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probe webrtc before reporting it as not configured

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The WebRTC gate read only the Frigate config, so an install with no go2rtc candidates showed WebRTC as not configured, even though go2rtc fills in default candidates at startup (including the internal one under the HA app) and WebRTC often worked. Unset candidates now run the connectivity probe first, and only a failed probe reports not configured. The not-configured reason shows as a neutral note that names the setting and says WebRTC is optional, rather than as a red error.
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Josh Hawkins
2026-10-15 21:16:44 -05:00
parent 3cd4c9c55c
commit 87a7551aaa
5 changed files with 106 additions and 12 deletions
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docs/docs/configuration/live.md
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@@ -38,7 +38,7 @@ Frigate [defaults to MSE](#why-does-frigate-prefer-mse-over-webrtc-for-live-view
- The choices are saved **per device, per camera** in your browser's local storage.
- **WebRTC is only selectable when it can actually work for that stream.** When it can't, the option is shown disabled with the reason inline, and a more detailed reason (the failing codecs, or why the connectivity check failed) is logged to your browser's console. Common reasons:
- **Not configured**: no `candidates` or `ice_servers` are set under `go2rtc.webrtc` (see [WebRTC extra configuration](#webrtc-extra-configuration)).
- **Not configured**: no `candidates` or `ice_servers` are set under `go2rtc.webrtc`, and the connectivity check failed with go2rtc's default candidates (see [WebRTC extra configuration](#webrtc-extra-configuration)). This is a setup note rather than an error, since live view works without WebRTC.
- **Could not connect**: e.g. port `8555` isn't reachable, or a STUN/TURN server is misconfigured. Frigate runs a one-time WebRTC connectivity check when the Live view opens; the option may briefly show as "checking" while it runs.
- **Unsupported video codec**: the stream's video codec can't be played over WebRTC in your browser, most commonly H.265/HEVC in Firefox or Edge.
- **Unsupported audio codec**: WebRTC needs opus or G.711 audio, so a stream whose playback audio is only AAC (without an added opus/G.711 track) can't carry audio over WebRTC. See [Audio Support](#audio-support) for how to add one.