fix: Hold the 2-way audio backchannel open for the shortest possible time (#2697)
The card claimed a camera's ONVIF audio backchannel in two places that had nothing to do with a call: the capability probe at camera init (`µphone`), and a pre-armed `sendonly` audio transceiver on every live WebRTC offer. Merely looking at a dashboard occupied the camera's speaker line. Outbound audio now travels on its own audio-only WebRTC connection, opened when a call is answered and closed when it ends. - The backchannel is claimed only for the duration of a call. Idle viewing claims nothing. - Two-way audio now works in `mse`, `mp4` and `mjpeg` modes (note: the outbound audio still traverses WebRTC). - No renegotiation and no video blink at call start or end. - A call that cannot carry audio now reports it and ends, instead of showing a live microphone that goes nowhere. - `live.microphone.always_connected` is now purely about the browser microphone permission prompt. - Call setup measured at 66ms (LAN) and ~260ms (cellular) for ICE and DTLS, plus ~300ms for `go2rtc` to open an RTSP backchannel. Verified against a live Frigate + `go2rtc` instance, and by unit tests at 100% coverage. - Closes #2691 - Closes #2039 - Closes #2178 Ref #2299 -- the probe no longer opens a backchannel, but it still runs per camera on every load and reconnect, and still dials the camera on the direct-`go2rtc` path. Caching remains to be done. Ref AlexxIT/go2rtc#1860 -- once a call has opened a backchannel, `go2rtc` keeps that media set up on the camera's RTSP session for the life of the producer. Diagnoses #2678
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@@ -42,7 +42,12 @@ cameras:
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> microphone button is intermittently missing on load, try increasing
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> `metadata_fetch_timeout_seconds` or use
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> [`capabilities.force`](./README.md?id=capabilities) to skip metadata
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> detection entirely.
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> detection entirely. If it is _never_ present, check the `go2rtc` stream
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> itself: Frigate [recommends](https://docs.frigate.video/configuration/live/)
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> adding `#backchannel=0` to the stream it restreams, which stops that stream
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> offering 2-way audio at all. Point `stream` at a talk-capable stream instead.
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> Note that _any_ `#` option on an RTSP source has this effect unless it also
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> includes `backchannel=1`.
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## `go2rtc (experimental)`
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@@ -1 +1 @@
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id="part1-tail"><text fill="#000000" font-family="sans-serif" font-size="14" lengthAdjust="spacing" textLength="31.9102" x="42" y="1112.8779">User</text><ellipse cx="60.9551" cy="1124.6797" fill="#E2E2F0" rx="8" ry="8" style="stroke:#181818;stroke-width:0.5;"/><path d="M60.9551,1132.6797 L60.9551,1159.6797 M47.9551,1140.6797 L73.9551,1140.6797 M60.9551,1159.6797 L47.9551,1174.6797 M60.9551,1159.6797 L73.9551,1174.6797" fill="none" style="stroke:#181818;stroke-width:0.5;"/></g><g class="participant participant-head" data-entity-uid="part2" data-qualified-name="Card" data-source-line="4" id="part2-head"><rect fill="#E2E2F0" height="30.2969" rx="2.5" ry="2.5" style="stroke:#181818;stroke-width:0.5;" width="40.1338" x="326.8052" y="50"/><text fill="#000000" font-family="sans-serif" font-size="14" lengthAdjust="spacing" textLength="26.1338" x="333.8052" y="69.9951">Call</text></g><g class="participant participant-tail" data-entity-uid="part2" data-qualified-name="Card" data-source-line="4" 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fill="#000000" font-family="sans-serif" font-size="14Line truncated
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<svg xmlns="http://www.w3.org/2000/svg" xmlns:xlink="http://www.w3.org/1999/xlink" contentStyleType="text/css" data-diagram-type="SEQUENCE" height="1283px" preserveAspectRatio="none" style="width:899px;height:1283px;background:#FFFFFF;" version="1.1" viewBox="0 0 899 1283" width="899px" zoomAndPan="magnify"><?plantuml 1.2026.3?><defs/><g><rect fill="none" height="110.5313" style="stroke:#000000;stroke-width:1.5;" width="838.5503" x="32" y="98.2969"/><rect fill="none" height="235" style="stroke:#000000;stroke-width:1.5;" width="734.6846" x="32" y="222.8281"/><rect fill="none" height="110.5313" style="stroke:#000000;stroke-width:1.5;" width="828.7109" x="32" y="471.8281"/><rect fill="none" height="75.3984" style="stroke:#000000;stroke-width:1.5;" width="734.6846" x="32" y="885.9531"/><g class="participant-lifeline" data-entity-uid="part1" data-qualified-name="User" data-source-line="3" id="part1-lifeline"><g><title>User</title><rect fill="#000000" fill-opacity="0.00000" 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id="part2-tail"><rect fill="#E2E2F0" height="30.2969" rx="2.5" ry="2.5" style="stroke:#181818;stroke-width:0.5;" width="40.1338" x="326.8052" y="1200.4141"/><text fill="#000000" font-family="sans-serif" font-size="14" lengthAdjust="spacing" textLength="26.1338" x="333.8052" y="1220.4092">Call</text></g><g class="participant participant-head" data-entity-uid="part3" data-qualified-name="Mic" data-source-line="5" id="part3-head"><rect fill="#E2E2F0" height="30.2969" rx="2.5" ry="2.5" style="stroke:#181818;stroke-width:0.5;" width="95.7988" x="603.7676" y="50"/><text fill="#000000" font-family="sans-serif" font-size="14" lengthAdjust="spacing" textLength="81.7988" x="610.7676" y="69.9951">Microphone</text></g><g class="participant participant-tail" data-entity-uid="part3" data-qualified-name="Mic" data-source-line="5" id="part3-tail"><rect fill="#E2E2F0" height="30.2969" rx="2.5" ry="2.5" style="stroke:#181818;stroke-width:0.5;" width="95.7988" x="603.7676" y="1200.4141"/><text fill="#000000" font-family="sans-serif" font-size="14Line truncated
