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* Add optional idle heartbeat for Birdseye (periodic frame emission when idle) birdseye: add optional idle heartbeat and FFmpeg tuning envs (default off) This adds an optional configuration field `birdseye.idle_heartbeat_fps` to enable a lightweight idle heartbeat mechanism in Birdseye. When set to a value greater than 0, Birdseye periodically re-sends the last composed frame during idle periods (no motion or active updates). This helps downstream consumers such as go2rtc, Alexa, or Scrypted to attach faster and maintain a low-latency RTSP stream when the system is idle. Key details: - Config-based (`birdseye.idle_heartbeat_fps`), default `0` (disabled). - Uses existing Birdseye rendering pipeline; minimal performance impact. - Does not alter behavior when unset. Documentation: added tip section in docs/configuration/restream.md. * Update docs/docs/configuration/restream.md Co-authored-by: Nicolas Mowen <nickmowen213@gmail.com> * Update docs/docs/configuration/reference.md Co-authored-by: Nicolas Mowen <nickmowen213@gmail.com> * Refactors Birdseye idle frame broadcasting Simplifies the idle frame broadcasting logic by removing the dedicated thread. The idle frame is now resent directly within the main loop, improving efficiency and reducing complexity. Also, limits the idle heartbeat FPS to a maximum of 10 since the framebuffer is limited to 10 anyway * ruff fix --------- Co-authored-by: Nicolas Mowen <nickmowen213@gmail.com> Co-authored-by: Francesco Durighetto <francesco.durighetto@subbyx.com> Co-authored-by: duri <duri@homelabubuntu.durihome.unifi>
172 lines
7.1 KiB
Markdown
172 lines
7.1 KiB
Markdown
---
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id: restream
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title: Restream
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---
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## RTSP
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Frigate can restream your video feed as an RTSP feed for other applications such as Home Assistant to utilize it at `rtsp://<frigate_host>:8554/<camera_name>`. Port 8554 must be open. [This allows you to use a video feed for detection in Frigate and Home Assistant live view at the same time without having to make two separate connections to the camera](#reduce-connections-to-camera). The video feed is copied from the original video feed directly to avoid re-encoding. This feed does not include any annotation by Frigate.
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Frigate uses [go2rtc](https://github.com/AlexxIT/go2rtc/tree/v1.9.10) to provide its restream and MSE/WebRTC capabilities. The go2rtc config is hosted at the `go2rtc` in the config, see [go2rtc docs](https://github.com/AlexxIT/go2rtc/tree/v1.9.10#configuration) for more advanced configurations and features.
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:::note
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You can access the go2rtc stream info at `/api/go2rtc/streams` which can be helpful to debug as well as provide useful information about your camera streams.
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:::
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### Birdseye Restream
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Birdseye RTSP restream can be accessed at `rtsp://<frigate_host>:8554/birdseye`. Enabling the birdseye restream will cause birdseye to run 24/7 which may increase CPU usage somewhat.
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```yaml
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birdseye:
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restream: True
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```
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:::tip
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To improve connection speed when using Birdseye via restream you can enable a small idle heartbeat by setting `birdseye.idle_heartbeat_fps` to a low value (e.g. `1–2`). This makes Frigate periodically push the last frame even when no motion is detected, reducing initial connection latency.
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:::
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### Securing Restream With Authentication
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The go2rtc restream can be secured with RTSP based username / password authentication. Ex:
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```yaml
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go2rtc:
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rtsp:
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username: "admin"
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password: "pass"
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streams: ...
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```
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**NOTE:** This does not apply to localhost requests, there is no need to provide credentials when using the restream as a source for frigate cameras.
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## Reduce Connections To Camera
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Some cameras only support one active connection or you may just want to have a single connection open to the camera. The RTSP restream allows this to be possible.
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### With Single Stream
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One connection is made to the camera. One for the restream, `detect` and `record` connect to the restream.
