Live View

Frigate has different live view options, some of which require the bundled go2rtc to be configured as shown in the step by step guide.

Live View Options

Live view options can be selected while viewing the live stream. The options are:

Source	Latency	Frame Rate	Resolution	Audio	Requires go2rtc	Other Limitations
jsmpeg	low	same as detect -> fps, capped at 10	same as detect	no	no	none
mse	low	native	native	yes (depends on audio codec)	yes	iPhone requires iOS 17.1+, Firefox is h.264 only
webrtc	lowest	native	native	yes (depends on audio codec)	yes	requires extra config, doesn't support h.265

Audio Support

MSE Requires AAC audio, WebRTC requires PCMU/PCMA, or opus audio. If you want to support both MSE and WebRTC then your restream config needs to make sure both are enabled.

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 http 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)

Setting Stream For Live UI

There may be some cameras that you would prefer to use the sub stream for live view, but the main stream for recording. This can be done via live -> stream_name.

go2rtc:
  streams:
    test_cam:
      - rtsp://192.168.1.5:554/live0 # <- stream which supports video & aac audio.
      - "ffmpeg:test_cam#audio=opus" # <- copy of the stream which transcodes audio to opus for webrtc
    test_cam_sub:
      - rtsp://192.168.1.5:554/substream # <- stream which supports video & aac audio.
      - "ffmpeg:test_cam_sub#audio=opus" # <- copy of the stream which transcodes audio to opus for webrtc

cameras:
  test_cam:
    ffmpeg:
      output_args:
        record: preset-record-generic-audio-copy
      inputs:
        - path: rtsp://127.0.0.1:8554/test_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/test_cam_sub # <--- the name here must match the name of the camera_sub in restream
          input_args: preset-rtsp-restream
          roles:
            - detect
    live:
      stream_name: test_cam_sub

WebRTC extra configuration:

WebRTC works by creating a TCP or UDP connection on port 8555. However, it requires additional configuration:

• For external access, over the internet, setup your router to forward port 8555 to port 8555 on the Frigate device, for both TCP and UDP.

• For internal/local access, unless you are running through the add-on, you will also need to set the WebRTC candidates list in the go2rtc config. For example, if 192.168.1.10 is the local IP of the device running Frigate:

  /config/frigate.yaml

  go2rtc:
    streams:
      test_cam: ...
    webrtc:
      candidates:
        - 192.168.1.10:8555
        - stun:8555

• For access through Tailscale, the Frigate system's Tailscale IP must be added as a WebRTC candidate. Tailscale IPs all start with 100., and are reserved within the 100.0.0.0/8 CIDR block.

tip

This extra configuration may not be required if Frigate has been installed as a Home Assistant add-on, as Frigate uses the Supervisor's API to generate a WebRTC candidate.

However, it is recommended if issues occur to define the candidates manually. You should do this if the Frigate add-on fails to generate a valid candidate. If an error occurs you will see some warnings like the below in the add-on logs page during the initialization:

[WARN] Failed to get IP address from supervisor
[WARN] Failed to get WebRTC port from supervisor

note

If you are having difficulties getting WebRTC to work and you are running Frigate with docker, you may want to try changing the container network mode:

• network: host, in this mode you don't need to forward any ports. The services inside of the Frigate container will have full access to the network interfaces of your host machine as if they were running natively and not in a container. Any port conflicts will need to be resolved. This network mode is recommended by go2rtc, but we recommend you only use it if necessary.
• network: bridge is the default network driver, a bridge network is a Link Layer device which forwards traffic between network segments. You need to forward any ports that you want to be accessible from the host IP.

If not running in host mode, port 8555 will need to be mapped for the container:

docker-compose.yml

services:
  frigate:
    ...
    ports:
      - "8555:8555/tcp" # WebRTC over tcp
      - "8555:8555/udp" # WebRTC over udp

See go2rtc WebRTC docs for more information about this.