a WebRTC extensions that allows to publish streams by using a URL, without passing through a web page. This allows to use WebRTC as a general purpose streaming protocol. If you are using a software that supports WHIP (for instance, latest versions of OBS Studio), you can publish a stream to the server by using this URL:

http://localhost:8889/mystream/whip

Depending on the network it may be difficult to establish a connection between server and clients, see WebRTC-specific features for remediations.

Known clients that can publish with WebRTC and WHIP are FFmpeg, GStreamer, OBS Studio.

WebRTC servers

In order to ingest into the server a WebRTC stream from an existing server, add the corresponding WHEP URL into the source parameter of a path:

paths:
  proxied:
    # url of the source stream, in the format whep://host:port/path (HTTP) or wheps:// (HTTPS)
    source: wheps://host:port/path

RTSP clients

RTSP is a protocol that allows to publish and read streams. It supports different underlying transport protocols and allows to encrypt streams in transit (see RTSP-specific features). In order to publish a stream to the server with the RTSP protocol, use this URL:

rtsp://localhost:8554/mystream

The resulting stream will be available in path /mystream.

Known clients that can publish with RTSP are FFmpeg, GStreamer, OBS Studio.

RTSP cameras and servers

Most IP cameras expose their video stream by using a RTSP server that is embedded into the camera itself. In particular, cameras that are compliant to ONVIF profile S or T meet this requirement. You can use MediaMTX to connect to one or multiple existing RTSP servers and read their video streams:

paths:
  proxied:
    # url of the source stream, in the format rtsp://user:pass@host:port/path
    source: rtsp://original-url

The resulting stream will be available in path /proxied.

The server supports any number of source streams (count is just limited by hardware capability) it's enough to add additional entries to the paths section:

paths:
  proxied1:
    source: rtsp://url1

  proxied2:
    source: rtsp://url1

RTMP clients

RTMP is a protocol that allows to read and publish streams, but is less versatile and less efficient than RTSP and WebRTC (doesn't support UDP, doesn't support most RTSP codecs, doesn't support feedback mechanism). Streams can be published to the server by using the URL:

rtmp://localhost/mystream

The resulting stream will be available in path /mystream.

In case authentication is enabled, credentials can be passed to the server by using the user and pass query parameters:

rtmp://localhost/mystream?user=myuser&pass=mypass

Known clients that can publish with RTMP are FFmpeg, GStreamer, OBS Studio.

RTMP cameras and servers

You can use MediaMTX to connect to one or multiple existing RTMP servers and read their video streams:

paths:
  proxied:
    # url of the source stream, in the format rtmp://user:pass@host:port/path
    source: rtmp://original-url

The resulting stream will be available in path /proxied.

HLS cameras and servers

HLS is a streaming protocol that works by splitting streams into segments, and by serving these segments and a playlist with the HTTP protocol. You can use MediaMTX to connect to one or multiple existing HLS servers and read their video streams:

paths:
  proxied:
    # url of the playlist of the stream, in the format http://user:pass@host:port/path
    source: http://original-url/stream/index.m3u8

The resulting stream will be available in path /proxied.

UDP/MPEG-TS

The server supports ingesting UDP/MPEG-TS packets (i.e. MPEG-TS packets sent with UDP). Packets can be unicast, broadcast or multicast. For instance, you can generate a multicast UDP/MPEG-TS stream with GStreamer:

gst-launch-1.0 -v mpegtsmux name=mux alignment=1 ! udpsink host=238.0.0.1 port=1234 \
videotestsrc ! video/x-raw,width=1280,height=720,format=I420 ! x264enc speed-preset=ultrafast bitrate=3000 key-int-max=60 ! video/x-h264,profile=high ! mux. \
audiotestsrc ! audioconvert ! avenc_aac ! mux.

or FFmpeg:

ffmpeg -re -f lavfi -i testsrc=size=1280x720:rate=30 \
-pix_fmt yuv420p -c:v libx264 -preset ultrafast -b:v 600k \
-f mpegts udp://238.0.0.1:1234?pkt_size=1316

Edit mediamtx.yml and replace everything inside section paths with the following content:

paths:
  mypath:
    source: udp://238.0.0.1:1234

The resulting stream will be available in path /mypath.

