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
461 lines
14 KiB
TypeScript
461 lines
14 KiB
TypeScript
import { afterEach, beforeEach, describe, expect, it, vi } from 'vitest';
|
|
|
|
import { WebRTCStreamSource } from '../../../../../../src/components-lib/live/providers/go2rtc-experimental/sources/webrtc';
|
|
import type {
|
|
StreamSourceContext,
|
|
VideoStreamTarget,
|
|
} from '../../../../../../src/components-lib/live/providers/go2rtc-experimental/types';
|
|
import {
|
|
FakeMediaStream,
|
|
FakeRTCPeerConnection,
|
|
type FakeMediaStreamTrack,
|
|
} from '../../../../../go2rtc/test-utils';
|
|
import { flushPromises } from '../../../../../test-utils';
|
|
import { FakeStreamSourceChannel } from '../test-utils';
|
|
|
|
// @vitest-environment jsdom
|
|
describe('WebRTCStreamSource', () => {
|
|
const setup = () => {
|
|
const video = document.createElement('video');
|
|
const channel = new FakeStreamSourceChannel();
|
|
const loadedCallback = vi.fn();
|
|
const failedCallback = vi.fn();
|
|
|
|
const context: StreamSourceContext<VideoStreamTarget> = {
|
|
target: { kind: 'video', video },
|
|
channel,
|
|
callbacks: { loadedCallback, failedCallback },
|
|
};
|
|
|
|
const pc = new FakeRTCPeerConnection();
|
|
const createPeerConnection = vi.fn(() => pc.asPeerConnection());
|
|
const source = new WebRTCStreamSource(context, {
|
|
createPeerConnection,
|
|
createMediaStream: (tracks) =>
|
|
new FakeMediaStream(tracks as unknown as FakeMediaStreamTrack[]).asMediaStream(),
|
|
});
|
|
|
|
return {
|
|
channel,
|
|
context,
|
|
createPeerConnection,
|
|
failedCallback,
|
|
loadedCallback,
|
|
pc,
|
|
source,
|
|
video,
|
|
};
|
|
};
|
|
|
|
beforeEach(() => {
|
|
vi.useFakeTimers();
|
|
});
|
|
|
|
afterEach(() => {
|
|
vi.useRealTimers();
|
|
});
|
|
|
|
describe('transceivers', () => {
|
|
it('should offer inbound video and audio only', () => {
|
|
const { source, pc } = setup();
|
|
source.start();
|
|
|
|
expect(pc.transceivers.map((transceiver) => transceiver.direction)).toEqual([
|
|
'recvonly',
|
|
'recvonly',
|
|
]);
|
|
});
|
|
});
|
|
|
|
describe('negotiation', () => {
|
|
it('should send an offer after setting the local description', async () => {
|
|
const { source, channel, pc } = setup();
|
|
source.start();
|
|
await flushPromises();
|
|
|
|
expect(pc.setLocalDescription).toHaveBeenCalled();
|
|
expect(channel.sent).toContainEqual({
|
|
type: 'webrtc/offer',
|
|
value: 'v=0\r\noffer',
|
|
});
|
|
});
|
|
|
|
it('should send an empty offer sdp when the browser omits it', async () => {
|
|
const { source, channel, pc } = setup();
|
|
pc.createOffer.mockResolvedValue({ type: 'offer', sdp: undefined });
|
|
source.start();
|
|
await flushPromises();
|
|
|
|
expect(channel.sent).toContainEqual({ type: 'webrtc/offer', value: '' });
|
|
});
|
|
|
|
it('should not send an offer if the connection was replaced mid-negotiation', async () => {
|
|
const { source, channel, pc } = setup();
|
|
let resolveOffer: (value: { type: string; sdp: string }) => void = () => {};
|
|
pc.createOffer.mockReturnValue(
|
|
new Promise((resolve) => {
|
|
resolveOffer = resolve;
|
|
}),
|
|
);
|
|
source.start();
|
|
source.stop();
|
|
resolveOffer({ type: 'offer', sdp: 'v=0\r\noffer' });
|
|
await flushPromises();
|
|
|
|
expect(channel.sent).not.toContainEqual(
|
|
expect.objectContaining({ type: 'webrtc/offer' }),
|
|
);
|
|
});
|
|
|
|
it('should not send an offer if replaced between offer and local description', async () => {
