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
(`&microphone`),
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

src/go2rtc/audio.ts

@@ -0,0 +1,76 @@
+import type { EnabledProxyConfig } from '../config/schema/common/proxy';
+import { homeAssistantSignAndFetch } from '../ha/fetch';
+import type { HomeAssistant } from '../ha/types';
+import { createProxiedEndpointIfNecessary } from '../ha/web-proxy';
+import type { Endpoint } from '../types';
+import { errorToConsole } from '../utils/basic';
+import { go2RTCStreamInfoSchema, type Go2RTCStreamInfo } from './types';
+
+const getGo2RTCStreamMetadata = async (
+  hass: HomeAssistant,
+  endpoint: Endpoint,
+  timeoutSeconds: number,
+): Promise<Go2RTCStreamInfo | null> => {
+  try {
+    return await homeAssistantSignAndFetch(hass, endpoint, go2RTCStreamInfoSchema, {
+      timeoutSeconds,
+    });
+  } catch (e) {
+    errorToConsole(e);
+    return null;
+  }
+};
+
+const streamSupports2WayAudio = (streamInfo: Go2RTCStreamInfo | null): boolean => {
+  if (!streamInfo?.producers) {
+    return false;
+  }
+  return streamInfo.producers.some(
+    (producer) =>
+      producer.medias?.some(
+        (media) =>
+          media.includes('audio') &&
+          (media.includes('sendonly') || media.includes('sendrecv')),
+      ) ?? false,
+  );
+};
+
+/**
+ * Fetch go2rtc metadata and determine if the stream supports 2-way audio.
+ * Handles proxy transformation if proxy config requires it.
+ * Returns false if the endpoint is not available or fetch fails.
+ *
+ * Note: Caller is responsible for checking if live_provider is 'go2rtc' before calling.
+ *
+ * @param hass Home Assistant instance.
+ * @param go2rtcMetadataEndpoint The go2rtc metadata endpoint.
+ * @param proxyConfig The resolved proxy configuration for live streams.
+ * @returns True if supports 2-way audio, false otherwise.
+ */
+export const supports2WayAudio = async (
+  hass: HomeAssistant,
+  metadataFetchTimeoutSeconds: number,
+  go2rtcMetadataEndpoint?: Endpoint | null,
+  proxyConfig?: EnabledProxyConfig,
+): Promise<boolean> => {
+  if (!go2rtcMetadataEndpoint) {
+    return false;
+  }
+
+  const endpoint = await createProxiedEndpointIfNecessary(
+    hass,
+    go2rtcMetadataEndpoint,
+    proxyConfig,
+    { openLimit: 1 },
+  );
+  if (!endpoint) {
+    return false;
+  }
+
+  const streamInfo = await getGo2RTCStreamMetadata(
+    hass,
+    endpoint,
+    metadataFetchTimeoutSeconds,
+  );
+  return streamSupports2WayAudio(streamInfo);
+};