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go2rtc/README.md
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Alexey Khit 3e77835583 Initial commit
2022-08-18 09:19:00 +03:00

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# go2rtc
**go2rtc** - ultimate camera streaming application with support RTSP, WebRTC, FFmpeg, RTMP, etc.
- zero-dependency and zero-config small app for all OS (Windows, macOS, Linux, ARM, etc.)
- zero-delay for all supported protocols (lowest possible streaming latency)
- zero-load on CPU for supported codecs
- on the fly transcoding for unsupported codecs via FFmpeg
- multi-source two-way [codecs negotiation](#codecs-negotiation)
- streaming from private networks via Ngrok or SSH-tunnels
## Codecs negotiation
For example, you want to watch stream from [Dahua IPC-K42](https://www.dahuasecurity.com/fr/products/All-Products/Network-Cameras/Wireless-Series/Wi-Fi-Series/4MP/IPC-K42) camera in your browser.
- this camera support codecs **H264, H265** for send video, and you select `H264` in camera settings
- this camera support codecs **AAC, PCMU, PCMA** for send audio (from mic), and you select `AAC/16000` in camera settings
- this camera support codecs **AAC, PCMU, PCMA** for receive audio (to speaker), you don't need to select them
- your browser support codecs **H264, VP8, VP9, AV1** for receive video, you don't need to select them
- your browser support codecs **OPUS, PCMU, PCMA** for send and receive audio, you don't need to select them
- you can't get camera audio directly, because their audio codecs doesn't match with your browser codecs
- so you decide to use transcoding via FFmpeg and add this setting to config YAML file
- you have chosen `OPUS/48000/2` codec, because it is higher quality than the PCMU/8000 or PCMA/8000
- now you have stream with two sources - **RTSP and FFmpeg**
`go2rtc` automatically match codecs for you browser and all your stream sources. This called **multi-source two-way codecs negotiation**. And this is one of the main features of this app.
**PS.** You can select PCMU or PCMA codec in camera setting and don't use transcoding at all. Or you can select AAC codec for main stream and PCMU codec for second stream and add both RTSP to YAML config, this also will work fine.
```yaml
streams:
dahua:
- rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif
- ffmpeg:rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif#audio=opus
```
![](codecs.svg)
## Configuration
Create file `go2rtc.yaml` next to the app. Modules:
- [Streams](#streams)
### Streams
**go2rtc** support different stream source types. You can setup only one link as stream source or multiple.
- [RTSP/RTSPS](#rtsp-source) - most cameras on market
- [RTMP](#rtmp-source)
- [FFmpeg/Exec](#ffmpeg-source) - FFmpeg integration
- [Hass](#hass-source) - Home Assistant integration
#### RTSP source
- Support **RTSP and RTSPS** links with multiple video and audio tracks
- Support **2 way audio** ONLY for [ONVIF Profile T](https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf) cameras (back channel connection)
**Attention:** proprietary 2 way audio standards are not supported!
```yaml
streams:
rtsp_camera: rtsp://rtsp:12345678@192.168.1.123:554/av_stream/ch0
```
If your camera support two RTSP links - you can add both of them as sources. This is useful when streams has different codecs, as example AAC audio with main stream and PCMU/PCMA audio with second stream:
**Attention:** Dahua cameras has different capabilities for different RTSP links. For example, it has support multiple codecs for two way audio with `&proto=Onvif` in link and only one coded without it.
```yaml
streams:
onvif_camera:
- rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif
- rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=1
```
#### RTMP source
You can get stream from RTMP server, for example [Frigate](https://docs.frigate.video/configuration/rtmp). Support ONLY `H264` video codec without audio.
```yaml
streams:
rtmp_stream: rtmp://192.168.1.123/live/camera1
```
#### FFmpeg source
You can get any stream or file or device via FFmpeg and push it to go2rtc via RTSP protocol.
