Revert changes in readme file

This commit is contained in:
Alex X
2024-04-29 12:26:53 +03:00
parent 152719441e
commit 64ac27d93d
+145 -157
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@@ -1,7 +1,7 @@
# go2rtc
[![](https://img.shields.io/github/stars/AlexxIT/go2rtc?style=flat-square&logo=github)](https://github.com/AlexxIT/go2rtc/stargazers)
[![](https://img.shields.io/docker/pulls/alexxit/go2rtc?style=flat-square&logo=docker&logoColor=white&label=pulls)](https://hub.docker.com/r/alexxit/go2rtc)
[![](https://img.shields.io/github/stars/AlexxIT/go2rtc?style=flat-square&logo=github)](https://github.com/AlexxIT/go2rtc/stargazers)
[![](https://img.shields.io/docker/pulls/alexxit/go2rtc?style=flat-square&logo=docker&logoColor=white&label=pulls)](https://hub.docker.com/r/alexxit/go2rtc)
[![](https://img.shields.io/github/downloads/AlexxIT/go2rtc/total?color=blue&style=flat-square&logo=github)](https://github.com/AlexxIT/go2rtc/releases)
[![](https://goreportcard.com/badge/github.com/AlexxIT/go2rtc)](https://goreportcard.com/report/github.com/AlexxIT/go2rtc)
@@ -20,9 +20,9 @@ Ultimate camera streaming application with support RTSP, WebRTC, HomeKit, FFmpeg
- on the fly transcoding for unsupported codecs via [FFmpeg](#source-ffmpeg)
- play audio files and live streams on some cameras with [speaker](#stream-to-camera)
- multi-source 2-way [codecs negotiation](#codecs-negotiation)
- mixing tracks from different sources to single stream
- auto match client supported codecs
- [2-way audio](#two-way-audio) for some cameras
- mixing tracks from different sources to single stream
- auto match client supported codecs
- [2-way audio](#two-way-audio) for some cameras
- streaming from private networks via [ngrok](#module-ngrok)
- can be [integrated to](#module-api) any smart home platform or be used as [standalone app](#go2rtc-binary)
@@ -37,60 +37,60 @@ Ultimate camera streaming application with support RTSP, WebRTC, HomeKit, FFmpeg
---
- [Fast start](#fast-start)
- [go2rtc: Binary](#go2rtc-binary)
- [go2rtc: Docker](#go2rtc-docker)
- [go2rtc: Home Assistant Add-on](#go2rtc-home-assistant-add-on)
- [go2rtc: Home Assistant Integration](#go2rtc-home-assistant-integration)
- [go2rtc: Dev version](#go2rtc-dev-version)
- [Configuration](#configuration)
- [Module: Streams](#module-streams)
- [Two way audio](#two-way-audio)
- [Source: RTSP](#source-rtsp)
- [Source: RTMP](#source-rtmp)
- [Source: HTTP](#source-http)
- [Source: ONVIF](#source-onvif)
- [Source: FFmpeg](#source-ffmpeg)
- [Source: FFmpeg Device](#source-ffmpeg-device)
- [Source: Exec](#source-exec)
- [Source: Echo](#source-echo)
- [Source: Expr](#source-expr)
- [Source: HomeKit](#source-homekit)
- [Source: Bubble](#source-bubble)
- [Source: DVRIP](#source-dvrip)
- [Source: Tapo](#source-tapo)
- [Source: Kasa](#source-kasa)
- [Source: GoPro](#source-gopro)
- [Source: Ivideon](#source-ivideon)
- [Source: Hass](#source-hass)
- [Source: ISAPI](#source-isapi)
- [Source: Nest](#source-nest)
- [Source: Roborock](#source-roborock)
- [Source: WebRTC](#source-webrtc)
- [Source: WebTorrent](#source-webtorrent)
- [Incoming sources](#incoming-sources)
- [Stream to camera](#stream-to-camera)
- [Publish stream](#publish-stream)
- [Module: API](#module-api)
- [Module: RTSP](#module-rtsp)
- [Module: RTMP](#module-rtmp)
- [Module: WebRTC](#module-webrtc)
- [Module: HomeKit](#module-homekit)
- [Module: WebTorrent](#module-webtorrent)
- [Module: ngrok](#module-ngrok)
- [Module: Hass](#module-hass)
- [Module: MP4](#module-mp4)
- [Module: HLS](#module-hls)
- [Module: MJPEG](#module-mjpeg)
- [Module: Log](#module-log)
- [Security](#security)
- [Codecs filters](#codecs-filters)
