advanced-camera-card/docs/configuration/cameras/live-provider.md

live_provider

Overview

The live_provider parameter determines what provides the live stream for a camera. Each provider offers different capabilities:

Live Provider	Latency	Frame Rate	Loading Time	Installation	Supports Proxying	Description
go2rtc	Best	High	Better	Builtin	✅	Uses go2rtc to stream live feeds and supports 2-way audio.
go2rtc-experimental	Best	High	Better	Builtin	✅	Experimental re-implementation of the go2rtc provider client (MSE/WebRTC/MP4/MJPEG) that is expected to eventually replace the go2rtc live provider. Uses the same configuration as go2rtc. See go2rtc (experimental).
ha (Native WebRTC)	Best	High	Better	Builtin	✖️	Use the built-in Home Assistant camera streams -- will offer a very low-latency feed direct to your browser.
ha (HLS)	Poor	High	Better	Builtin	✖️	Use the built-in Home Assistant camera streams -- HLS fallback when a WebRTC connection cannot be established.
ha (when configured with LL-HLS)	Better	High	Better	Builtin	✖️	Use the built-in Home Assistant camera streams -- can be configured to use an LL-HLS feed for lower latency.
image	Poor	Poor	Best	Builtin	✖️	Use refreshing snapshots of the built-in Home Assistant camera streams.
jsmpeg	Better	Low	Poor	Builtin	✖️	Use a the JSMPEG stream.
webrtc-card	Best	High	Better	Separate installation required	✖️	Embed's AlexxIT's WebRTC Card to stream live feed, requires manual extra setup. See webrtc_card. Not to be confused with native Home Assistant WebRTC (use the ha provider).

go2rtc

The go2rtc block configures use of the go2rtc live provider. This configuration is included as part of a camera entry in the cameras list.

cameras:
  - camera_entity: camera.office
    live_provider: go2rtc
    go2rtc:
      # [...]

Option	Default	Description
metadata_fetch_timeout_seconds	2	Timeout for the go2rtc stream metadata fetch (this is used to detect stream capabilities such as 2-way-audio).
modes	go2rtc default: [webrtc, mse, mp4, mjpeg]	An ordered list of go2rtc modes to use. Valid values are webrtc, mse, mp4 or mjpeg values.
stream	Determined by camera engine (e.g. frigate camera name).	A valid go2rtc stream name.
url	Determined by camera engine (e.g. the frigate engine will automatically generate a URL for the go2rtc backend that runs in the Frigate container).	The root go2rtc URL the card should stream the video from. This is only needed for non-Frigate usecases, or advanced Frigate usecases. Example: http://my-custom-go2rtc:1984

Note

If url is manually set and proxy.live is set to auto on the camera (the default), the video stream will automatically be proxied via the Home Assistant process if the hass-web-proxy-integration is detected. See proxying.

Tip

If you are certain your camera hardware supports 2-way audio but the microphone button is intermittently missing on load, try increasing metadata_fetch_timeout_seconds or use capabilities.force to skip metadata detection entirely.

go2rtc (experimental)

[!WARNING] go2rtc-experimental is a from-scratch re-implementation of the go2rtc live provider client (MSE, WebRTC, MP4 and MJPEG). It is under active development and is expected to eventually replace the go2rtc live provider. Please try it and report any issues you encounter.

It uses the exact same configuration as go2rtc: select it with live_provider: go2rtc-experimental and configure the same go2rtc block.

cameras:
  - camera_entity: camera.office
    live_provider: go2rtc-experimental
    go2rtc:
      # [...]

ha

The ha block configures use of the default Home Assistant (ha) live provider. It has no configuration options.

cameras:
  - camera_entity: camera.office
    live_provider: ha

The native stream provider in Home Assistant dynamically chooses between HLS and WebRTC streams. It prefers the lowest-latency stream (WebRTC) and only falls back to HLS when that is the only way to get audio.

WebRTC stream	HLS Stream	Muted?	Resulting Stream Selection
Has audio	Either	Either	WebRTC (lowest latency)
Has no audio	Has audio	Yes	WebRTC (lowest latency)
Has no audio	Has audio	No	HLS (for audio)
Has no audio	Has no audio	Either	WebRTC (lowest latency)

Note

When using the ha provider through Advanced Camera Card, streams are chosen by the same logic as the table above (the logic Home Assistant uses). The one difference is when the choice is made: the card re-runs the selection whenever you unmute (whether from the card's mute button or the video's own controls), whereas Home Assistant effectively fixes "muted" per dashboard card and never re-selects after that. So for a camera whose low-latency stream has no audio, unmuting in the card switches you to the audio-enabled stream, which may take a moment to load and run with a little more latency That audio-enabled stream will remain loaded thereafter.

image

All configuration is under:

cameras:
  - camera_entity: camera.office
    live_provider: image
    image:
      # [...]