Known clients that can publish with WebRTC and WHIP are FFmpeg and GStreamer.

Read from the server

By software

FFmpeg

FFmpeg can read a stream from the server in multiple ways (RTSP, RTMP, HLS, WebRTC with WHEP, SRT). The recommended one consists in reading with RTSP:

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

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

GStreamer

GStreamer can read a stream from the server in multiple ways (RTSP, RTMP, HLS, WebRTC with WHEP, SRT). The recommended one consists in reading with RTSP:

gst-launch-1.0 rtspsrc location=rtsp://127.0.0.1:8554/mystream latency=0 ! decodebin ! autovideosink

The RTSP protocol supports multiple underlying transport protocols, each with its own characteristics (see RTSP-specific features). You can change the transport protocol by using the protocols flag:

gst-launch-1.0 rtspsrc protocols=tcp location=rtsp://127.0.0.1:8554/mystream latency=0 ! decodebin ! autovideosink

If encryption is enabled, set tls-validation-flags to 0:

gst-launch-1.0 rtspsrc tls-validation-flags=0 location=rtsps://ip:8322/...

VLC

VLC can read a stream from the server in multiple ways (RTSP, RTMP, HLS, SRT). The recommended one consists in reading with RTSP:

vlc --network-caching=50 rtsp://localhost:8554/mystream

The RTSP protocol supports multiple underlying transport protocols, each with its own characteristics (see RTSP-specific features).

In order to use the TCP transport protocol, use the --rtsp_tcp flag:

vlc --network-caching=50 --rtsp-tcp rtsp://localhost:8554/mystream

In order to use the UDP-multicast transport protocol, append ?vlcmulticast to the URL:

vlc --network-caching=50 rtsp://localhost:8554/mystream?vlcmulticast

You can change the transport protocol by using the --rtsp_ flag:

Ubuntu bug

The VLC shipped with Ubuntu 21.10 doesn't support playing RTSP due to a license issue (see here and here). To fix the issue, remove the default VLC instance and install the snap version:

sudo apt purge -y vlc
snap install vlc

Encrypted streams

At the moment VLC doesn't support reading encrypted RTSP streams. However, you can use a proxy like stunnel or nginx or a dedicated MediaMTX instance to decrypt streams before reading them.

Web browsers

Web browsers can read a stream from the server in multiple ways (WebRTC or HLS).

You can read a stream by using the WebRTC protocol by visiting the web page:

http://localhost:8889/mystream

This web page can be embedded into another web page by using an iframe:

<iframe src="http://mediamtx-ip:8889/mystream" scrolling="no"></iframe>

For more advanced setups, you can create and serve a custom web page by starting from the source code of the read page.

Web browsers can also read a stream with the HLS protocol. Latency is higher but there are less problems related to connectivity between server and clients, furthermore the server load can be balanced by using a common HTTP CDN (like CloudFront or Cloudflare), and this allows to handle readers in the order of millions. Visit the web page:

http://localhost:8888/mystream

This web page can be embedded into another web page by using an iframe:

<iframe src="http://mediamtx-ip:8888/mystream" scrolling="no"></iframe>

By protocol

SRT

SRT is a protocol that allows to publish and read live data stream, providing encryption, integrity and a retransmission mechanism. It is usually used to transfer media streams encoded with MPEG-TS. In order to read a stream from the server with the SRT protocol, use this URL:

srt://localhost:8890?streamid=read:mystream

Replace mystream with the path name.

If credentials are enabled, append username and password to streamid;

srt://localhost:8890?streamid=publish:mystream:user:pass

Known clients that can read with SRT are FFmpeg, GStreamer and VLC.

WebRTC

WebRTC is an API that makes use of a set of protocols and methods to connect two clients together and allow them to exchange real-time media or data streams. You can read a stream with WebRTC and a web browser by visiting:

http://localhost:8889/mystream

WHEP is a WebRTC extensions that allows to read streams by using a URL, without passing through a web page. This allows to use WebRTC as a general purpose streaming protocol. If you are using a software that supports WHEP, you can read a stream from the server by using this URL:

http://localhost:8889/mystream/whep

Depending on the network it may be difficult to establish a connection between server and clients, see WebRTC-specific features for remediations.