- [Codecs madness](#codecs-madness)
- [Codecs negotiation](#codecs-negotiation)
- [Projects using go2rtc](#projects-using-go2rtc)
- [Camera experience](#cameras-experience)
- [TIPS](#tips)
- [FAQ](#faq)
* [Fast start](#fast-start)
* [go2rtc: Binary](#go2rtc-binary)
* [go2rtc: Docker](#go2rtc-docker)
* [go2rtc: Home Assistant Add-on](#go2rtc-home-assistant-add-on)
* [go2rtc: Home Assistant Integration](#go2rtc-home-assistant-integration)
* [go2rtc: Dev version](#go2rtc-dev-version)
* [Configuration](#configuration)
* [Module: Streams](#module-streams)
* [Two way audio](#two-way-audio)
* [Source: RTSP](#source-rtsp)
* [Source: RTMP](#source-rtmp)
* [Source: HTTP](#source-http)
* [Source: ONVIF](#source-onvif)
* [Source: FFmpeg](#source-ffmpeg)
* [Source: FFmpeg Device](#source-ffmpeg-device)
* [Source: Exec](#source-exec)
* [Source: Echo](#source-echo)
* [Source: Expr](#source-expr)
* [Source: HomeKit](#source-homekit)
* [Source: Bubble](#source-bubble)
* [Source: DVRIP](#source-dvrip)
* [Source: Tapo](#source-tapo)
* [Source: Kasa](#source-kasa)
* [Source: GoPro](#source-gopro)
* [Source: Ivideon](#source-ivideon)
* [Source: Hass](#source-hass)
* [Source: ISAPI](#source-isapi)
* [Source: Nest](#source-nest)
* [Source: Roborock](#source-roborock)
* [Source: WebRTC](#source-webrtc)
* [Source: WebTorrent](#source-webtorrent)
* [Incoming sources](#incoming-sources)
* [Stream to camera](#stream-to-camera)
* [Publish stream](#publish-stream)
* [Module: API](#module-api)
* [Module: RTSP](#module-rtsp)
* [Module: RTMP](#module-rtmp)
* [Module: WebRTC](#module-webrtc)
* [Module: HomeKit](#module-homekit)
* [Module: WebTorrent](#module-webtorrent)
* [Module: ngrok](#module-ngrok)
* [Module: Hass](#module-hass)
* [Module: MP4](#module-mp4)
* [Module: HLS](#module-hls)
* [Module: MJPEG](#module-mjpeg)
* [Module: Log](#module-log)
* [Security](#security)
* [Codecs filters](#codecs-filters)
* [Codecs madness](#codecs-madness)
* [Codecs negotiation](#codecs-negotiation)
* [Projects using go2rtc](#projects-using-go2rtc)
* [Camera experience](#cameras-experience)
* [TIPS](#tips)
* [FAQ](#faq)
## Fast start
@@ -134,8 +134,8 @@ Container [alexxit/go2rtc](https://hub.docker.com/r/alexxit/go2rtc) with support
[![](https://my.home-assistant.io/badges/supervisor_addon.svg)](https://my.home-assistant.io/redirect/supervisor_addon/?addon=a889bffc_go2rtc&repository_url=https%3A%2F%2Fgithub.com%2FAlexxIT%2Fhassio-addons)
1. Install Add-On:
- Settings > Add-ons > Plus > Repositories > Add `https://github.com/AlexxIT/hassio-addons`
- go2rtc > Install > Start
- Settings > Add-ons > Plus > Repositories > Add `https://github.com/AlexxIT/hassio-addons`
- go2rtc > Install > Start
2. Setup [Integration](#module-hass)
### go2rtc: Home Assistant Integration
@@ -152,7 +152,7 @@ Latest, but maybe unstable version:
## Configuration
- by default go2rtc will search `go2rtc.yaml` in the current work directory
- by default go2rtc will search `go2rtc.yaml` in the current work dirrectory
- `api` server will start on default **1984 port** (TCP)
- `rtsp` server will start on default **8554 port** (TCP)
- `webrtc` will use port **8555** (TCP/UDP) for connections
@@ -230,7 +230,7 @@ streams:
amcrest_doorbell:
- rtsp://username:password@192.168.1.123:554/cam/realmonitor?channel=1&subtype=0#backchannel=0
unifi_camera: rtspx://192.168.1.123:7441/fD6ouM72bWoFijxK
glichy_camera: ffmpeg:rstp://username:password@192.168.1.123/live/ch00_1
glichy_camera: ffmpeg:rstp://username:password@192.168.1.123/live/ch00_1
```
**Recommendations**
@@ -249,7 +249,7 @@ streams:
Format: `rtsp...#{param1}#{param2}#{param3}`
- Add custom timeout `#timeout=30` (in seconds)
