frigate/docs/docs/configuration/restream.md

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Add go2rtc and add restream role / live source (#4082) * Pull go2rtc dependency * Add go2rtc to local services and add to s6 * Add relay controller for go2rtc * Add restream role * Add restream role * Add restream to nginx * Add camera live source config * Disable RTMP by default and use restream * Use go2rtc for camera config * Fix go2rtc move * Start restream on frigate start * Send restream to camera level * Fix restream * Make sure jsmpeg works as expected * Make view rspect live size config * Tweak player options to fit live view * Adjust VideoPlayer to accept live option which disables irrelevant controls * Add multiple options from restream live view * Add base for webrtc option * Setup specific restream modules * Make mp4 the default streaming for now * Expose 8554 for rtsp relay from go2rtc * Formatting * Update docs to suggest new restream method. * Update docs to reflect restream role * Update docs to reflect restream role * Add webrtc player * Improvements to webRTC * Support webrtc * Cleanup * Adjust rtmp test and add restream test * Fix tests * Add restream tests * Add live view docs and show different options * Small docs tweak * Support all stream types * Update to beta 9 of go2rtc * Formatting * Make jsmpeg the default * Support wss if made from https * Support wss if made from https * Use onEffect * Set url outside onEffect * Fix passed deps * Update docs about required host mode * Try memo instead * Close websocket on changing camera * Formatting * Close pc connection * Set video source to null on cleanup * Use full path since go2rtc can't see PATH var * Adjust audio codec to enable browser audio by default * Cleanup stream creation * Add restream tests * Format tests * Mock requests * Adjust paths * Move stream configs to restream * Remove live source * Remove live config * Use live persistence for which view to use on each camera * Fix live sizes * Only use jsmpeg sizes for jsmpeg live * Set max live size * Remove access of live config * Add selector for live view source in web view * Remove RTMP from default list of roles * Update docs * Fix tests * Fix docs for live view modes * make default undefined to avoid race condition * Wait until camera source is loaded to avoid race condition * Fix tests * Add config to go2rtc * Work with config * Set full path for config * Set to use stun * Check for mounted file * Look for frigate-go2rtc * Update docs to reflect webRTC configuration. * Add link to go2rtc config * Update docs to be more clear * Update docs to be more clear * Update format Co-authored-by: Felipe Santos <felipecassiors@gmail.com> * Update live docs * Improve bash startup script * Add option to force audio compatibility * Formatting * Fix mapping * Fix broken link * Update go2rtc version * Get go2rtc webui working * Add support for mse * Remove mp4 option * Undo changes to video player * Update docs for new live view options * Make separate path for mse * Remove unused * Remove mp4 path * Try to get go2rtc proxy working * Try to get go2rtc proxy working * Remove unused callback * Allow websocket on restrea dashboard * Make mse default stream option * Fix mse sizing * don't assume roles is defined * Remove nginx mapping to go2rtc ui Co-authored-by: Felipe Santos <felipecassiors@gmail.com> Co-authored-by: Blake Blackshear <blakeb@blakeshome.com>
2022-11-02 14:36:09 +03:00
---
id: restream
title: Restream
---
## RTSP
Add go2rtc and add restream role / live source (#4082) * Pull go2rtc dependency * Add go2rtc to local services and add to s6 * Add relay controller for go2rtc * Add restream role * Add restream role * Add restream to nginx * Add camera live source config * Disable RTMP by default and use restream * Use go2rtc for camera config * Fix go2rtc move * Start restream on frigate start * Send restream to camera level * Fix restream * Make sure jsmpeg works as expected * Make view rspect live size config * Tweak player options to fit live view * Adjust VideoPlayer to accept live option which disables irrelevant controls * Add multiple options from restream live view * Add base for webrtc option * Setup specific restream modules * Make mp4 the default streaming for now * Expose 8554 for rtsp relay from go2rtc * Formatting * Update docs to suggest new restream method. * Update docs to reflect restream role * Update docs to reflect restream role * Add webrtc player * Improvements to webRTC * Support webrtc * Cleanup * Adjust rtmp test and add restream test * Fix tests * Add restream tests * Add live view docs and show different options * Small docs tweak * Support all stream types * Update to beta 9 of go2rtc * Formatting * Make jsmpeg the default * Support wss if made from https * Support wss if made from https * Use onEffect * Set url outside onEffect * Fix passed deps * Update docs about required host mode * Try memo instead * Close websocket on changing camera * Formatting * Close pc connection * Set video source to null on cleanup * Use full path since go2rtc can't see PATH var * Adjust audio codec to enable browser audio by default * Cleanup stream creation * Add restream tests * Format tests * Mock requests * Adjust paths * Move stream configs to restream * Remove live source * Remove live config * Use live persistence for which view to use on each camera * Fix live sizes * Only use jsmpeg sizes for jsmpeg live * Set max live size * Remove access of live config * Add selector for live view source in web view * Remove RTMP from default list of roles * Update docs * Fix tests * Fix docs for live view modes * make default undefined to avoid race condition * Wait until camera source is loaded to avoid race condition * Fix tests * Add config to go2rtc * Work with config * Set full path for config * Set to use stun * Check for mounted file * Look for frigate-go2rtc * Update docs to reflect webRTC configuration. * Add link to go2rtc config * Update docs to be more clear * Update docs to be more clear * Update format Co-authored-by: Felipe Santos <felipecassiors@gmail.com> * Update live docs * Improve bash startup script * Add option to force audio compatibility * Formatting * Fix mapping * Fix broken link * Update go2rtc version * Get go2rtc webui working * Add support for mse * Remove mp4 option * Undo changes to video player * Update docs for new live view options * Make separate path for mse * Remove unused * Remove mp4 path * Try to get go2rtc proxy working * Try to get go2rtc proxy working * Remove unused callback * Allow websocket on restrea dashboard * Make mse default stream option * Fix mse sizing * don't assume roles is defined * Remove nginx mapping to go2rtc ui Co-authored-by: Felipe Santos <felipecassiors@gmail.com> Co-authored-by: Blake Blackshear <blakeb@blakeshome.com>
2022-11-02 14:36:09 +03:00
2023-01-13 16:18:15 +03:00
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.
Add go2rtc and add restream role / live source (#4082) * Pull go2rtc dependency * Add go2rtc to local services and add to s6 * Add relay controller for go2rtc * Add restream role * Add restream role * Add restream to nginx * Add camera live source config * Disable RTMP by default and use restream * Use go2rtc for camera config * Fix go2rtc move * Start restream on frigate start * Send restream to camera level * Fix restream * Make sure jsmpeg works as expected * Make view rspect live size config * Tweak player options to fit live view * Adjust VideoPlayer to accept live option which disables irrelevant controls * Add multiple options from restream live view * Add base for webrtc option * Setup specific restream modules * Make mp4 the default streaming for now * Expose 8554 for rtsp relay from go2rtc * Formatting * Update docs to suggest new restream method. * Update docs to reflect restream role * Update docs to reflect restream role * Add webrtc player * Improvements to webRTC * Support webrtc * Cleanup * Adjust rtmp test and add restream test * Fix tests * Add restream tests * Add live view docs and show different options * Small docs tweak * Support all stream types * Update to beta 9 of go2rtc * Formatting * Make jsmpeg the default * Support wss if made from https * Support wss if made from https * Use onEffect * Set url outside onEffect * Fix passed deps * Update docs about required host mode * Try memo instead * Close websocket on changing camera * Formatting * Close pc connection * Set video source to null on cleanup * Use full path since go2rtc can't see PATH var * Adjust audio codec to enable browser audio by default * Cleanup stream creation * Add restream tests * Format tests * Mock requests * Adjust paths * Move stream configs to restream * Remove live source * Remove live config * Use live persistence for which view to use on each camera * Fix live sizes * Only use jsmpeg sizes for jsmpeg live * Set max live size * Remove access of live config * Add selector for live view source in web view * Remove RTMP from default list of roles * Update docs * Fix tests * Fix docs for live view modes * make default undefined to avoid race condition * Wait until camera source is loaded to avoid race condition * Fix tests * Add config to go2rtc * Work with config * Set full path for config * Set to use stun * Check for mounted file * Look for frigate-go2rtc * Update docs to reflect webRTC configuration. * Add link to go2rtc config * Update docs to be more clear * Update docs to be more clear * Update format Co-authored-by: Felipe Santos <felipecassiors@gmail.com> * Update live docs * Improve bash startup script * Add option to force audio compatibility * Formatting * Fix mapping * Fix broken link * Update go2rtc version * Get go2rtc webui working * Add support for mse * Remove mp4 option * Undo changes to video player * Update docs for new live view options * Make separate path for mse * Remove unused * Remove mp4 path * Try to get go2rtc proxy working * Try to get go2rtc proxy working * Remove unused callback * Allow websocket on restrea dashboard * Make mse default stream option * Fix mse sizing * don't assume roles is defined * Remove nginx mapping to go2rtc ui Co-authored-by: Felipe Santos <felipecassiors@gmail.com> Co-authored-by: Blake Blackshear <blakeb@blakeshome.com>
2022-11-02 14:36:09 +03:00
Frigate uses [go2rtc](https://github.com/AlexxIT/go2rtc/tree/v1.8.1) 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/v1.8.1#configuration) for more advanced configurations and features.
