* Update version * Create scaffolding for case management (#21293) * implement case management for export apis (#21295) * refactor vainfo to search for first GPU (#21296) use existing LibvaGpuSelector to pick appropritate libva device * Case management UI (#21299) * Refactor export cards to match existing cards in other UI pages * Show cases separately from exports * Add proper filtering and display of cases * Add ability to edit and select cases for exports * Cleanup typing * Hide if no unassigned * Cleanup hiding logic * fix scrolling * Improve layout * Camera connection quality indicator (#21297) * add camera connection quality metrics and indicator * formatting * move stall calcs to watchdog * clean up * change watchdog to 1s and separately track time for ffmpeg retry_interval * implement status caching to reduce message volume * Export filter UI (#21322) * Get started on export filters * implement basic filter * Implement filtering and adjust api * Improve filter handling * Improve navigation * Cleanup * handle scrolling * Refactor temperature reporting for detectors and implement Hailo temp reading (#21395) * Add Hailo temperature retrieval * Refactor `get_hailo_temps()` to use ctxmanager * Show Hailo temps in system UI * Move hailo_platform import to get_hailo_temps * Refactor temperatures calculations to use within detector block * Adjust webUI to handle new location --------- Co-authored-by: tigattack <10629864+tigattack@users.noreply.github.com> * Camera-specific hwaccel settings for timelapse exports (correct base) (#21386) * added hwaccel_args to camera.record.export config struct * populate camera.record.export.hwaccel_args with a cascade up to camera then global if 'auto' * use new hwaccel args in export * added documentation for camera-specific hwaccel export * fix c/p error * missed an import * fleshed out the docs and comments a bit * ruff lint * separated out the tips in the doc * fix documentation * fix and simplify reference config doc * Add support for GPU and NPU temperatures (#21495) * Add rockchip temps * Add support for GPU and NPU temperatures in the frontend * Add support for Nvidia temperature * Improve separation * Adjust graph scaling * Exports Improvements (#21521) * Add images to case folder view * Add ability to select case in export dialog * Add to mobile review too * Add API to handle deleting recordings (#21520) * Add recording delete API * Re-organize recordings apis * Fix import * Consolidate query types * Add media sync API endpoint (#21526) * add media cleanup functions * add endpoint * remove scheduled sync recordings from cleanup * move to utils dir * tweak import * remove sync_recordings and add config migrator * remove sync_recordings * docs * remove key * clean up docs * docs fix * docs tweak * Media sync API refactor and UI (#21542) * generic job infrastructure * types and dispatcher changes for jobs * save data in memory only for completed jobs * implement media sync job and endpoints * change logs to debug * websocket hook and types * frontend * i18n * docs tweaks * endpoint descriptions * tweak docs * use same logging pattern in sync_recordings as the other sync functions (#21625) * Fix incorrect counting in sync_recordings (#21626) * Update go2rtc to v1.9.13 (#21648) Co-authored-by: Eugeny Tulupov <eugeny.tulupov@spirent.com> * Refactor Time-Lapse Export (#21668) * refactor time lapse creation to be a separate API call with ability to pass arbitrary ffmpeg args * Add CPU fallback * Optimize empty directory cleanup for recordings (#21695) The previous empty directory cleanup did a full recursive directory walk, which can be extremely slow. This new implementation only removes directories which have a chance of being empty due to a recent file deletion. * Implement llama.cpp GenAI Provider (#21690) * Implement llama.cpp GenAI Provider * Add docs * Update links * Fix broken mqtt links * Fix more broken anchors * Remove parents in remove_empty_directories (#21726) The original implementation did a full directory tree walk to find and remove empty directories, so this implementation should remove the parents as well, like the original did. * Implement LLM Chat API with tool calling support (#21731) * Implement initial tools definiton APIs * Add initial chat completion API with tool support * Implement other providers * Cleanup * Offline preview image (#21752) * use latest preview frame for latest image when camera is offline * remove frame extraction logic * tests * frontend * add description to api endpoint * Update to ROCm 7.2.0 (#21753) * Update to ROCm 7.2.0 * ROCm now works properly with JinaV1 * Arcface has compilation error * Add live context tool to LLM (#21754) * Add live context tool * Improve handling of images in request * Improve prompt caching * Add networking options for configuring