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|
Before Width: | Height: | Size: 27 KiB After Width: | Height: | Size: 28 KiB |
@@ -40,6 +40,12 @@ note over User, Mic
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can ring even where microphone access needs a tap to be granted
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end note
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note over Card
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The card opens an audio connection to the camera and holds it only while the
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call is in progress. If the camera has no 2-way audio, or another application
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is already using it, the call fails with an error
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end note
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Card --> User : You hear the caller
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note over Card
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Inbound audio unmutes if configured (<color:#43a047>**""live.auto_unmute""**</color>)
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@@ -61,6 +67,7 @@ note over User, Card
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<color:#43a047>**""live.controls.call.lock""**</color> is disabled -- navigating away
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end note
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note over Card : The audio connection to the camera closes, releasing it for other applications
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Card --> User : Inbound audio mutes (<color:#43a047>**""live.auto_mute""**</color>)
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Card -> Mic : Microphone mutes and disconnects
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note over Mic
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@@ -17,13 +17,19 @@ challenging.
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- Only Frigate cameras are supported.
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- Only the `go2rtc` and `go2rtc-experimental` live providers are supported.
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- Only the `webrtc` mode supports 2-way audio.
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- The browser must be able to reach `go2rtc` over WebRTC. Outbound audio always
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travels on its own WebRTC connection, regardless of what mode is carrying the
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video.
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If your setup supports 2-way audio but detection is intermittent on load:
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- Increase `cameras[].go2rtc.metadata_fetch_timeout_seconds`.
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- Or force the capability with `cameras[].capabilities.force: ['2-way-audio']`.
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If detection never succeeds for a camera, the `go2rtc` stream itself may not
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offer 2-way audio -- see
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[`go2rtc` live provider configuration](../configuration/cameras/live-provider.md?id=go2rtc).
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## Example configuration
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```yaml
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@@ -65,10 +71,14 @@ enabled by default and appears in the `live` view whenever the selected camera
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in the overlay to speak. Both behaviors are configurable via
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[`live.microphone.auto_unmute`](../configuration/live.md?id=microphone) and
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[`live.auto_unmute`](../configuration/live.md).
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- The camera will always load _without_ the microphone connected, unless the
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- The camera loads _without_ the microphone connected, unless the
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[`always_connected`](../configuration/live.md?id=microphone) microphone option
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is set to `true`. On the first call there may be a brief `webrtc` connection
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reset to include 2-way audio.
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is set to `true`. Starting a call opens a separate connection that carries your
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voice to the camera; ending the call closes it. The camera's audio input is
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therefore occupied only while a call is in progress, leaving it free for other
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applications the rest of the time. Expect under half a second between starting
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a call and being audible on a local network, and a little more remotely. The
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video keeps playing throughout.
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- The browser asks for microphone permission when a call needs it. How often it
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asks depends on the browser: Chrome remembers the choice for the site, Safari
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asks once per page load, and Firefox asks for every call unless _Remember this
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