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
• Configuration
• Command-line
• API documentation
• Limitations
• Performance
• Authors
• License

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

• RTSPtoWeb
• RTSPtoWebRTC
• RTSPtoWSMP4f
• RTSPtoImage
• RTSPtoHLS
• RTSPtoHLSLL

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