Format: `ffmpeg:{input}#{params}`. Examples:
```yaml
streams:
# [FILE] all tracks will be copied without transcoding codecs
file1: ffmpeg:~/media/BigBuckBunny.mp4
# [FILE] video will be transcoded to H264, audio will be skipped
file2: ffmpeg:~/media/BigBuckBunny.mp4#video=h264
# [FILE] video will be copied, audio will be transcoded to pcmu
file3: ffmpeg:~/media/BigBuckBunny.mp4#video=copy&audio=pcmu
# [HLS] video will be copied, audio will be skipped
hls: ffmpeg:https://devstreaming-cdn.apple.com/videos/streaming/examples/bipbop_16x9/gear5/prog_index.m3u8#video=copy
# [MJPEG] video will be transcoded to H264
mjpeg: ffmpeg:http://185.97.122.128/cgi-bin/faststream.jpg?stream=half&fps=15#video=h264
# [RTSP] video and audio will be copied
rtsp: ffmpeg:rtsp://rtsp:12345678@192.168.1.123:554/av_stream/ch0#video=copy&audio=copy
```
All trascoding formats has built-in templates. But you can override them via YAML config:
```yaml
ffmpeg:
bin: ffmpeg # path to ffmpeg binary
link: -hide_banner -i {input} # if input is link
file: -hide_banner -re -stream_loop -1 -i {input} # if input not link
rtsp: -hide_banner -fflags nobuffer -flags low_delay -rtsp_transport tcp -i {input} # if input is RTSP link
output: -rtsp_transport tcp -f rtsp {output} # output
h264: "-codec:v libx264 -g 30 -preset superfast -tune zerolatency -profile main -level 4.1"
h264/ultra: "-codec:v libx264 -g 30 -preset ultrafast -tune zerolatency"
h264/high: "-codec:v libx264 -g 30 -preset superfast -tune zerolatency"
h265: "-codec:v libx265 -g 30 -preset ultrafast -tune zerolatency"
opus: "-codec:a libopus -ar 48000 -ac 2"
pcmu: "-codec:a pcm_mulaw -ar 8000 -ac 1"
pcmu/16000: "-codec:a pcm_mulaw -ar 16000 -ac 1"
pcmu/48000: "-codec:a pcm_mulaw -ar 48000 -ac 1"
pcma: "-codec:a pcm_alaw -ar 8000 -ac 1"
pcma/16000: "-codec:a pcm_alaw -ar 16000 -ac 1"
pcma/48000: "-codec:a pcm_alaw -ar 48000 -ac 1"
aac/16000: "-codec:a aac -ar 16000 -ac 1"
```
#### Exec source
FFmpeg source just a shortcut to exec source. You can get any stream or file or device via FFmpeg or GStreamer and push it to go2rtc via RTSP protocol:
```yaml
streams:
stream1: exec:ffmpeg -hide_banner -re -stream_loop -1 -i ~/media/BigBuckBunny.mp4 -c copy -rtsp_transport tcp -f rtsp {output}
```
#### Hass source
Support import camera links from [Home Assistant](https://www.home-assistant.io/) config files.
- support ONLY [Generic Camera](https://www.home-assistant.io/integrations/generic/), setup via GUI
```yaml
hass:
config: "~/.homeassistant"
streams:
generic_camera: hass:Camera1 # Settings > Integrations > Integration Name
```
### API server
```yaml
api:
listen: ":3000" # HTTP API port
base_path: "" # API prefix for serve on suburl
static_dir: "www" # folder for static files
```
### RTSP server
```yaml
rtsp:
listen: ":554"
```
### WebRTC server
```yaml
webrtc:
listen: ":8555" # address of your local server (TCP)
candidates:
- 216.58.210.174:8555 # if you have static public IP-address
- 192.168.1.123:8555 # ip you have problems with UDP in LAN
- stun # if you have dynamic public IP-address (auto discovery via STUN)
ice_servers:
- urls: [stun:stun.l.google.com:19302]
- urls: [turn:123.123.123.123:3478]
username: your_user
credential: your_pass
```
### Ngrok
```yaml
ngrok:
command: ngrok tcp 8555 --authtoken eW91IHNoYWxsIG5vdCBwYXNzCnlvdSBzaGFsbCBub3QgcGFzcw
```
or
```yaml
ngrok:
command: ngrok start --all --config ngrok.yml
```
### Log
```yaml
log:
level: info # default level
api: trace
exec: debug
ngrok: info
rtsp: warn
streams: error
webrtc: fatal
```
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