|
|
const { source, channel, pc } = setup();
|
|
let resolveLocal: () => void = () => {};
|
|
pc.setLocalDescription.mockReturnValue(
|
|
new Promise((resolve) => {
|
|
resolveLocal = resolve;
|
|
}),
|
|
);
|
|
source.start();
|
|
await flushPromises();
|
|
source.stop();
|
|
resolveLocal();
|
|
await flushPromises();
|
|
|
|
expect(channel.sent).not.toContainEqual(
|
|
expect.objectContaining({ type: 'webrtc/offer' }),
|
|
);
|
|
});
|
|
|
|
it('should fail on a negotiation error for the current connection', async () => {
|
|
const { source, pc, failedCallback } = setup();
|
|
pc.createOffer.mockRejectedValue(new Error('offer failed'));
|
|
source.start();
|
|
await flushPromises();
|
|
|
|
expect(failedCallback).toHaveBeenCalledWith('media_error');
|
|
});
|
|
|
|
it('should ignore a negotiation error after the connection was replaced', async () => {
|
|
const { source, pc, failedCallback } = setup();
|
|
let rejectOffer: (reason: Error) => void = () => {};
|
|
pc.createOffer.mockReturnValue(
|
|
new Promise((_resolve, reject) => {
|
|
rejectOffer = reject;
|
|
}),
|
|
);
|
|
source.start();
|
|
source.stop();
|
|
rejectOffer(new Error('offer failed'));
|
|
await flushPromises();
|
|
|
|
expect(failedCallback).not.toHaveBeenCalled();
|
|
});
|
|
});
|
|
|
|
describe('signaling', () => {
|
|
it('should send ICE candidates to the server', () => {
|
|
const { source, channel, pc } = setup();
|
|
source.start();
|
|
pc.fireIceCandidate('candidate:1 1 udp 2 1.2.3.4 5 typ host');
|
|
|
|
expect(channel.sent).toContainEqual({
|
|
type: 'webrtc/candidate',
|
|
value: 'candidate:1 1 udp 2 1.2.3.4 5 typ host',
|
|
});
|
|
});
|
|
|
|
it('should send an empty candidate at end-of-candidates', () => {
|
|
const { source, channel, pc } = setup();
|
|
source.start();
|
|
pc.fireIceCandidate(null);
|
|
|
|
expect(channel.sent).toContainEqual({ type: 'webrtc/candidate', value: '' });
|
|
});
|
|
|
|
it('should not send a candidate after stop', () => {
|
|
const { source, channel, pc } = setup();
|
|
source.start();
|
|
source.stop();
|
|
|
|
const sentBefore = channel.sent.length;
|
|
pc.fireIceCandidate('candidate:1 1 udp 2 1.2.3.4 5 typ host');
|
|
|
|
expect(channel.sent).toHaveLength(sentBefore);
|
|
});
|
|
|
|
it('should apply the server answer', () => {
|
|
const { source, channel, pc } = setup();
|
|
source.start();
|
|
channel.receiveMessage({ type: 'webrtc/answer', value: 'v=0\r\nanswer' });
|
|
|
|
expect(pc.setRemoteDescription).toHaveBeenCalledWith({
|
|
type: 'answer',
|
|
sdp: 'v=0\r\nanswer',
|
|
});
|
|
});
|
|
|
|
it('should swallow a rejected setRemoteDescription', async () => {
|
|
const { source, channel, pc } = setup();
|
|
pc.setRemoteDescription.mockRejectedValue(new Error('bad answer'));
|
|
source.start();
|
|
|
|
expect(() =>
|
|
channel.receiveMessage({ type: 'webrtc/answer', value: 'v=0\r\nanswer' }),
|
|
).not.toThrow();
|
|
await flushPromises();
|
|
});
|
|
|
|
it('should swallow a rejected addIceCandidate', async () => {
|
|
const { source, channel, pc } = setup();
|
|
pc.addIceCandidate.mockRejectedValue(new Error('bad candidate'));
|
|
source.start();
|
|
|
|
expect(() =>
|
|
channel.receiveMessage({
|
|
type: 'webrtc/candidate',
|
|
value: 'candidate:1 1 udp 2 1.2.3.4 5 typ host',
|
|
}),
|
|
).not.toThrow();
|
|
await flushPromises();
|
|
});
|
|
|
|
it('should add server ICE candidates with a fixed sdpMid', () => {
|
|
const { source, channel, pc } = setup();
|
|
source.start();