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```yaml
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go2rtc:
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streams:
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name_your_rtsp_cam: # <- for RTSP streams
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- rtsp://192.168.1.5:554/live0 # <- stream which supports video & aac audio
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- "ffmpeg:name_your_rtsp_cam#audio=opus" # <- copy of the stream which transcodes audio to the missing codec (usually will be opus)
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name_your_http_cam: # <- for other streams
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- http://192.168.50.155/flv?port=1935&app=bcs&stream=channel0_main.bcs&user=user&password=password # <- stream which supports video & aac audio
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- "ffmpeg:name_your_http_cam#audio=opus" # <- copy of the stream which transcodes audio to the missing codec (usually will be opus)
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cameras:
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name_your_rtsp_cam:
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ffmpeg:
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output_args:
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record: preset-record-generic-audio-copy
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inputs:
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- path: rtsp://127.0.0.1:8554/name_your_rtsp_cam # <--- the name here must match the name of the camera in restream
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input_args: preset-rtsp-restream
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roles:
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- record
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- detect
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- audio # <- only necessary if audio detection is enabled
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name_your_http_cam:
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ffmpeg:
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output_args:
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record: preset-record-generic-audio-copy
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inputs:
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- path: rtsp://127.0.0.1:8554/name_your_http_cam # <--- the name here must match the name of the camera in restream
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input_args: preset-rtsp-restream
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roles:
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- record
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- detect
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- audio # <- only necessary if audio detection is enabled
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```
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### With Sub Stream
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Two connections are made to the camera. One for the sub stream, one for the restream, `record` connects to the restream.
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```yaml
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go2rtc:
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streams:
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name_your_rtsp_cam:
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- rtsp://192.168.1.5:554/live0 # <- stream which supports video & aac audio. This is only supported for rtsp streams, http must use ffmpeg
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- "ffmpeg:name_your_rtsp_cam#audio=opus" # <- copy of the stream which transcodes audio to opus
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name_your_rtsp_cam_sub:
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- rtsp://192.168.1.5:554/substream # <- stream which supports video & aac audio. This is only supported for rtsp streams, http must use ffmpeg
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- "ffmpeg:name_your_rtsp_cam_sub#audio=opus" # <- copy of the stream which transcodes audio to opus
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name_your_http_cam:
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- http://192.168.50.155/flv?port=1935&app=bcs&stream=channel0_main.bcs&user=user&password=password # <- stream which supports video & aac audio. This is only supported for rtsp streams, http must use ffmpeg
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- "ffmpeg:name_your_http_cam#audio=opus" # <- copy of the stream which transcodes audio to opus
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name_your_http_cam_sub:
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- http://192.168.50.155/flv?port=1935&app=bcs&stream=channel0_ext.bcs&user=user&password=password # <- stream which supports video & aac audio. This is only supported for rtsp streams, http must use ffmpeg
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- "ffmpeg:name_your_http_cam_sub#audio=opus" # <- copy of the stream which transcodes audio to opus
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cameras:
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name_your_rtsp_cam:
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ffmpeg:
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output_args:
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record: preset-record-generic-audio-copy
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inputs:
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- path: rtsp://127.0.0.1:8554/name_your_rtsp_cam # <--- the name here must match the name of the camera in restream
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input_args: preset-rtsp-restream
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roles:
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- record
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- path: rtsp://127.0.0.1:8554/name_your_rtsp_cam_sub # <--- the name here must match the name of the camera_sub in restream
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input_args: preset-rtsp-restream
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roles:
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- audio # <- only necessary if audio detection is enabled
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- detect
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name_your_http_cam:
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ffmpeg:
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output_args:
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record: preset-record-generic-audio-copy
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inputs:
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- path: rtsp://127.0.0.1:8554/name_your_http_cam # <--- the name here must match the name of the camera in restream
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input_args: preset-rtsp-restream
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roles:
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- record
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- path: rtsp://127.0.0.1:8554/name_your_http_cam_sub # <--- the name here must match the name of the camera_sub in restream
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input_args: preset-rtsp-restream
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roles:
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- audio # <- only necessary if audio detection is enabled
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- detect
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```
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## Handling Complex Passwords
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go2rtc expects URL-encoded passwords in the config, [urlencoder.org](https://urlencoder.org) can be used for this purpose.
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For example:
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```yaml
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go2rtc:
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streams:
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my_camera: rtsp://username:$@foo%@192.168.1.100
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```
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becomes
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```yaml
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go2rtc:
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streams:
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my_camera: rtsp://username:$%40foo%25@192.168.1.100
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```
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See [this comment](https://github.com/AlexxIT/go2rtc/issues/1217#issuecomment-2242296489) for more information.
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## Advanced Restream Configurations
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The [exec](https://github.com/AlexxIT/go2rtc/tree/v1.9.10#source-exec) source in go2rtc can be used for custom ffmpeg commands. An example is below:
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NOTE: The output will need to be passed with two curly braces `{{output}}`
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```yaml
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go2rtc:
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streams:
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stream1: exec:ffmpeg -hide_banner -re -stream_loop -1 -i /media/BigBuckBunny.mp4 -c copy -rtsp_transport tcp -f rtsp {{output}}
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``` |