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Update live view docs with camera firmware settings recommendations
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@ -17,6 +17,16 @@ The jsmpeg live view will use more browser and client GPU resources. Using go2rt
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| mse | native | native | yes (depends on audio codec) | yes | iPhone requires iOS 17.1+, Firefox is h.264 only. This is Frigate's default when go2rtc is configured. |
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| mse | native | native | yes (depends on audio codec) | yes | iPhone requires iOS 17.1+, Firefox is h.264 only. This is Frigate's default when go2rtc is configured. |
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| webrtc | native | native | yes (depends on audio codec) | yes | Requires extra configuration, doesn't support h.265. Frigate attempts to use WebRTC when MSE fails or when using a camera's two-way talk feature. |
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| webrtc | native | native | yes (depends on audio codec) | yes | Requires extra configuration, doesn't support h.265. Frigate attempts to use WebRTC when MSE fails or when using a camera's two-way talk feature. |
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### Camera Settings Recommendations
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If you are using go2rtc, you should adjust the following settings in your camera's firmware for the best experience with Live view:
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- Video codec: H.264 - provides the most compatible video codec with all live view technologies and browsers.
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- Audio codec: AAC - provides the most compatible video codec with all live view technologies and browsers.
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- I-frame interval (sometimes called the keyframe interval, the interframe space, or the GOP length): match your camera's frame rate, or choose "1x" (for interframe space on Reolink cameras). For example, if your stream outputs 20fps, your i-frame interval should be 20 (or 1x on Reolink). See [this page](https://gardinal.net/understanding-the-keyframe-interval/) for more on keyframes.
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The default video and audio codec on your camera may not always be compatible with your browser, which is why setting them to H.264 and AAC is recommended. See the [go2rtc docs](https://github.com/AlexxIT/go2rtc?tab=readme-ov-file#codecs-madness) for codec support information.
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### Audio Support
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### Audio Support
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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.
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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.
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