|
|
channel.receiveMessage({
|
|
type: 'webrtc/candidate',
|
|
value: 'candidate:2 1 udp 1 5.6.7.8 9 typ host',
|
|
});
|
|
|
|
expect(pc.addIceCandidate).toHaveBeenCalledWith({
|
|
candidate: 'candidate:2 1 udp 1 5.6.7.8 9 typ host',
|
|
sdpMid: '0',
|
|
});
|
|
});
|
|
|
|
it('should ignore an empty server ICE candidate', () => {
|
|
const { source, channel, pc } = setup();
|
|
source.start();
|
|
channel.receiveMessage({ type: 'webrtc/candidate', value: '' });
|
|
|
|
expect(pc.addIceCandidate).not.toHaveBeenCalled();
|
|
});
|
|
|
|
it('should ignore messages with a non-string value', () => {
|
|
const { source, channel, pc } = setup();
|
|
source.start();
|
|
channel.receiveMessage({ type: 'webrtc/answer', value: 42 });
|
|
|
|
expect(pc.setRemoteDescription).not.toHaveBeenCalled();
|
|
});
|
|
|
|
it('should ignore messages after the connection was replaced', () => {
|
|
const { source, channel, pc } = setup();
|
|
source.start();
|
|
source.stop();
|
|
channel.receiveMessage({ type: 'webrtc/answer', value: 'v=0\r\nanswer' });
|
|
|
|
expect(pc.setRemoteDescription).not.toHaveBeenCalled();
|
|
});
|
|
});
|
|
|
|
describe('server errors', () => {
|
|
it('should fail on a webrtc server error', () => {
|
|
const { source, channel, failedCallback } = setup();
|
|
source.start();
|
|
channel.receiveMessage({ type: 'error', value: 'webrtc/offer: stream not found' });
|
|
|
|
expect(failedCallback).toHaveBeenCalledWith('server_error');
|
|
});
|
|
|
|
it('should ignore server errors for other modes', () => {
|
|
const { source, channel, failedCallback } = setup();
|
|
source.start();
|
|
channel.receiveMessage({ type: 'error', value: 'mse: stream not found' });
|
|
|
|
expect(failedCallback).not.toHaveBeenCalled();
|
|
});
|
|
});
|
|
|
|
describe('connection', () => {
|
|
it('should attach the received stream and report loaded', () => {
|
|
const { source, pc, video, loadedCallback } = setup();
|
|
source.start();
|
|
pc.fireConnectionStateChange('connected');
|
|
|
|
expect(video.srcObject).not.toBeNull();
|
|
|
|
video.dispatchEvent(new Event('loadeddata'));
|
|
expect(loadedCallback).toHaveBeenCalledTimes(1);
|
|
});
|
|
|
|
it('should only build the received stream once', () => {
|
|
const { source, pc, video } = setup();
|
|
source.start();
|
|
pc.fireConnectionStateChange('connected');
|
|
const firstStream = video.srcObject;
|
|
pc.fireConnectionStateChange('connected');
|
|
|
|
expect(video.srcObject).toBe(firstStream);
|
|
});
|
|
|
|
it('should fail when the connection fails', () => {
|
|
const { source, pc, failedCallback } = setup();
|
|
source.start();
|
|
pc.fireConnectionStateChange('failed');
|
|
|
|
expect(failedCallback).toHaveBeenCalledWith('media_error');
|
|
});
|
|
|
|
it('should not fail on a recoverable disconnect', () => {
|
|
const { source, pc, failedCallback } = setup();
|
|
source.start();
|
|
pc.fireConnectionStateChange('disconnected');
|
|
|
|
expect(failedCallback).not.toHaveBeenCalled();
|
|
});
|
|
|
|
it('should fail when a disconnect escalates to failed', () => {
|
|
const { source, pc, failedCallback } = setup();
|
|
source.start();
|
|
pc.fireConnectionStateChange('disconnected');
|
|
pc.fireConnectionStateChange('failed');
|
|
|
|
expect(failedCallback).toHaveBeenCalledWith('media_error');
|
|
});
|
|
|
|
it('should ignore intermediate connection states', () => {
|
|
const { source, pc, video, failedCallback } = setup();
|
|
source.start();
|
|
pc.fireConnectionStateChange('connecting');