- Ignore audio - `#media=video` or ignore video - `#media=audio`
- Ignore audio - `#media=video` or ignore video - `#media=audio`
- Ignore two way audio API `#backchannel=0` - important for some glitchy cameras
- Use WebSocket transport `#transport=ws...`
@@ -258,7 +258,7 @@ Format: `rtsp...#{param1}#{param2}#{param3}`
```yaml
streams:
# WebSocket with authorization, RTSP - without
axis-rtsp-ws: rtsp://192.168.1.123:4567/axis-media/media.amp?overview=0&camera=1&resolution=1280x720&videoframeskipmode=empty&Axis-Orig-Sw=true#transport=ws://user:pass@192.168.1.123:4567/rtsp-over-websocket
axis-rtsp-ws: rtsp://192.168.1.123:4567/axis-media/media.amp?overview=0&camera=1&resolution=1280x720&videoframeskipmode=empty&Axis-Orig-Sw=true#transport=ws://user:pass@192.168.1.123:4567/rtsp-over-websocket
# WebSocket without authorization, RTSP - with
dahua-rtsp-ws: rtsp://user:pass@192.168.1.123/cam/realmonitor?channel=1&subtype=1&proto=Private3#transport=ws://192.168.1.123/rtspoverwebsocket
```
@@ -287,7 +287,7 @@ Source also support HTTP and TCP streams with autodetection for different format
streams:
# [HTTP-FLV] stream in video/x-flv format
http_flv: http://192.168.1.123:20880/api/camera/stream/780900131155/657617
# [JPEG] snapshots from Dahua camera, will be converted to MJPEG stream
dahua_snap: http://admin:password@192.168.1.123/cgi-bin/snapshot.cgi?channel=1
@@ -305,7 +305,7 @@ streams:
#### Source: ONVIF
_[New in v1.5.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.5.0)_
*[New in v1.5.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.5.0)*
The source is not very useful if you already know RTSP and snapshot links for your camera. But it can be useful if you don't.
@@ -354,7 +354,7 @@ But you can override them via YAML config. You can also add your own formats to
```yaml
ffmpeg:
bin: ffmpeg # path to ffmpeg binary
bin: ffmpeg # path to ffmpeg binary
h264: "-codec:v libx264 -g:v 30 -preset:v superfast -tune:v zerolatency -profile:v main -level:v 4.1"
mycodec: "-any args that supported by ffmpeg..."
myinput: "-fflags nobuffer -flags low_delay -timeout 5000000 -i {input}"
@@ -390,7 +390,7 @@ Format: `ffmpeg:device?{input-params}#{param1}#{param2}#{param3}`
```yaml
streams:
linux_usbcam: ffmpeg:device?video=0&video_size=1280x720#video=h264
linux_usbcam: ffmpeg:device?video=0&video_size=1280x720#video=h264
windows_webcam: ffmpeg:device?video=0#video=h264
macos_facetime: ffmpeg:device?video=0&audio=1&video_size=1280x720&framerate=30#video=h264#audio=pcma
```
@@ -399,23 +399,12 @@ streams:
#### Source: Exec
Exec source can run any external application and expect data from it. Two transports are supported - **pipe** (_from [v1.5.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.5.0)_) and **RTSP**.
Exec source can run any external application and expect data from it. Two transports are supported - **pipe** (*from [v1.5.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.5.0)*) and **RTSP**.
If you want to use **RTSP** transport - the command must contain the `{output}` argument in any place. On launch, it will be replaced by the local address of the RTSP server.
**pipe** reads data from app stdout in different formats: **MJPEG**, **H.264/H.265 bitstream**, **MPEG-TS**.
Pipe commands support two parameters:
- **killsignal**: Signal which will be send to stop the process (default: sigkill)
- **killtimeout**: Time in seconds after the process will killed with sigkill (default: 5)
The **killtimeout** parameter is only relevant if **killsignal** is not sigkill.
Setting **killtimeout** to a negative number (or zero) will result in an immediate SIGKILL.
See `man 7 signal` to get a full list of all the signals.