Known clients that can read with WebRTC and WHEP are FFmpeg, GStreamer and web browsers.

RTSP

RTSP is a protocol that allows to publish and read streams. It supports different underlying transport protocols and allows to encrypt streams in transit (see RTSP-specific features). In order to read a stream with the RTSP protocol, use this URL:

rtsp://localhost:8554/mystream

Known clients that can read with RTSP are FFmpeg, GStreamer and VLC.

Latency

The RTSP protocol doesn't introduce any latency by itself. Latency is usually introduced by clients, that put frames in a buffer to compensate network fluctuations. In order to decrease latency, the best way consists in tuning the client. For instance, latency can be decreased with VLC by decreasing the Network caching parameter, that is available in the Open network stream dialog or alternatively ca be set with the command line:

vlc --network-caching=50 rtsp://...

RTMP

RTMP is a protocol that allows to read and publish streams, but is less versatile and less efficient than RTSP and WebRTC ((doesn't support UDP, doesn't support most RTSP codecs, doesn't support feedback mechanism)). Streams can be read from the server by using the URL:

rtmp://localhost/mystream

In case authentication is enabled, credentials can be passed to the server by using the user and pass query parameters:

rtmp://localhost/mystream?user=myuser&pass=mypass

Known clients that can read with RTMP are FFmpeg, GStreamer and VLC.

HLS

HLS is a protocol that works by splitting streams into segments, and by serving these segments and a playlist with the HTTP protocol. You can use MediaMTX to generate a HLS stream, that is accessible through a web page:

http://localhost:8888/mystream

and can also be accessed without using the browsers, by software that supports the HLS protocol (for instance VLC or MediaMTX itself) by using this URL:

http://localhost:8888/mystream/index.m3u8

Although the server can produce HLS with a variety of video and audio codecs (that are listed at the beginning of the README), not all browsers can read all codecs.

You can check what codecs your browser can read by using this tool.

If you want to support most browsers, you can to re-encode the stream by using the H264 and AAC codecs, for instance by using FFmpeg:

ffmpeg -i rtsp://original-source \
-pix_fmt yuv420p -c:v libx264 -preset ultrafast -b:v 600k \
-c:a aac -b:a 160k \
-f rtsp rtsp://localhost:8554/mystream

Known clients that can read with HLS are FFmpeg, GStreamer, VLC and web browsers.

LL-HLS

Low-Latency HLS is a recently standardized variant of the protocol that allows to greatly reduce playback latency. It works by splitting segments into parts, that are served before the segment is complete. LL-HLS is enabled by default. If the stream is not shown correctly, try tuning the hlsPartDuration parameter, for instance:

hlsPartDuration: 500ms

Compatibility with Apple devices

In order to correctly display Low-Latency HLS streams in Safari running on Apple devices (iOS or macOS), a 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

Set the hlsEncryption, hlsServerKey and hlsServerCert parameters in the configuration file:

hlsEncryption: yes
hlsServerKey: server.key
hlsServerCert: server.crt

Keep also in mind that not all H264 video streams can be played on Apple Devices due to some intrinsic properties (distance between I-Frames, profile). If the video can't be played correctly, you can either:

re-encode it by following instructions in this README
disable the Low-latency variant of HLS and go back to the legacy variant:
```
  hlsVariant: mpegts
```

Latency

in HLS, latency is introduced since a client must wait for the server to generate segments before downloading them. This latency amounts to 500ms-3s when the low-latency HLS variant is enabled (and it is by default), otherwise amounts to 1-15secs.