Option	Default	Description
entity		The entity to use when mode is set to entity. This entity is expected to have an entity_picture attribute that specifies the image URL.
entity_parameters		Optional URL parameters to add to the URL generated for entity-based modes (i.e. when mode is camera or entity).
mode	auto	Value must be one of url (to fetch an arbitrary image URL), camera (to show a still of the currently selected camera entity using either camera_entity or webrtc_card.entity in that order of precedence), entity (to show an image associated with a named entity, see the entity parameter below), default (to show the default embedded image), or screensaver (to show a random image from picsum.photos, refreshing every 60s by default). If auto, the mode is chosen automatically based on whether url or entity parameters have been specified.
refresh_seconds	auto	The image will be refreshed at least every refresh_seconds (it may refresh more frequently, e.g. whenever Home Assistant updates its camera security token). 0 implies no refreshing. When set to auto, uses 1 for all modes except screensaver which uses 60.
url		A static image URL to be used when the mode is set to url or when a temporary image is required (e.g. may appear momentarily prior to load of a camera snapshot in the camera mode). Note that a _t=[timestamp] cache-busting value will be added automatically.

jsmpeg

All configuration is under:

cameras:
  - camera_entity: camera.office
    live_provider: jsmpeg
    jsmpeg:
      # [...]

Option	Default	Description
options		Advanced users only: Control the underlying JSMPEG library options. Supports setting these JSMPEG options {audio, video, pauseWhenHidden, disableGl, disableWebAssembly, preserveDrawingBuffer, progressive, throttled, chunkSize, maxAudioLag, videoBufferSize, audioBufferSize}. This is not necessary for the vast majority of users: only set these flags if you know what you're doing, as you may entirely break video rendering in the card.

webrtc_card

WebRTC Card support blends the use of the ultra-realtime WebRTC card live view with convenient access to Frigate events/snapshots/UI. AlexxIT's WebRTC Integration/Card must be installed and configured separately (see details) before it can be used with this card.

Note

The webrtc_card default configuration disables the WebRTC card's intersection parameter (which auto-stops the media when a certain fraction of the video is no longer visible), since it interferes with the card pan & zoom. Instead, see the auto_pause parameter.

cameras:
  - camera_entity: camera.office
    live_provider: webrtc-card
    webrtc_card:
      # [...]

Option	Default	Description
entity		The RTSP camera entity to pass to the WebRTC Card for this camera.
url	Depends on the camera engine (e.g. Frigate cameras will automatically use the camera name since this is the recommended setup).	The RTSP url to pass to the WebRTC Card, e.g. rtsp://USERNAME:PASSWORD@CAMERA:554/RTSP_PATH
*		Any options specified in the webrtc_card: YAML dictionary are silently passed through to the AlexxIT's WebRTC Card. See WebRTC Configuration for full details this external card provides, e.g. ui: true will enable the WebRTC Card UI.

Fully expanded reference

cameras:
  - camera_entity: camera.office_ha
    live_provider: ha
  - camera_entity: camera.office_webrtc_card
    live_provider: webrtc-card
    webrtc_card:
      entity: camera.office_rtsp
      url: 'rtsp://username:password@camera:554/av_stream/ch0'
      # Arbitrary WebRTC Card options, see https://github.com/AlexxIT/WebRTC#configuration .
      ui: true
  - camera_entity: camera.office_go2rtc
    live_provider: go2rtc
    go2rtc:
      modes:
        - webrtc
        - mse
        - mp4
        - mjpeg
      stream: sitting_room
      url: 'https://my.custom.go2rtc.backend'
      metadata_fetch_timeout_seconds: 2
  - camera_entity: camera.office_go2rtc_experimental
    live_provider: go2rtc-experimental
    go2rtc:
      modes:
        - webrtc
        - mse
        - mp4
        - mjpeg
      stream: sitting_room
      url: 'https://my.custom.go2rtc.backend'
      metadata_fetch_timeout_seconds: 2
  - camera_entity: camera.office_jsmpeg
    live_provider: jsmpeg
    jsmpeg:
      options:
        audio: false
        video: true
        pauseWhenHidden: false
        disableGl: false
        disableWebAssembly: false
        preserveDrawingBuffer: false
        progressive: true
        throttled: true
        chunkSize: 1048576
        maxAudioLag: 10
        videoBufferSize: 524288
        audioBufferSize: 131072
  - camera_entity: camera.office_image
    live_provider: image
    image:
      mode: auto
      refresh_seconds: auto
      url: 'https://path/to/image.png'
      entity: image.office_person
      entity_parameters: 'width=400&height=200'
cameras_global:
  live_provider: go2rtc