Format: `exec:{command}#{param1}#{param2}`
The source can be used with:
- [FFmpeg](https://ffmpeg.org/) - go2rtc ffmpeg source just a shortcut to exec source
@@ -428,7 +417,6 @@ streams:
stream: exec:ffmpeg -re -i /media/BigBuckBunny.mp4 -c copy -rtsp_transport tcp -f rtsp {output}
picam_h264: exec:libcamera-vid -t 0 --inline -o -
picam_mjpeg: exec:libcamera-vid -t 0 --codec mjpeg -o -
canon: exec:gphoto2 --capture-movie --stdout#killsignal=sigint
```
#### Source: Echo
@@ -446,7 +434,7 @@ streams:
#### Source: Expr
_[New in v1.8.2](https://github.com/AlexxIT/go2rtc/releases/tag/v1.8.2)_
*[New in v1.8.2](https://github.com/AlexxIT/go2rtc/releases/tag/v1.8.2)*
Like `echo` source, but uses the built-in [expr](https://github.com/antonmedv/expr) expression language ([read more](https://github.com/AlexxIT/go2rtc/blob/master/internal/expr/README.md)).
@@ -485,7 +473,7 @@ RTSP link with "normal" audio for any player: `rtsp://192.168.1.123:8554/aqara_g
#### Source: Bubble
_[New in v1.6.1](https://github.com/AlexxIT/go2rtc/releases/tag/v1.6.1)_
*[New in v1.6.1](https://github.com/AlexxIT/go2rtc/releases/tag/v1.6.1)*
Other names: [ESeeCloud](http://www.eseecloud.com/), [dvr163](http://help.dvr163.com/).
@@ -499,7 +487,7 @@ streams:
#### Source: DVRIP
_[New in v1.2.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.2.0)_
*[New in v1.2.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.2.0)*
Other names: DVR-IP, NetSurveillance, Sofia protocol (NETsurveillance ActiveX plugin XMeye SDK).
@@ -519,7 +507,7 @@ streams:
#### Source: Tapo
_[New in v1.2.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.2.0)_
*[New in v1.2.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.2.0)*
[TP-Link Tapo](https://www.tapo.com/) proprietary camera protocol with **two way audio** support.
@@ -545,7 +533,7 @@ echo -n "cloud password" | shasum -a 256 | awk '{print toupper($0)}'
#### Source: Kasa
_[New in v1.7.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.7.0)_
*[New in v1.7.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.7.0)*
[TP-Link Kasa](https://www.kasasmart.com/) non-standard protocol [more info](https://medium.com/@hu3vjeen/reverse-engineering-tp-link-kc100-bac4641bf1cd).
@@ -556,7 +544,7 @@ streams:
#### Source: GoPro
_[New in v1.8.3](https://github.com/AlexxIT/go2rtc/releases/tag/v1.8.3)_
*[New in v1.8.3](https://github.com/AlexxIT/go2rtc/releases/tag/v1.8.3)*
Support streaming from [GoPro](https://gopro.com/) cameras, connected via USB or Wi-Fi to Linux, Mac, Windows. [Read more](https://github.com/AlexxIT/go2rtc/tree/master/internal/gopro).
@@ -580,14 +568,14 @@ Support import camera links from [Home Assistant](https://www.home-assistant.io/
```yaml
hass:
config: "/config" # skip this setting if you Hass Add-on user
config: "/config" # skip this setting if you Hass Add-on user
streams:
generic_camera: hass:Camera1 # Settings > Integrations > Integration Name
generic_camera: hass:Camera1 # Settings > Integrations > Integration Name
aqara_g3: hass:Camera-Hub-G3-AB12
```
**WebRTC Cameras** (_from [v1.6.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.6.0)_)
**WebRTC Cameras** (*from [v1.6.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.6.0)*)
Any cameras in WebRTC format are supported. But at the moment Home Assistant only supports some [Nest](https://www.home-assistant.io/integrations/nest/) cameras in this fomat.
@@ -607,7 +595,7 @@ By default, the Home Assistant API does not allow you to get dynamic RTSP link t
#### Source: ISAPI
_[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)_
*[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)*
This source type support only backchannel audio for Hikvision ISAPI protocol. So it should be used as second source in addition to the RTSP protocol.
@@ -620,7 +608,7 @@ streams:
#### Source: Nest
_[New in v1.6.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.6.0)_
*[New in v1.6.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.6.0)*
Currently only WebRTC cameras are supported. Stream reconnects every 5 minutes.