To decrease the latency, you can:

try decreasing the hlsPartDuration parameter
try decreasing the hlsSegmentDuration parameter
The segment duration is influenced by the interval between the IDR frames of the video track. An IDR frame is a frame that can be decoded independently from the others. The server changes the segment duration in order to include at least one IDR frame into each segment. Therefore, you need to decrease the interval between the IDR frames. This can be done in two ways:
- if the stream is being hardware-generated (i.e. by a camera), there's usually a setting called Key-Frame Interval in the camera configuration page
- otherwise, the stream must be re-encoded. It's possible to tune the IDR frame interval by using ffmpeg's -g option:
```
ffmpeg -i rtsp://original-stream -pix_fmt yuv420p -c:v libx264 -preset ultrafast -b:v 600k -max_muxing_queue_size 1024 -g 30 -f rtsp rtsp://localhost:$RTSP_PORT/compressed
```

Other features

Configuration

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

There are 3 ways to change the configuration:

By editing the mediamtx.yml file, that is
- included into the release bundle
- available in the root folder of the Docker image (/mediamtx.yml); it can be overridden in this way:
```
docker run --rm -it --network=host -v $PWD/mediamtx.yml:/mediamtx.yml bluenviron/mediamtx
```
The configuration can be changed dynamically 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.
By overriding configuration parameters with environment variables, in the format MTX_PARAMNAME, where PARAMNAME is the uppercase name of a parameter. For instance, the rtspAddress parameter can be overridden in the following way:
```
MTX_RTSPADDRESS="127.0.0.1:8554" ./mediamtx
```
Parameters that have array as value can be overriden by setting a comma-separated list. For example:
```
MTX_PROTOCOLS="tcp,udp"
```
Parameters in maps can be overridden by using underscores, in the following way:
```
MTX_PATHS_TEST_SOURCE=rtsp://myurl ./mediamtx
```
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 MTX_PATHS_TEST_SOURCE=rtsp://myurl bluenviron/mediamtx
```
By using the API.

Authentication

Edit mediamtx.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 in transit.

Authentication can be delegated to an external HTTP server:

externalAuthenticationURL: http://myauthserver/auth

Each time a user needs to be authenticated, the specified URL will be requested with the POST method and this payload:

{
  "ip": "ip",
  "user": "user",
  "password": "password",
  "path": "path",
  "protocol": "rtsp|rtmp|hls|webrtc",
  "id": "id",
  "action": "read|publish",
  "query": "query"
}

If the URL returns a status code that begins with 20 (i.e. 200), authentication is successful, otherwise it fails.

Please be aware that it's perfectly normal for the authentication server to receive requests with empty users and passwords, i.e.:

{
  "user": "",
  "password": "",
}

This happens because a RTSP client doesn't provide credentials until it is asked to. In order to receive the credentials, the authentication server must reply with status code 401, then the client will send 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:

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;
The string is prefixed with the nonce;
The string is encoded with base64.

After performing the encryption, put the base64-encoded result into the configuration file, and launch the server with the MTX_CONFKEY variable:

MTX_CONFKEY=mykey ./mediamtx

Remuxing, re-encoding, compression

To change the format, codec or compression of a stream, use FFmpeg or GStreamer together with MediaMTX. 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 mediamtx.yml and replace everything inside section paths with the following content:

paths:
  all:
  original:
    runOnReady: >
      ffmpeg -i rtsp://localhost:$RTSP_PORT/$MTX_PATH
        -pix_fmt yuv420p -c:v libx264 -preset ultrafast -b:v 600k
        -max_muxing_queue_size 1024 -f rtsp rtsp://localhost:$RTSP_PORT/compressed      
    runOnReadyRestart: yes

Record streams to disk

To save available streams to disk, set the record and the recordPath parameter in the configuration file:

# Record streams to disk.
record: yes
# 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

All available recording parameters are listed in the sample configuration file.