2023-02-09 06:26:38 +03:00
:::note
You can access the go2rtc webUI at `http://frigate_ip:5000/live/webrtc` 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.
```yaml
birdseye:
restream: true
```
### Securing Restream With Authentication
The go2rtc restream can be secured with RTSP based username / password authentication. Ex:
```yaml
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.
## RTMP (Deprecated)
Add go2rtc and add restream role / live source (#4082) * Pull go2rtc dependency * Add go2rtc to local services and add to s6 * Add relay controller for go2rtc * Add restream role * Add restream role * Add restream to nginx * Add camera live source config * Disable RTMP by default and use restream * Use go2rtc for camera config * Fix go2rtc move * Start restream on frigate start * Send restream to camera level * Fix restream * Make sure jsmpeg works as expected * Make view rspect live size config * Tweak player options to fit live view * Adjust VideoPlayer to accept live option which disables irrelevant controls * Add multiple options from restream live view * Add base for webrtc option * Setup specific restream modules * Make mp4 the default streaming for now * Expose 8554 for rtsp relay from go2rtc * Formatting * Update docs to suggest new restream method. * Update docs to reflect restream role * Update docs to reflect restream role * Add webrtc player * Improvements to webRTC * Support webrtc * Cleanup * Adjust rtmp test and add restream test * Fix tests * Add restream tests * Add live view docs and show different options * Small docs tweak * Support all stream types * Update to beta 9 of go2rtc * Formatting * Make jsmpeg the default * Support wss if made from https * Support wss if made from https * Use onEffect * Set url outside onEffect * Fix passed deps * Update docs about required host mode * Try memo instead * Close websocket on changing camera * Formatting * Close pc connection * Set video source to null on cleanup * Use full path since go2rtc can't see PATH var * Adjust audio codec to enable browser audio by default * Cleanup stream creation * Add restream tests * Format tests * Mock requests * Adjust paths * Move stream configs to restream * Remove live source * Remove live config * Use live persistence for which view to use on each camera * Fix live sizes * Only use jsmpeg sizes for jsmpeg live * Set max live size * Remove access of live config * Add selector for live view source in web view * Remove RTMP from default list of roles * Update docs * Fix tests * Fix docs for live view modes * make default undefined to avoid race condition * Wait until camera source is loaded to avoid race condition * Fix tests * Add config to go2rtc * Work with config * Set full path for config * Set to use stun * Check for mounted file * Look for frigate-go2rtc * Update docs to reflect webRTC configuration. * Add link to go2rtc config * Update docs to be more clear * Update docs to be more clear * Update format Co-authored-by: Felipe Santos <felipecassiors@gmail.com> * Update live docs * Improve bash startup script * Add option to force audio compatibility * Formatting * Fix mapping * Fix broken link * Update go2rtc version * Get go2rtc webui working * Add support for mse * Remove mp4 option * Undo changes to video player * Update docs for new live view options * Make separate path for mse * Remove unused * Remove mp4 path * Try to get go2rtc proxy working * Try to get go2rtc proxy working * Remove unused callback * Allow websocket on restrea dashboard * Make mse default stream option * Fix mse sizing * don't assume roles is defined * Remove nginx mapping to go2rtc ui Co-authored-by: Felipe Santos <felipecassiors@gmail.com> Co-authored-by: Blake Blackshear <blakeb@blakeshome.com>
2022-11-02 14:36:09 +03:00
In previous Frigate versions RTMP was used for re-streaming. RTMP has disadvantages however including being incompatible with H.265, high bitrates, and certain audio codecs. RTMP is deprecated and it is recommended to move to the new restream role.
## 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.
```yaml
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.
```yaml
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
```
## Advanced Restream Configurations
The [exec](https://github.com/AlexxIT/go2rtc/tree/v1.8.1#source-exec) source in go2rtc can be used for custom ffmpeg commands. An example is below:
NOTE: The output will need to be passed with two curly braces `{{output}}`
```yaml
go2rtc:
streams:
stream1: exec:ffmpeg -hide_banner -re -stream_loop -1 -i /media/BigBuckBunny.mp4 -c copy -rtsp_transport tcp -f rtsp {{output}}
```