@@ -633,7 +621,7 @@ streams:
#### Source: Roborock
_[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)_
*[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)*
This source type support Roborock vacuums with cameras. Known working models:
@@ -646,7 +634,7 @@ If you have graphic pin for your vacuum - add it as numeric pin (lines: 123, 456
#### Source: WebRTC
_[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)_
*[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)*
This source type support four connection formats.
@@ -658,24 +646,24 @@ This source type support four connection formats.
This format is only supported in go2rtc. Unlike WHEP it supports asynchronous WebRTC connection and two way audio.
**openipc** (_from [v1.7.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.7.0)_)
**openipc** (*from [v1.7.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.7.0)*)
Support connection to [OpenIPC](https://openipc.org/) cameras.
**wyze** (_from [v1.6.1](https://github.com/AlexxIT/go2rtc/releases/tag/v1.6.1)_)
**wyze** (*from [v1.6.1](https://github.com/AlexxIT/go2rtc/releases/tag/v1.6.1)*)
Supports connection to [Wyze](https://www.wyze.com/) cameras, using WebRTC protocol. You can use [docker-wyze-bridge](https://github.com/mrlt8/docker-wyze-bridge) project to get connection credentials.
**kinesis** (_from [v1.6.1](https://github.com/AlexxIT/go2rtc/releases/tag/v1.6.1)_)
**kinesis** (*from [v1.6.1](https://github.com/AlexxIT/go2rtc/releases/tag/v1.6.1)*)
Supports [Amazon Kinesis Video Streams](https://aws.amazon.com/kinesis/video-streams/), using WebRTC protocol. You need to specify signalling WebSocket URL with all credentials in query params, `client_id` and `ice_servers` list in [JSON format](https://developer.mozilla.org/en-US/docs/Web/API/RTCIceServer).
```yaml
streams:
webrtc-whep: webrtc:http://192.168.1.123:1984/api/webrtc?src=camera1
webrtc-go2rtc: webrtc:ws://192.168.1.123:1984/api/ws?src=camera1
webrtc-whep: webrtc:http://192.168.1.123:1984/api/webrtc?src=camera1
webrtc-go2rtc: webrtc:ws://192.168.1.123:1984/api/ws?src=camera1
webrtc-openipc: webrtc:ws://192.168.1.123/webrtc_ws#format=openipc#ice_servers=[{"urls":"stun:stun.kinesisvideo.eu-north-1.amazonaws.com:443"}]
webrtc-wyze: webrtc:http://192.168.1.123:5000/signaling/camera1?kvs#format=wyze
webrtc-wyze: webrtc:http://192.168.1.123:5000/signaling/camera1?kvs#format=wyze
webrtc-kinesis: webrtc:wss://...amazonaws.com/?...#format=kinesis#client_id=...#ice_servers=[{...},{...}]
```
@@ -683,7 +671,7 @@ streams:
#### Source: WebTorrent
_[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)_
*[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)*
This source can get a stream from another go2rtc via [WebTorrent](#module-webtorrent) protocol.
@@ -723,7 +711,7 @@ By default, go2rtc establishes a connection to the source when any client reques
#### Incoming: Browser
_[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)_
*[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)*
You can turn the browser of any PC or mobile into an IP-camera with support video and two way audio. Or even broadcast your PC screen:
@@ -735,7 +723,7 @@ You can turn the browser of any PC or mobile into an IP-camera with support vide
#### Incoming: WebRTC/WHIP
_[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)_
*[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)*
You can use **OBS Studio** or any other broadcast software with [WHIP](https://www.ietf.org/archive/id/draft-ietf-wish-whip-01.html) protocol support. This standard has not yet been approved. But you can download OBS Studio [dev version](https://github.com/obsproject/obs-studio/actions/runs/3969201209):
@@ -743,7 +731,7 @@ You can use **OBS Studio** or any other broadcast software with [WHIP](https://w
#### Stream to camera
_[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)_
*[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)*
go2rtc support play audio files (ex. music or [TTS](https://www.home-assistant.io/integrations/#text-to-speech)) and live streams (ex. radio) on cameras with [two way audio](#two-way-audio) support (RTSP/ONVIF cameras, TP-Link Tapo, Hikvision ISAPI, Roborock vacuums, any Browser).
