README
¶
Ghostream
Boooo! A simple streaming server with authentication and open-source technologies.
This project was developped at Cr@ns to stream events.
Note This project is no longer maintained!
As an alternative, you should try Galène which supports WebRTC-HTTP ingestion protocol (WHIP) for low-latency streaming. OBS Studio introduced WHIP output in version 30.0. Galène supports WHIP since Galène 0.8.
Features:
- WebRTC playback with a lightweight web interface.
- SRT stream input, supported by FFMpeg, OBS and Gstreamer.
- Low-latency streaming, sub-second with web player.
- Authentication of incoming stream using LDAP server.
- Possibility to forward stream to other streaming servers.
Installation on Debian/Ubuntu
On Ubuntu 20.10+ or Debian 11+, you can install directly ghostream,
sudo apt install git golang ffmpeg libsrt1-openssl
go get gitlab.crans.org/nounous/ghostream
On Ubuntu 20.04 or Debian Buster, you may manually install libsrt 1.4.1: install libsrt1-openssl 1.4.1 then libsrt-openssl-dev 1.4.1.
For development, you may clone this repository, then go run main.go.
Installation with Docker
An example is given in docs/docker-compose.yml. It uses Traefik reverse proxy.
You can also launch the Docker image using,
docker build . -t ghostream
docker run -it --rm -p 2112:2112 -p 9710:9710/udp -p 8080:8080 -p 10000-10005:10000-10005/udp ghostream
Configuration
Ghostream can be configured by placing ghostream.yml in /etc/ghostream/.
You can overwrite the configuration path with GHOSTREAM_CONFIG environnement variable.
You can also overwride any value using environnement variables, e.g. GHOSTREAM_AUTH_BACKEND=ldap will change the authentification backend.
Streaming
As stated by OBS wiki, when streaming you should adapt the latency to 2.5 * (the round-trip time with server, in μs).
With OBS
As OBS uses FFMpeg, you need to have FFMpeg compiled with SRT support. To check if SR is available, run ffmpeg -protocols | grep srt.
On Windows and MacOS, OBS comes with his own FFMpeg that will work.
In OBS, go to "Settings" -> "Output" -> "Recording" the select "Output to URL" and change the URL to srt://127.0.0.1:9710?streamid=demo:demo.
For container, you may use MPEGTS for now (will change).
With GStreamer
To stream your X11 screen,
gst-launch-1.0 ximagesrc startx=0 show-pointer=true use-damage=0 \
! videoconvert \
! x264enc bitrate=32000 tune=zerolatency speed-preset=veryfast byte-stream=true threads=1 key-int-max=15 intra-refresh=true ! video/x-h264, profile=baseline, framerate=30/1 \
! mpegtsmux \
! srtserversink uri=srt://127.0.0.1:9710/ latency=1000000 streamid=demo:demo
This might not work at the moment.
Playing stream
With a web browser and WebRTC
Ghostream expose a web server on 0.0.0.0:8080 by default.
By opening this in a browser, you will be able to get instructions on how to stream, and if you append /streamname to the URL, then you will be able to watch the stream named streamname.
The web player also integrates a side widget that is configurable.
Integrate the player in an iframe
To integrate the player without the side widget, you can append ?nowidget to the URL.
<iframe src="https://example.com/stream_name?nowidget" scrolling="no" allowfullscreen="true" width="1280" height="750.4" frameborder="0"></iframe>
The iframe size should be a 16/9 ratio, with additionnal 30.4px for the control bar.
With ffplay
You may directly open the SRT stream with ffplay:
ffplay -fflags nobuffer srt://127.0.0.1:9710?streamid=demo
With MPV
As MPV uses ffmpeg libav, support for SRT streams can be easily added. See current pull request.
Troubleshooting
ld returns an error when launching ghostream
When missing libsrt-openssl-dev on Debian/Ubuntu,
then srtgo package is unable to build.
~/ghostream$ go run main.go
# github.com/haivision/srtgo
/usr/bin/ld: cannot find -lsrt
/usr/bin/ld: cannot find -lsrt
/usr/bin/ld: cannot find -lsrt
collect2: error: ld returned 1 exit status
References
- Phil Cluff (2019), Streaming video on the internet without MPEG.
- MDN web docs, Signaling and video calling.
- WebRTC For The Curious
- OBS Wiki, Streaming With SRT Protocol.
- Livepeer media server, Evaluate Go-FFmpeg Bindings
Documentation
¶
Overview ¶
Package main provides the full-featured server with configuration loading and communication between routines.
Source Files
Directories
¶
| Path | Synopsis |
|---|---|
|
Package auth manages backends to auth incoming streams
|
Package auth manages backends to auth incoming streams |
|
basic
Package basic provides a basic authentification backend
|
Package basic provides a basic authentification backend |
|
ldap
Package ldap provides a LDAP authentification backend
|
Package ldap provides a LDAP authentification backend |
|
internal
|
|
|
config
Package config loads application settings
|
Package config loads application settings |
|
monitoring
Package monitoring serves Prometheus monitoring endpoints
|
Package monitoring serves Prometheus monitoring endpoints |
|
Package messaging defines a structure to communication between inputs and outputs
|
Package messaging defines a structure to communication between inputs and outputs |
|
stream
|
|
|
forwarding
Package forwarding forwards incoming stream to other streaming services
|
Package forwarding forwards incoming stream to other streaming services |
|
ovenmediaengine
Package ovenmediaengine provides the forwarding to an ovenmediaengine server to handle the web client
|
Package ovenmediaengine provides the forwarding to an ovenmediaengine server to handle the web client |
|
srt
Package srt serves a SRT server
|
Package srt serves a SRT server |
|
telnet
Package telnet expose text version of stream.
|
Package telnet expose text version of stream. |
|
webrtc
Package webrtc provides the backend to simulate a WebRTC client to send stream
|
Package webrtc provides the backend to simulate a WebRTC client to send stream |
|
Package transcoder manages transcoders
|
Package transcoder manages transcoders |
|
text
Package text transcode a video to text
|
Package text transcode a video to text |
|
Package web serves the JavaScript player and WebRTC negotiation
|
Package web serves the JavaScript player and WebRTC negotiation |