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
- Closes#2556
- Closes#2450
**Key intended features:**
- go2rtc compatible
- 100% test coverage to significantly improve ability to test, maintain
and work around browser weirdnesses (e.g. Safari).
- Written from the ground up in the style of the rest of the project.
**To use:**
- Change `live_provider` from `go2rtc` to `go2rtc-experimental`.