@@ -753,7 +741,7 @@ API example:
POST http://localhost:1984/api/streams?dst=camera1&src=ffmpeg:http://example.com/song.mp3#audio=pcma#input=file
```
- you can stream: local files, web files, live streams or any format, supported by FFmpeg
- you can stream: local files, web files, live streams or any format, supported by FFmpeg
- you should use [ffmpeg source](#source-ffmpeg) for transcoding audio to codec, that your camera supports
- you can check camera codecs on the go2rtc WebUI info page when the stream is active
- some cameras support only low quality `PCMA/8000` codec (ex. [Tapo](#source-tapo))
@@ -765,7 +753,7 @@ POST http://localhost:1984/api/streams?dst=camera1&src=ffmpeg:http://example.com
### Publish stream
_[New in v1.8.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.8.0)_
*[New in v1.8.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.8.0)*
You can publish any stream to streaming services (YouTube, Telegram, etc.) via RTMP/RTMPS. Important:
@@ -816,22 +804,22 @@ The HTTP API is the main part for interacting with the application. Default addr
```yaml
api:
listen: ":1984" # default ":1984", HTTP API port ("" - disabled)
username: "admin" # default "", Basic auth for WebUI
password: "pass" # default "", Basic auth for WebUI
base_path: "/rtc" # default "", API prefix for serve on suburl (/api => /rtc/api)
static_dir: "www" # default "", folder for static files (custom web interface)
origin: "*" # default "", allow CORS requests (only * supported)
listen: ":1984" # default ":1984", HTTP API port ("" - disabled)
username: "admin" # default "", Basic auth for WebUI
password: "pass" # default "", Basic auth for WebUI
base_path: "/rtc" # default "", API prefix for serve on suburl (/api => /rtc/api)
static_dir: "www" # default "", folder for static files (custom web interface)
origin: "*" # default "", allow CORS requests (only * supported)
tls_listen: ":443" # default "", enable HTTPS server
tls_cert: | # default "", PEM-encoded fullchain certificate for HTTPS
tls_cert: | # default "", PEM-encoded fullchain certificate for HTTPS
-----BEGIN CERTIFICATE-----
...
-----END CERTIFICATE-----
tls_key: | # default "", PEM-encoded private key for HTTPS
tls_key: | # default "", PEM-encoded private key for HTTPS
-----BEGIN PRIVATE KEY-----
...
-----END PRIVATE KEY-----
unix_listen: "/tmp/go2rtc.sock" # default "", unix socket listener for API
unix_listen: "/tmp/go2rtc.sock" # default "", unix socket listener for API
```
**PS:**
@@ -847,10 +835,10 @@ You can enable external password protection for your RTSP streams. Password prot
```yaml
rtsp:
listen: ":8554" # RTSP Server TCP port, default - 8554
username: "admin" # optional, default - disabled
password: "pass" # optional, default - disabled
default_query: "video&audio" # optional, default codecs filters
listen: ":8554" # RTSP Server TCP port, default - 8554
username: "admin" # optional, default - disabled
password: "pass" # optional, default - disabled
default_query: "video&audio" # optional, default codecs filters
```
By default go2rtc provide RTSP-stream with only one first video and only one first audio. You can change it with the `default_query` setting:
@@ -864,15 +852,15 @@ Read more about [codecs filters](#codecs-filters).
### Module: RTMP
_[New in v1.8.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.8.0)_
*[New in v1.8.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.8.0)*
You can get any stream as RTMP-stream: `rtmp://192.168.1.123/{stream_name}`. Only H264/AAC codecs supported right now.
[Incoming stream](#incoming-sources) in RTMP-format tested only with [OBS Studio](https://obsproject.com/) and Dahua camera. Different FFmpeg versions has differnt problems with this format.
[Incoming stream](#incoming-sources) in RTMP-format tested only with [OBS Studio](https://obsproject.com/) and Dahua camera. Different FFmpeg versions has differnt problems with this format.
```yaml
rtmp:
listen: ":1935" # by default - disabled!
listen: ":1935" # by default - disabled!
