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
30 lines
857 B
TypeScript
30 lines
857 B
TypeScript
import { describe, expect, it, vi } from 'vitest';
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import {
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createBrowserPeerConnection,
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GO2RTC_PEER_CONNECTION_CONFIG,
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} from '../../src/go2rtc/peer-connection';
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describe('peer-connection', () => {
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it('should configure two STUN servers with max-bundle', () => {
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expect(GO2RTC_PEER_CONNECTION_CONFIG).toEqual({
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bundlePolicy: 'max-bundle',
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iceServers: [
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{
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urls: ['stun:stun.l.google.com:19302', 'stun:stun.cloudflare.com:3478'],
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},
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],
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});
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});
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it('should construct a real peer connection', () => {
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const RTCPeerConnectionMock = vi.fn();
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vi.stubGlobal('RTCPeerConnection', RTCPeerConnectionMock);
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createBrowserPeerConnection(GO2RTC_PEER_CONNECTION_CONFIG);
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expect(RTCPeerConnectionMock).toHaveBeenCalledWith(GO2RTC_PEER_CONNECTION_CONFIG);
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vi.unstubAllGlobals();
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});
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});
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