RTSPtoWeb converts your RTSP streams to formats consumable in a web browser like MSE (Media Source Extensions), WebRTC, or HLS. It's fully native Golang without the use of FFmpeg or GStreamer!
- Download source
$ git clone https://github.com/deepch/RTSPtoWeb
- CD to Directory
$ cd RTSPtoWeb/
- Test Run
$ GO111MODULE=on go run *.go
- Open Browser
open web browser http://127.0.0.1:8083 work chrome, safari, firefox
- Run docker container
$ docker run --name rtsp-to-web --network host ghcr.io/deepch/rtsptoweb:latest
- Open Browser
open web browser http://127.0.0.1:8083 in chrome, safari, firefox
You may override the configuration /PATH_TO_CONFIG/config.json
and mount as a docker volume:
$ docker run --name rtsp-to-web \
-v /PATH_TO_CONFIG/config.json:/config/config.json \
--network host \
ghcr.io/deepch/rtsptoweb:latest
debug - enable debug output
log_level - log level (trace, debug, info, warning, error, fatal, or panic)
http_demo - serve static files
http_debug - debug http api server
http_login - http auth login
http_password - http auth password
http_port - http server port
http_dir - path to serve static files from
ice_servers - array of servers to use for STUN/TURN
ice_username - username to use for STUN/TURN
ice_credential - credential to use for STUN/TURN
webrtc_port_min - minimum WebRTC port to use (UDP)
webrtc_port_max - maximum WebRTC port to use (UDP)
https
https_auto_tls
https_auto_tls_name
https_cert
https_key
https_port
rtsp_port - rtsp server port
name - stream name
name - channel name
url - channel rtsp url
on_demand - stream mode static (run any time) or ondemand (run only has viewers)
debug - enable debug output (RTSP client)
audio - enable audio
status - default stream status
1 - enable config
"token": {
"enable": true,
"backend": "http://127.0.0.1/file.php"
}
2 - try
rtsp://127.0.0.1:5541/demo/0?token=you_key
file.php need response json
status: "1" or "0"
- on demand (on_demand=true) - only pull video from the source when there's a viewer
- static (on_demand=false) - pull video from the source constantly
{
"server": {
"debug": true,
"log_level": "info",
"http_demo": true,
"http_debug": false,
"http_login": "demo",
"http_password": "demo",
"http_port": ":8083",
"ice_servers": ["stun:stun.l.google.com:19302"],
"rtsp_port": ":5541"
},
"streams": {
"demo1": {
"name": "test video stream 1",
"channels": {
"0": {
"name": "ch1",
"url": "rtsp://admin:admin@YOU_CAMERA_IP/uri",
"on_demand": true,
"debug": false,
"audio": true,
"status": 0
},
"1": {
"name": "ch2",
"url": "rtsp://admin:admin@YOU_CAMERA_IP/uri",
"on_demand": true,
"debug": false,
"audio": true,
"status": 0
}
}
},
"demo2": {
"name": "test video stream 2",
"channels": {
"0": {
"name": "ch1",
"url": "rtsp://admin:admin@YOU_CAMERA_IP/uri",
"on_demand": true,
"debug": false,
"status": 0
},
"1": {
"name": "ch2",
"url": "rtsp://admin:admin@YOU_CAMERA_IP/uri",
"on_demand": true,
"debug": false,
"status": 0
}
}
}
},
"channel_defaults": {
"on_demand": true
}
}
./RTSPtoWeb --help
Usage of ./RTSPtoWeb:
-config string
config patch (/etc/server/config.json or config.json) (default "config.json")
-debug
set debug mode (default true)
See the API docs
Video Codecs Supported: H264 all profiles
Audio Codecs Supported: no
CPU usage ≈0.2%-1% one (thread) core cpu intel core i7 per stream
See also the list of contributors who participated in this project.
This project licensed. License - see the LICENSE.md file for details
webrtc follows license MIT license.
joy4 follows license MIT license.
Examples of working with video on golang
- You can make one-time donations via PayPal. I'll probably buy a coffee tea. 🍵