listening ports (#21779) * feat: add X-Frame-Time when returning snapshot (#21932) Co-authored-by: Florent MORICONI <170678386+fmcloudconsulting@users.noreply.github.com> * Improve jsmpeg player websocket handling (#21943) * improve jsmpeg player websocket handling prevent websocket console messages from appearing when player is destroyed * reformat files after ruff upgrade * Allow API Events to be Detections or Alerts, depending on the Event Label (#21923) * - API created events will be alerts OR detections, depending on the event label, defaulting to alerts - Indefinite API events will extend the recording segment until those events are ended - API event start time is the actual start time, instead of having a pre-buffer of record.event_pre_capture * Instead of checking for indefinite events on a camera before deciding if we should end the segment, only update last_detection_time and last_alert_time if frame_time is greater, which should have the same effect * Add the ability to set a pre_capture number of seconds when creating a manual event via the API. Default behavior unchanged * Remove unnecessary _publish_segment_start() call * Formatting * handle last_alert_time or last_detection_time being None when checking them against the frame_time * comment manual_info["label"].split(": ")[0] for clarity * ffmpeg Preview Segment Optimization for "high" and "very_high" (#21996) * Introduce qmax parameter for ffmpeg preview encoding Added PREVIEW_QMAX_PARAM to control ffmpeg encoding quality. * formatting * Fix spacing in qmax parameters for preview quality * Adapt to new Gemini format * Fix frame time access * Remove exceptions * Cleanup --------- Co-authored-by: Josh Hawkins <32435876+hawkeye217@users.noreply.github.com> Co-authored-by: tigattack <10629864+tigattack@users.noreply.github.com> Co-authored-by: Andrew Roberts <adroberts@gmail.com> Co-authored-by: Eugeny Tulupov <zhekka3@gmail.com> Co-authored-by: Eugeny Tulupov <eugeny.tulupov@spirent.com> Co-authored-by: John Shaw <1753078+johnshaw@users.noreply.github.com> Co-authored-by: Eric Work <work.eric@gmail.com> Co-authored-by: FL42 <46161216+fl42@users.noreply.github.com> Co-authored-by: Florent MORICONI <170678386+fmcloudconsulting@users.noreply.github.com> Co-authored-by: nulledy <254504350+nulledy@users.noreply.github.com>
9.9 KiB
| id | title |
|---|---|
| restream | Restream |
RTSP
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. The video feed is copied from the original video feed directly to avoid re-encoding. This feed does not include any annotation by Frigate.
Frigate uses go2rtc to provide its restream and MSE/WebRTC capabilities. The go2rtc config is hosted at the go2rtc in the config, see go2rtc docs for more advanced configurations and features.
:::note
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.
:::
Birdseye Restream
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.
birdseye:
restream: True
:::tip
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.
:::
Securing Restream With Authentication
The go2rtc restream can be secured with RTSP based username / password authentication. Ex:
go2rtc:
rtsp:
username: "admin"
password: "pass"
streams: ...
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.
Reduce Connections To Camera
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.
With Single Stream
One connection is made to the camera. One for the restream, detect and record connect to the restream.
go2rtc:
streams:
name_your_rtsp_cam: # <- for RTSP streams
- rtsp://192.168.1.5:554/live0 # <- stream which supports video & aac audio
- "ffmpeg:name_your_rtsp_cam#audio=opus" # <- copy of the stream which transcodes audio to the missing codec (usually will be opus)
name_your_http_cam: # <- for other streams
- http://192.168.50.155/flv?port=1935&app=bcs&stream=channel0_main.bcs&user=user&password=password # <- stream which supports video & aac audio
- "ffmpeg:name_your_http_cam#audio=opus" # <- copy of the stream which transcodes audio to the missing codec (usually will be opus)
cameras:
name_your_rtsp_cam:
ffmpeg:
output_args:
record: preset-record-generic-audio-copy
inputs:
- path: rtsp://127.0.0.1:8554/name_your_rtsp_cam # <--- the name here must match the name of the camera in restream
input_args: preset-rtsp-restream
roles:
- record
- detect
- audio # <- only necessary if audio detection is enabled
name_your_http_cam:
ffmpeg:
output_args:
record: preset-record-generic-audio-copy
inputs:
- path: rtsp://127.0.0.1:8554/name_your_http_cam # <--- the name here must match the name of the camera in restream
input_args: preset-rtsp-restream
roles:
- record
- detect
- audio # <- only necessary if audio detection is enabled
With Sub Stream
Two connections are made to the camera. One for the sub stream, one for the restream, record connects to the restream.