Currently the server supports recording tracks encoded with the following codecs:

Video: AV1, VP9, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video
Audio: Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)

Forward streams to another server

To forward incoming streams to another server, use FFmpeg inside the runOnReady parameter:

paths:
  all:
    runOnReady: >
      ffmpeg -i rtsp://localhost:$RTSP_PORT/$MTX_PATH
      -c copy
      -f rtsp rtsp://another-server/another-path      
    runOnReadyRestart: yes

On-demand publishing

Edit mediamtx.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:$RTSP_PORT/$MTX_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.

Start on boot

Linux*

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

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

sudo mv mediamtx /usr/local/bin/
sudo mv mediamtx.yml /usr/local/etc/

Create the service:

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

Enable and start the service:

sudo systemctl daemon-reload
sudo systemctl enable mediamtx
sudo systemctl start mediamtx

Windows*

Download the WinSW v2 executable and place it into the same folder of mediamtx.exe.

In the same folder, create a file named WinSW-x64.xml with this content:

<service>
  <id>mediamtx</id>
  <name>mediamtx</name>
  <description></description>
  <executable>%BASE%/mediamtx.exe</executable>
</service>

Open a terminal, navigate to the folder and run:

WinSW-x64 install

The server is now installed as a system service and will start at boot time.

Hooks

The server allows to specify commands that are executed when a certain event happens, allowing the propagation of events to external software.

runOnConnect allows to run a command 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: curl http://my-custom-server/webhook?conn_type=$MTX_CONN_TYPE&conn_id=$MTX_CONN_ID
# Restart the command if it exits.
runOnConnectRestart: no

runOnDisconnect allows to run a command when a client disconnects from the server:

# Environment variables are the same of runOnConnect.
runOnDisconnect: curl http://my-custom-server/webhook?conn_type=$MTX_CONN_TYPE&conn_id=$MTX_CONN_ID

runOnInit allows to run a command when a path is initialized. This can be used to publish a stream when the server is launched:

paths:
  mypath:
    # 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: ffmpeg -i my_file.mp4 -c copy -f rtsp rtsp://localhost:8554/mypath
    # Restart the command if it exits.
    runOnInitRestart: no

runOnDemand allows to run a command when a path is requested by a reader. This can be used to publish a stream on demand:

paths:
  mypath:
    # 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.
    runOnDemand: ffmpeg -i my_file.mp4 -c copy -f rtsp rtsp://localhost:8554/mypath
    # Restart the command if it exits.
    runOnDemandRestart: no

runOnReady allows to run a command when a stream is ready to be read:

paths:
  mypath:
    # This is terminated with SIGINT when the stream is not ready anymore.
    # The following environment variables are available:
    # * MTX_PATH: path name
    # * MTX_SOURCE_TYPE: source type
    # * MTX_SOURCE_ID: source ID
    # * RTSP_PORT: RTSP server port
    # * G1, G2, ...: regular expression groups, if path name is
    #   a regular expression.
    runOnReady: curl http://my-custom-server/webhook?source_type=$MTX_SOURCE_TYPE&source_id=$MTX_SOURCE_ID
    # Restart the command if it exits.
    runOnReadyRestart: no

runOnNotReady allows to run a command when a stream is not available anymore:

paths:
  mypath:
    # Environment variables are the same of runOnReady.
    runOnNotReady: curl http://my-custom-server/webhook?source_type=$MTX_SOURCE_TYPE&source_id=$MTX_SOURCE_ID

runOnRead allows to run a command when a client starts reading:

paths:
  mypath:
    # This is terminated with SIGINT when a client stops reading.
    # The following environment variables are available:
    # * MTX_PATH: path name
    # * MTX_READER_TYPE: reader type
    # * MTX_READER_ID: reader ID
    # * RTSP_PORT: RTSP server port
    # * G1, G2, ...: regular expression groups, if path name is
    #   a regular expression.
    runOnRead: curl http://my-custom-server/webhook?reader_type=$MTX_READER_TYPE&reader_id=$MTX_READER_ID
    # Restart the command if it exits.
    runOnReadRestart: no

runOnUnread allows to run a command when a client stops reading:

paths:
  mypath:
    # Command to run when a client stops reading.
    # Environment variables are the same of runOnRead.
    runOnUnread: curl http://my-custom-server/webhook?reader_type=$MTX_READER_TYPE&reader_id=$MTX_READER_ID

API

The server can be queried and controlled with its API, that must be enabled by setting the api parameter in the configuration:

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/v2/paths/list

Full documentation of the API is available on the dedicated site.

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:

curl localhost:9998/metrics

Obtaining:

# metrics of every path
paths{name="[path_name]",state="[state]"} 1
paths_bytes_received{name="[path_name]",state="[state]"} 1234

# metrics of every HLS muxer
hls_muxers{name="[name]"} 1
hls_muxers_bytes_sent{name="[name]"} 187

# metrics of every RTSP connection
rtsp_conns{id="[id]"} 1
rtsp_conns_bytes_received{id="[id]"} 1234
rtsp_conns_bytes_sent{id="[id]"} 187

# metrics of every RTSP session
rtsp_sessions{id="[id]",state="idle"} 1
rtsp_sessions_bytes_received{id="[id]",state="[state]"} 1234
rtsp_sessions_bytes_sent{id="[id]",state="[state]"} 187

# metrics of every RTSPS connection
rtsps_conns{id="[id]"} 1
rtsps_conns_bytes_received{id="[id]"} 1234
rtsps_conns_bytes_sent{id="[id]"} 187

# metrics of every RTSPS session
rtsps_sessions{id="[id]",state="[state]"} 1
rtsps_sessions_bytes_received{id="[id]",state="[state]"} 1234
rtsps_sessions_bytes_sent{id="[id]",state="[state]"} 187

# metrics of every RTMP connection
rtmp_conns{id="[id]",state="[state]"} 1
rtmp_conns_bytes_received{id="[id]",state="[state]"} 1234
rtmp_conns_bytes_sent{id="[id]",state="[state]"} 187

# metrics of every WebRTC session
webrtc_sessions{id="[id]"} 1
webrtc_sessions_bytes_received{id="[id]",state="[state]"} 1234
webrtc_sessions_bytes_sent{id="[id]",state="[state]"} 187

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

RTSP-specific features

Transport protocols

The RTSP protocol supports different underlying transport protocols, that are chosen by clients during the handshake with the server:

UDP: the most performant, but doesn't work when there's a NAT/firewall between server and clients. It doesn't support encryption.
UDP-multicast: allows to save bandwidth when clients are all in the same LAN, by sending packets once to a fixed multicast IP. It doesn't support encryption.
TCP: the most versatile, does support encryption.

The default transport protocol is UDP. To change the transport protocol, you have to tune the configuration of your client of choice.

Encryption

Incoming and outgoing RTSP streams can be encrypted with TLS, obtaining the RTSPS protocol. A 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 mediamtx.yml, and set the protocols, encryption, serverKey and serverCert parameters:

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

Streams can be published and read with the rtsps scheme and the 8322 port:

rtsps://localhost:8322/mystream

Corrupted frames

In some scenarios, when publishing or reading from the server with RTSP, frames can get corrupted. This can be caused by multiple reasons:

the write queue of the server is too small and can't keep up with the stream throughput. A solution consists in increasing its size:
```
writeQueueSize: 1024
```
The stream throughput is too big and the stream can't be transmitted correctly with the UDP transport 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:
```
protocols: [tcp]
```
In case the source is a camera:
```
paths:
  test:
    source: rtsp://..
    sourceProtocol: tcp
```
The stream throughput is too big to be handled by the network between server and readers. Upgrade the network or decrease the stream bitrate by re-encoding it.

RTMP-specific features

Encryption

RTMP connections can be encrypted with TLS, obtaining the RTMPS protocol. A 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 mediamtx.yml, and set the rtmpEncryption, rtmpServerKey and rtmpServerCert parameters:

rtmpEncryption: optional
rtmpServerKey: server.key
rtmpServerCert: server.crt

Streams can be published and read with the rtmps scheme and the 1937 port:

rtmps://localhost:1937/...

Be aware that RTMPS is currently unsupported by all major players. However, you can use a proxy like stunnel or nginx or a dedicated MediaMTX instance to decrypt streams before reading them.

WebRTC-specific features

