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Code Releases Activity

Rewrite restream (#5106)

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* Tear out restream config

* Rework birdseye restream

* Create go2rtc config handler

* Fix bug

* Write start script

* Rework style

* Fix python run syntax

* Output as json instead of yaml

* Put old live config back and fix birdseye references

* Fix camera webUI

* Add frigate env var subsitutions

* Fix webui checks

* Check keys

* Remove unused prest

* Fix tests

* Update restream docs

* Update restream docs

* Update live docs

* Update camera specific recommendation

* Update more docs

* add links for the docs

Co-authored-by: Felipe Santos <felipecassiors@gmail.com>

* Update note about supported audio codecs

* Move restream to go2rtc

* Docs fixes

* Add verification of stream name

* Ensure that webUI uses camera name

* Update docs to reflect new live stream name

* Fix check

* Formatting

* Remove audio from detect

Co-authored-by: Felipe Santos <felipecassiors@gmail.com>

* Fix docs

* Don't handle env variable substitution

* Add go2rtc version

* Clarify docs

Co-authored-by: Felipe Santos <felipecassiors@gmail.com>
This commit is contained in:
Nicolas Mowen
2023-01-16 17:50:35 -06:00
committed by GitHub
co-authored by Felipe Santos
parent a7751f210b
commit 19afb035ff
17 changed files with 212 additions and 406 deletions
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docs/docs/configuration/live.md
+75 -26
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@@ -9,44 +9,93 @@ Frigate has different live view options, some of which require [restream](restre
Live view options can be selected while viewing the live stream. The options are:
| Source | Latency | Frame Rate | Resolution | Audio | Requires Restream | Other Limitations |
| ------ | ------- | -------------------------------------- | -------------- | ---------------------------- | ----------------- | -------------------------------------------- |
| jsmpeg | low | same as `detect -> fps`, capped at 10 | same as detect | no | no | none |
| mse | low | native | native | yes (depends on audio codec) | yes | not supported on iOS, Firefox is h.264 only |
| webrtc | lowest | native | native | yes (depends on audio codec) | yes | requires extra config, doesn't support h.265 |
| Source | Latency | Frame Rate | Resolution | Audio | Requires Restream | Other Limitations |
| ------ | ------- | ------------------------------------- | -------------- | ---------------------------- | ----------------- | -------------------------------------------- |
| jsmpeg | low | same as `detect -> fps`, capped at 10 | same as detect | no | no | none |
| mse | low | native | native | yes (depends on audio codec) | yes | not supported on iOS, Firefox is h.264 only |
| webrtc | lowest | native | native | yes (depends on audio codec) | yes | requires extra config, doesn't support h.265 |
### Audio Support
MSE Requires AAC audio, WebRTC requires PCMU/PCMA, or opus audio. If you want to support both MSE and WebRTC then your restream config needs to use ffmpeg to set both.
```yaml
go2rtc:
streams:
test_cam: ffmpeg:rtsp://192.168.1.5:554/live0#video=copy#audio=aac#audio=opus
```
However, chances are that your camera already provides at least one usable audio type, so you just need restream to add the missing one. For example, if your camera outputs audio in AAC format:
```yaml
go2rtc:
streams:
test_cam: ffmpeg:rtsp://192.168.1.5:554/live0#video=copy#audio=copy#audio=opus
```
Which will reuse your camera AAC audio, while also adding one track in OPUS format.
If your camera uses RTSP and supports the audio type for the live view you want to use, then you can pass the camera stream to go2rtc without ffmpeg.
```yaml
go2rtc:
streams:
test_cam: rtsp://192.168.1.5:554/live0
```
### Setting Stream For Live UI
There may be some cameras that you would prefer to use the sub stream for live view, but the main stream for recording. This can be done via `live -> stream_name`.
```yaml
go2rtc:
streams:
test_cam: ffmpeg:rtsp://192.168.1.5:554/live0#video=copy#audio=aac#audio=opus
test_cam_sub: ffmpeg:rtsp://192.168.1.5:554/substream#video=copy#audio=aac#audio=opus
cameras:
test_cam:
ffmpeg:
output_args:
record: preset-record-generic-audio-copy
inputs:
- path: rtsp://127.0.0.1:8554/test_cam?video=copy&audio=aac # <--- the name here must match the name of the camera in restream
input_args: preset-rtsp-restream
roles:
- record
- path: rtsp://127.0.0.1:8554/test_cam_sub?video=copy # <--- the name here must match the name of the camera_sub in restream
input_args: preset-rtsp-restream
roles:
- detect
live:
stream_name: test_cam_sub
```
### WebRTC extra configuration:
WebRTC works by creating a TCP or UDP connection on port `8555`. However, it requires additional configuration:
* For external access, over the internet, setup your router to forward port `8555` to port `8555` on the Frigate device, for both TCP and UDP.
* For internal/local access, you will need to use a custom go2rtc config:
1. Create your own go2rtc config, based on [Frigate's internal go2rtc config](https://github.com/blakeblackshear/frigate/blob/dev/docker/rootfs/usr/local/go2rtc/go2rtc.yaml).
2. Add your internal IP to the list of `candidates`. Here is an example, assuming that `192.168.1.10` is the local IP of the device running Frigate:
- For external access, over the internet, setup your router to forward port `8555` to port `8555` on the Frigate device, for both TCP and UDP.
- For internal/local access, you will need to use a custom go2rtc config:
```yaml title="/config/frigate-go2rtc.yaml"
log:
format: text
1. Add your internal IP to the list of `candidates`. Here is an example, assuming that `192.168.1.10` is the local IP of the device running Frigate:
webrtc:
candidates:
- 192.168.1.10:8555
- stun:8555
```
3. Mount this config file at `/config/frigate-go2rtc.yaml`. Here is an example, if you run Frigate through docker-compose:
```yaml title="docker-compose.yaml"
volumes:
- /path/to/your/go2rtc.yaml:/config/frigate-go2rtc.yaml
```
```yaml
go2rtc:
streams:
test_cam: ...
webrtc:
candidates:
- 192.168.1.10:8555
- stun:8555
```
:::note
If you are having difficulties getting WebRTC to work and you are running Frigate with docker, you may want to try changing the container network mode:
* `network: host`, in this mode you don't need to forward any ports. The services inside of the Frigate container will have full access to the network interfaces of your host machine as if they were running natively and not in a container. Any port conflicts will need to be resolved. This network mode is recommended by go2rtc, but we recommend you only use it if necessary.
* `network: bridge` creates a virtual network interface for the container, and the container will have full access to it. You also don't need to forward any ports, however, the IP for accessing Frigate locally will differ from the IP of the host machine. Your router will see Frigate as if it was a new device connected in the network.
- `network: host`, in this mode you don't need to forward any ports. The services inside of the Frigate container will have full access to the network interfaces of your host machine as if they were running natively and not in a container. Any port conflicts will need to be resolved. This network mode is recommended by go2rtc, but we recommend you only use it if necessary.
- `network: bridge` creates a virtual network interface for the container, and the container will have full access to it. You also don't need to forward any ports, however, the IP for accessing Frigate locally will differ from the IP of the host machine. Your router will see Frigate as if it was a new device connected in the network.
:::