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基於v2.4.3的個人客製版>_^

客製項目

  1. 預設 base image（Dockerfile）
  2. 更改預設帳密（config.json、envs.js）
  3. 新增容器化執行擋（start.sh）
  4. 新增 Desc、Format 屬性於 ChannelST

RTSPtoWeb share you ip camera to world!

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!

Table of Contents

Installation

Installation from source

  1. Download source 
    $ git clone https://github.com/deepch/RTSPtoWeb
  2. CD to Directory 
     $ cd RTSPtoWeb/
  3. Test Run 
     $ GO111MODULE=on go run *.go
  4. Open Browser 
    open web browser http://127.0.0.1:8083 work chrome, safari, firefox

Installation from docker

  1. Run docker container 
    $ docker run --name rtsp-to-web --network host ghcr.io/deepch/rtsptoweb:latest
  2. 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

Configuration

Server settings

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

Stream settings

name            - stream name

Channel settings

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

Authorization play video

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"

RTSP pull modes

  • 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

Example config.json

{
  "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
  }
}

Command-line

Use help to show available args

./RTSPtoWeb --help

Response

Usage of ./RTSPtoWeb:
  -config string
        config patch (/etc/server/config.json or config.json) (default "config.json")
  -debug
        set debug mode (default true)

API documentation

See the API docs

Limitations

Video Codecs Supported: H264 all profiles

Audio Codecs Supported: no

Performance

CPU usage ≈0.2%-1% one (thread) core cpu intel core i7 per stream

Authors

  • Andrey Semochkin - Initial work video - deepch
  • Dmitriy Vladykin - Initial work web UI - vdalex25

See also the list of contributors who participated in this project.

License

This project licensed. License - see the LICENSE.md file for details

webrtc follows license MIT license.

joy4 follows license MIT license.

Other Example

Examples of working with video on golang

paypal.me/AndreySemochkin - You can make one-time donations via PayPal. I'll probably buy a coffee tea. 🍵