Connectivity issues

If the server is hosted inside a container or is behind a NAT, additional configuration is required in order to allow the two WebRTC parts (the browser and the server) to establish a connection (WebRTC/ICE connection).

A first method consists into forcing all WebRTC/ICE connections to pass through a single UDP server port, by using the parameters:

# public IP of the server
webrtcICEHostNAT1To1IPs: [192.168.x.x]
# any port of choice
webrtcICEUDPMuxAddress: :8189

The NAT / container must then be configured in order to route all incoming UDP packets on port 8189 to the server. If you're using Docker, this can be achieved with the flag:

docker run --rm -it \
-p 8189:8189/udp
....
bluenviron/mediamtx

If the UDP protocol is blocked by a firewall, all WebRTC/ICE connections can be forced to pass through a single TCP server port:

# public IP of the server
webrtcICEHostNAT1To1IPs: [192.168.x.x]
# any port of choice
webrtcICETCPMuxAddress: :8189

The NAT / container must then be configured in order to redirect all incoming TCP packets on port 8189 to the server. If you're using Docker, this can be achieved with the flag:

docker run --rm -it \
-p 8189:8189
....
bluenviron/mediamtx

Finally, if none of these methods work, you can force all WebRTC/ICE connections to pass through a TURN server, like coturn, that must be configured externally. The server address and credentials must be set in the configuration file:

webrtcICEServers2:
- url: turn:host:port
  username: user
  password: password

Where user and pass are the username and password of the server. Note that port is not optional.

If the server uses a secret-based authentication (for instance, coturn with the use-auth-secret option), it must be configured by using AUTH_SECRET as username, and the secret as password:

webrtcICEServers2:
- url: turn:host:port
  username: AUTH_SECRET
  password: secret

where secret is the secret of the TURN server. MediaMTX will generate a set of credentials by using the secret, and credentials will be sent to clients before the WebRTC/ICE connection is established.

Compile from source

Standard

Install Go ≥ 1.20, download the repository, open a terminal in it and run:

go build .

The command will produce the mediamtx binary.

Raspberry Pi

The server can be compiled with native support for the Raspberry Pi Camera. Compilation must be performed on a Raspberry Pi, with the following dependencies:

Go ≥ 1.20
libcamera-dev
libfreetype-dev
xxd

Download the repository, open a terminal in it and run:

cd internal/rpicamera/exe
make
cd ../../../
go build -tags rpicamera .

The command will produce the mediamtx binary.

Compile for all supported platforms

Install Docker and launch:

make binaries

The command will produce tarballs in folder binaries/.

Specifications

name	area
RTSP / RTP / RTCP specifications	RTSP
HLS specifications	HLS
RTMP	RTMP
Enhanced RTMP	RTMP
WebRTC: Real-Time Communication in Browsers	WebRTC
WebRTC HTTP Ingestion Protocol (WHIP)	WebRTC
WebRTC HTTP Egress Protocol (WHEP)	WebRTC
The SRT Protocol	SRT
Codec specifications	codecs
Golang project layout	project layout

58 KiB Raw Blame History

Table of contents

Installation

Standalone binary

Docker image

Arch Linux package

OpenWRT package

Basic usage

Publish to the server

By software

FFmpeg

GStreamer

OBS Studio

OpenCV

Web browsers

By device

Generic webcam

Raspberry Pi Cameras

By protocol

SRT clients

SRT cameras and servers

WebRTC clients

WebRTC servers

RTSP clients

RTSP cameras and servers

RTMP clients

RTMP cameras and servers

HLS cameras and servers

UDP/MPEG-TS

Read from the server

By software

FFmpeg

GStreamer

VLC

Ubuntu bug

Encrypted streams

Web browsers

By protocol

SRT

WebRTC

RTSP

Latency

RTMP

HLS

LL-HLS

Compatibility with Apple devices

Latency

Other features

Configuration

Authentication

Encrypt the configuration

Remuxing, re-encoding, compression

Record streams to disk

Forward streams to another server

On-demand publishing

Start on boot

Linux*

Windows*

Hooks

API

Metrics

pprof

RTSP-specific features

Transport protocols

Encryption

Corrupted frames

RTMP-specific features

Encryption

WebRTC-specific features

Connectivity issues

Compile from source

Standard

Raspberry Pi

Compile for all supported platforms

Specifications

Related projects

58 KiB

Raw Blame History