```
### Module: WebRTC
@@ -890,7 +878,7 @@ But about 10-20% of users may need to configure additional settings for external
```yaml
webrtc:
listen: ":8555" # address of your local server and port (TCP/UDP)
listen: ":8555" # address of your local server and port (TCP/UDP)
```
**Static public IP**
@@ -901,7 +889,7 @@ webrtc:
```yaml
webrtc:
candidates:
- 216.58.210.174:8555 # if you have static public IP-address
- 216.58.210.174:8555 # if you have static public IP-address
```
**Dynamic public IP**
@@ -913,7 +901,7 @@ webrtc:
```yaml
webrtc:
candidates:
- stun:8555 # if you have dynamic public IP-address
- stun:8555 # if you have dynamic public IP-address
```
**Private IP**
@@ -944,7 +932,7 @@ webrtc:
### Module: HomeKit
_[New in v1.7.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.7.0)_
*[New in v1.7.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.7.0)*
HomeKit module can work in two modes:
@@ -961,7 +949,7 @@ HomeKit module can work in two modes:
streams:
dahua1: rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0
homekit:
dahua1: # same stream ID from streams list, default PIN - 19550224
dahua1: # same stream ID from streams list, default PIN - 19550224
```
**Full config**
@@ -970,15 +958,15 @@ homekit:
streams:
dahua1:
- rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0
- ffmpeg:dahua1#video=h264#hardware # if your camera doesn't support H264, important for HomeKit
- ffmpeg:dahua1#audio=opus # only OPUS audio supported by HomeKit
- ffmpeg:dahua1#video=h264#hardware # if your camera doesn't support H264, important for HomeKit
- ffmpeg:dahua1#audio=opus # only OPUS audio supported by HomeKit
homekit:
dahua1: # same stream ID from streams list
pin: 12345678 # custom PIN, default: 19550224
name: Dahua camera # custom camera name, default: generated from stream ID
device_id: dahua1 # custom ID, default: generated from stream ID
device_private: dahua1 # custom key, default: generated from stream ID
dahua1: # same stream ID from streams list
pin: 12345678 # custom PIN, default: 19550224
name: Dahua camera # custom camera name, default: generated from stream ID
device_id: dahua1 # custom ID, default: generated from stream ID
device_private: dahua1 # custom key, default: generated from stream ID
```
**Proxy HomeKit camera**
@@ -990,15 +978,15 @@ homekit:
streams:
aqara1:
- homekit://...
- ffmpeg:aqara1#audio=aac#audio=opus # optional audio transcoding
- ffmpeg:aqara1#audio=aac#audio=opus # optional audio transcoding
homekit:
aqara1: # same stream ID from streams list
aqara1: # same stream ID from streams list
```
### Module: WebTorrent
_[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)_
*[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)*
This module support:
@@ -1015,9 +1003,9 @@ You can create permanent external links in go2rtc config:
```yaml
webtorrent:
shares:
super-secret-share: # share name, should be unique among all go2rtc users!
super-secret-share: # share name, should be unique among all go2rtc users!
pwd: super-secret-password
src: rtsp-dahua1 # stream name from streams section
src: rtsp-dahua1 # stream name from streams section
```
Link example: https://alexxit.github.io/go2rtc/#share=02SNtgjKXY&pwd=wznEQqznxW&media=video+audio
@@ -1069,12 +1057,12 @@ version: "2"
authtoken: eW91IHNoYWxsIG5vdCBwYXNzCnlvdSBzaGFsbCBub3QgcGFzcw
tunnels:
api:
addr: 1984 # use the same port as in go2rtc config
addr: 1984 # use the same port as in go2rtc config
proto: http
basic_auth:
- admin:password # you can set login/pass for your web interface
- admin:password # you can set login/pass for your web interface
webrtc:
addr: 8555 # use the same port as in go2rtc config
addr: 8555 # use the same port as in go2rtc config
proto: tcp
```
@@ -1130,7 +1118,7 @@ API examples:
- MP4 snapshot: `http://192.168.1.123:1984/api/frame.mp4?src=camera1` (H264, H265)
- MP4 stream: `http://192.168.1.123:1984/api/stream.mp4?src=camera1` (H264, H265, AAC)
- MP4 file: `http://192.168.1.123:1984/api/stream.mp4?src=camera1` (H264, H265\*, AAC, OPUS, MP3, PCMA, PCMU, PCM)
- MP4 file: `http://192.168.1.123:1984/api/stream.mp4?src=camera1` (H264, H265*, AAC, OPUS, MP3, PCMA, PCMU, PCM)
- You can use `mp4`, `mp4=flac` and `mp4=all` param for codec filters
- You can use `duration` param in seconds (ex. `duration=15`)
- You can use `filename` param (ex. `filename=record.mp4`)
@@ -1139,11 +1127,11 @@ API examples:
Read more about [codecs filters](#codecs-filters).
**PS.** Rotate and scale params don't use transcoding and change video using metadata.
**PS.** Rotate and scale params don't use transcoding and change video using metadata.
### Module: HLS
_[New in v1.1.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.1.0)_
*[New in v1.1.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.1.0)*
[HLS](https://en.wikipedia.org/wiki/HTTP_Live_Streaming) is the worst technology for real-time streaming. It can only be useful on devices that do not support more modern technology, like [WebRTC](#module-webrtc), [MSE/MP4](#module-mp4).
