Categorygithub.com/deepch/RTSPtoWeb
modulepackage
2.4.3+incompatible
Repository: https://github.com/deepch/rtsptoweb.git
Documentation: pkg.go.dev

# README

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. :tea:

# Functions

CrossOrigin Access-Control-Allow-Origin any methods.
No description provided by the author
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HTTPAPIServer start http server routes.
No description provided by the author
HTTPAPIServerIndex index file.
HTTPAPIServerStreamAdd function add new stream.
HTTPAPIServerStreamChannelAdd function add new stream.
HTTPAPIServerStreamChannelCodec function return codec info struct.
HTTPAPIServerStreamChannelDelete function delete stream.
HTTPAPIServerStreamChannelEdit function edit stream.
HTTPAPIServerStreamChannelInfo function return stream info struct.
HTTPAPIServerStreamChannelReload function reload stream.
HTTPAPIServerStreamDelete function delete stream.
HTTPAPIServerStreamEdit function edit stream.
HTTPAPIServerStreamHLSLLInit send client ts segment.
HTTPAPIServerStreamHLSLLM3U8 send client m3u8 play list.
HTTPAPIServerStreamHLSLLM4Fragment send client ts segment.
HTTPAPIServerStreamHLSLLM4Segment send client ts segment.
HTTPAPIServerStreamHLSM3U8 send client m3u8 play list.
HTTPAPIServerStreamHLSTS send client ts segment.
HTTPAPIServerStreamInfo function return stream info struct.
HTTPAPIServerStreamMSE func.
HTTPAPIServerStreamDelete function reload stream.
HTTPAPIServerStreams function return stream list.
HTTPAPIServerStreamSaveToMP4 func.
HTTPAPIServerStreamsMultiControlAdd function add new stream's.
HTTPAPIServerStreamsMultiControlDelete function delete stream's.
HTTPAPIServerStreamWebRTC stream video over WebRTC.
No description provided by the author
NewHLSMuxer Segments.
NewStreamCore do load config file.
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RTSPServer func.
RTSPServerClientHandle func.
handleRTSPServerPlay func.
handleRTSPServerPlay func.
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StreamServerRunStream core stream.
StreamServerRunStreamDo stream run do mux.
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# Constants

Default stream type.
Default stream status type.
Default stream status type.
Default stream type.
Default streams signals.
< Y Restart.
Default streams signals.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
RTSP response status codes.
Default stream type.

# Variables

Default www static file dir.
No description provided by the author
Default stream errors.
Default stream errors.
Default stream errors.
Default stream errors.
Default stream errors.
Default stream errors.
Default stream errors.
Default stream errors.
Default stream errors.
Default stream errors.
Default stream errors.
Default stream errors.
Default stream errors.
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Default stream errors.
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# Structs

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ClientST client storage section.
Fragment struct.
Message resp struct.
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MuxerHLS struct.
Segment struct.
SegmentOld HLS cache section.
ServerST server storage section.
StorageST main storage struct.
ServerST stream storage section.
Token auth.