--- id: restream title: Restream --- ## RTSP Frigate can restream your video feed as an RTSP feed for other applications such as Home Assistant to utilize it at `rtsp://:8554/`. 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. Frigate uses [go2rtc](https://github.com/AlexxIT/go2rtc/tree/v1.2.0) 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/v1.2.0#configuration) for more advanced configurations and features. :::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 enabled at `birdseye -> restream` and accessed at `rtsp://:8554/birdseye`. Enabling the restream will cause birdseye to run 24/7 which may increase CPU usage somewhat. ### 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) 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: rtsp_cam: # <- for RTSP streams - rtsp://192.168.1.5:554/live0 # <- stream which supports video & aac audio - "ffmpeg:rtsp_cam#audio=opus" # <- copy of the stream which transcodes audio to the missing codec (usually will be opus) 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:http_cam#audio=opus" # <- copy of the stream which transcodes audio to the missing codec (usually will be opus) cameras: rtsp_cam: ffmpeg: output_args: record: preset-record-generic-audio-copy inputs: - path: rtsp://127.0.0.1:8554/rtsp_cam # <--- the name here must match the name of the camera in restream input_args: preset-rtsp-restream roles: - record - detect http_cam: ffmpeg: output_args: record: preset-record-generic-audio-copy inputs: - path: rtsp://127.0.0.1:8554/http_cam # <--- the name here must match the name of the camera in restream input_args: preset-rtsp-restream roles: - record - detect ``` ### 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: 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:rtsp_cam#audio=opus" # <- copy of the stream which transcodes audio to opus 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:rtsp_cam_sub#audio=opus" # <- copy of the stream which transcodes audio to opus 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:http_cam#audio=opus" # <- copy of the stream which transcodes audio to opus 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:http_cam_sub#audio=opus" # <- copy of the stream which transcodes audio to opus cameras: rtsp_cam: ffmpeg: output_args: record: preset-record-generic-audio-copy inputs: - path: rtsp://127.0.0.1:8554/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/rtsp_cam_sub # <--- the name here must match the name of the camera_sub in restream input_args: preset-rtsp-restream roles: - detect http_cam: ffmpeg: output_args: record: preset-record-generic-audio-copy inputs: - path: rtsp://127.0.0.1:8554/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/http_cam_sub # <--- the name here must match the name of the camera_sub in restream input_args: preset-rtsp-restream roles: - detect ``` ## Advanced Restream Configurations The [exec](https://github.com/AlexxIT/go2rtc/tree/v1.2.0#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}} ```