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.
You can access the go2rtc stream info at http://frigate_ip:5000/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
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.
RTMP (Deprecated)
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 use the built in go2rtc config for restreaming.
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
Advanced Restream Configurations
The 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}}
go2rtc:
streams:
stream1: exec:ffmpeg -hide_banner -re -stream_loop -1 -i /media/BigBuckBunny.mp4 -c copy -rtsp_transport tcp -f rtsp {{output}}