@@ -1180,7 +1168,7 @@ API examples:
- MJPEG stream: `http://192.168.1.123:1984/api/stream.mjpeg?src=camera1`
- JPEG snapshots: `http://192.168.1.123:1984/api/frame.jpeg?src=camera1`
- You can use `width`/`w` and/or `height`/`h` params
- You can use `width`/`w` and/or `height`/`h` params
- You can use `rotate` param with `90`, `180`, `270` or `-90` values
- You can use `hardware`/`hw` param [read more](https://github.com/AlexxIT/go2rtc/wiki/Hardware-acceleration)
@@ -1190,7 +1178,7 @@ You can set different log levels for different modules.
```yaml
log:
level: info # default level
level: info # default level
api: trace
exec: debug
ngrok: info
@@ -1252,18 +1240,18 @@ Some examples:
`AVC/H.264` video can be played almost anywhere. But `HEVC/H.265` has a lot of limitations in supporting with different devices and browsers. It's all about patents and money, you can't do anything about it.
| Device | WebRTC | MSE | HTTP\* | HLS |
| ------------------------------------------------------------------------ | ---------------------------------------- | --------------------------------------- | -------------------------------------------- | ---------------------------- |
| _latency_ | best | medium | bad | bad |
| - Desktop Chrome 107+ <br/> - Desktop Edge <br/> - Android Chrome 107+ | H264 <br/> PCMU, PCMA <br/> OPUS | H264, H265* <br/> AAC, FLAC* <br/> OPUS | H264, H265* <br/> AAC, FLAC* <br/> OPUS, MP3 | no |
| Desktop Firefox | H264 <br/> PCMU, PCMA <br/> OPUS | H264 <br/> AAC, FLAC\* <br/> OPUS | H264 <br/> AAC, FLAC\* <br/> OPUS | no |
| - Desktop Safari 14+ <br/> - iPad Safari 14+ <br/> - iPhone Safari 17.1+ | H264, H265\* <br/> PCMU, PCMA <br/> OPUS | H264, H265 <br/> AAC, FLAC\* | **no!** | H264, H265 <br/> AAC, FLAC\* |
| iPhone Safari 14+ | H264, H265\* <br/> PCMU, PCMA <br/> OPUS | **no!** | **no!** | H264, H265 <br/> AAC, FLAC\* |
| macOS [Hass App][1] | no | no | no | H264, H265 <br/> AAC, FLAC\* |
| Device | WebRTC | MSE | HTTP* | HLS |
|--------------------------------------------------------------------------|-----------------------------------------|-----------------------------------------|----------------------------------------------|-----------------------------|
| *latency* | best | medium | bad | bad |
| - Desktop Chrome 107+ <br/> - Desktop Edge <br/> - Android Chrome 107+ | H264 <br/> PCMU, PCMA <br/> OPUS | H264, H265* <br/> AAC, FLAC* <br/> OPUS | H264, H265* <br/> AAC, FLAC* <br/> OPUS, MP3 | no |
| Desktop Firefox | H264 <br/> PCMU, PCMA <br/> OPUS | H264 <br/> AAC, FLAC* <br/> OPUS | H264 <br/> AAC, FLAC* <br/> OPUS | no |
| - Desktop Safari 14+ <br/> - iPad Safari 14+ <br/> - iPhone Safari 17.1+ | H264, H265* <br/> PCMU, PCMA <br/> OPUS | H264, H265 <br/> AAC, FLAC* | **no!** | H264, H265 <br/> AAC, FLAC* |
| iPhone Safari 14+ | H264, H265* <br/> PCMU, PCMA <br/> OPUS | **no!** | **no!** | H264, H265 <br/> AAC, FLAC* |
| macOS [Hass App][1] | no | no | no | H264, H265 <br/> AAC, FLAC* |
[1]: https://apps.apple.com/app/home-assistant/id1099568401
`HTTP*` - HTTP Progressive Streaming, not related with [Progressive download](https://en.wikipedia.org/wiki/Progressive_download), because the file has no size and no end
`HTTP*` - HTTP Progressive Streaming, not related with [Progressive download](https://en.wikipedia.org/wiki/Progressive_download), because the file has no size and no end
- Chrome H265: [read this](https://chromestatus.com/feature/5186511939567616) and [read this](https://github.com/StaZhu/enable-chromium-hevc-hardware-decoding)
- Edge H265: [read this](https://www.reddit.com/r/MicrosoftEdge/comments/v9iw8k/enable_hevc_support_in_edge/)