go2rtc:
streams:
name_your_rtsp_cam:
- rtsp://192.168.1.5:554/live0 # <- stream which supports video & aac audio. This is only supported for rtsp streams, http must use ffmpeg
- "ffmpeg:name_your_rtsp_cam#audio=opus" # <- copy of the stream which transcodes audio to opus
name_your_rtsp_cam_sub:
- rtsp://192.168.1.5:554/substream # <- stream which supports video & aac audio. This is only supported for rtsp streams, http must use ffmpeg
- "ffmpeg:name_your_rtsp_cam_sub#audio=opus" # <- copy of the stream which transcodes audio to opus
name_your_http_cam:
- 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
- "ffmpeg:name_your_http_cam#audio=opus" # <- copy of the stream which transcodes audio to opus
name_your_http_cam_sub:
- 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
- "ffmpeg:name_your_http_cam_sub#audio=opus" # <- copy of the stream which transcodes audio to opus
cameras:
name_your_rtsp_cam:
ffmpeg:
output_args:
record: preset-record-generic-audio-copy
inputs:
- path: rtsp://127.0.0.1:8554/name_your_rtsp_cam # <--- 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/name_your_rtsp_cam_sub # <--- the name here must match the name of the camera_sub in restream
input_args: preset-rtsp-restream
roles:
- audio # <- only necessary if audio detection is enabled
- detect
name_your_http_cam:
ffmpeg:
output_args:
record: preset-record-generic-audio-copy
inputs:
- path: rtsp://127.0.0.1:8554/name_your_http_cam # <--- 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/name_your_http_cam_sub # <--- the name here must match the name of the camera_sub in restream
input_args: preset-rtsp-restream
roles:
- audio # <- only necessary if audio detection is enabled
- detect
Handling Complex Passwords
go2rtc expects URL-encoded passwords in the config, urlencoder.org can be used for this purpose.
For example:
go2rtc:
streams:
my_camera: rtsp://username:$@foo%@192.168.1.100
becomes
go2rtc:
streams:
my_camera: rtsp://username:$%40foo%25@192.168.1.100
See this comment for more information.
Preventing go2rtc from blocking two-way audio
For cameras that support two-way talk, go2rtc will automatically establish an audio output backchannel when connecting to an RTSP stream. This backchannel blocks access to the camera's audio output for two-way talk functionality, preventing both Frigate and other applications from using it.
To prevent this, you must configure two separate stream instances:
- One stream instance with
#backchannel=0for Frigate's viewing, recording, and detection (prevents go2rtc from establishing the blocking backchannel) - A second stream instance without
#backchannel=0for two-way talk functionality (can be used by Frigate's WebRTC viewer or other applications)
Configuration example:
go2rtc:
streams:
front_door:
- rtsp://user:password@10.0.10.10:554/cam/realmonitor?channel=1&subtype=2#backchannel=0
front_door_twoway:
- rtsp://user:password@10.0.10.10:554/cam/realmonitor?channel=1&subtype=2
In this configuration:
front_doorstream is used by Frigate for viewing, recording, and detection. The#backchannel=0parameter prevents go2rtc from establishing the audio output backchannel, so it won't block two-way talk access.front_door_twowaystream is used for two-way talk functionality. This stream can be used by Frigate's WebRTC viewer when two-way talk is enabled, or by other applications (like Home Assistant Advanced Camera Card) that need access to the camera's audio output channel.
Security: Restricted Stream Sources
For security reasons, the echo:, expr:, and exec: stream sources are disabled by default in go2rtc. These sources allow arbitrary command execution and can pose security risks if misconfigured.
If you attempt to use these sources in your configuration, the streams will be removed and an error message will be printed in the logs.
To enable these sources, you must set the environment variable GO2RTC_ALLOW_ARBITRARY_EXEC=true. This can be done in your Docker Compose file or container environment:
environment:
- GO2RTC_ALLOW_ARBITRARY_EXEC=true
:::warning
Enabling arbitrary exec sources allows execution of arbitrary commands through go2rtc stream configurations. Only enable this if you understand the security implications and trust all sources of your configuration.
:::
Advanced Restream Configurations
The exec source in go2rtc can be used for custom ffmpeg commands. An example is below:
:::warning
The exec:, echo:, and expr: sources are disabled by default for security. You must set GO2RTC_ALLOW_ARBITRARY_EXEC=true to use them. See Security: Restricted Stream Sources for more information.
:::
:::warning
The exec:, echo:, and expr: sources are disabled by default for security. You must set GO2RTC_ALLOW_ARBITRARY_EXEC=true to use them. See Security: Restricted Stream Sources for more information.
:::
NOTE: The output will need to be passed with two curly braces {{output}}
go2rtc:
streams:
stream1: exec:ffmpeg -hide_banner -re -stream_loop -1 -i /media/BigBuckBunny.mp4 -c copy -rtsp_transport tcp -f rtsp {{output}}