|
|
|
|
expect(video.srcObject).toBeFalsy();
|
|
expect(failedCallback).not.toHaveBeenCalled();
|
|
});
|
|
|
|
it('should ignore connection state changes after stop', () => {
|
|
const { source, pc, failedCallback } = setup();
|
|
source.start();
|
|
source.stop();
|
|
pc.fireConnectionStateChange('failed');
|
|
|
|
expect(failedCallback).not.toHaveBeenCalled();
|
|
});
|
|
|
|
it('should build the media stream with the global MediaStream by default', () => {
|
|
vi.stubGlobal('MediaStream', FakeMediaStream);
|
|
const video = document.createElement('video');
|
|
const channel = new FakeStreamSourceChannel();
|
|
const pc = new FakeRTCPeerConnection();
|
|
const source = new WebRTCStreamSource(
|
|
{
|
|
target: { kind: 'video', video },
|
|
channel,
|
|
callbacks: { loadedCallback: vi.fn(), failedCallback: vi.fn() },
|
|
},
|
|
{ createPeerConnection: () => pc.asPeerConnection() },
|
|
);
|
|
source.start();
|
|
pc.fireConnectionStateChange('connected');
|
|
|
|
expect(source.getMediaStream()).toBeInstanceOf(FakeMediaStream);
|
|
vi.unstubAllGlobals();
|
|
});
|
|
});
|
|
|
|
describe('connect timeout', () => {
|
|
it('should fail if no frame decodes within the connect timeout', () => {
|
|
const { source, pc, failedCallback } = setup();
|
|
source.start();
|
|
pc.fireConnectionStateChange('connected');
|
|
// Connected, but the real video never fires loadeddata.
|
|
vi.advanceTimersByTime(5 * 1000);
|
|
|
|
expect(failedCallback).toHaveBeenCalledWith('connect_timeout');
|
|
});
|
|
|
|
it('should cancel the connect timeout once loaded', () => {
|
|
const { source, pc, video, failedCallback } = setup();
|
|
source.start();
|
|
pc.fireConnectionStateChange('connected');
|
|
video.dispatchEvent(new Event('loadeddata'));
|
|
vi.advanceTimersByTime(5 * 1000);
|
|
|
|
expect(failedCallback).not.toHaveBeenCalledWith('connect_timeout');
|
|
});
|
|
});
|
|
|
|
describe('lifecycle', () => {
|
|
it('should close the connection on stop and clear the video', () => {
|
|
const { source, pc, video } = setup();
|
|
source.start();
|
|
pc.fireConnectionStateChange('connected');
|
|
source.stop();
|
|
|
|
expect(pc.close).toHaveBeenCalled();
|
|
expect(video.srcObject).toBeNull();
|
|
expect(source.getMediaStream()).toBeNull();
|
|
});
|
|
|
|
it('should tolerate stopping before starting', () => {
|
|
const { source } = setup();
|
|
|
|
expect(() => source.stop()).not.toThrow();
|
|
});
|
|
});
|
|
|
|
describe('reporting', () => {
|
|
it('should expose the media stream and peer connection', () => {
|
|
const { source, pc } = setup();
|
|
source.start();
|
|
pc.fireConnectionStateChange('connected');
|
|
|
|
expect(source.getMediaStream()).not.toBeNull();
|
|
expect(source.getPeerConnection()).toBe(pc.asPeerConnection());
|
|
});
|
|
|
|
it('should report its media capabilities', () => {
|
|
const { source, pc } = setup();
|
|
source.start();
|
|
pc.fireConnectionStateChange('connected');
|
|
|
|
expect(source.getCapabilities()).toEqual({
|
|
supportsPause: true,
|
|
hasAudio: expect.any(Boolean),
|
|
});
|
|
});
|
|
|
|
it('should report webrtc technology', () => {
|
|
const { source } = setup();
|
|
|
|
expect(source.getTechnology()).toEqual(['webrtc']);
|
|
});
|
|
|
|
it('should report a stream profile from the tracks and SDP', () => {
|
|
const { source, pc } = setup();
|
|
source.start();
|
|
pc.setRemoteDescription({ sdp: 'a=rtpmap:98 H265/90000\r\n' });
|
|
pc.fireConnectionStateChange('connected');
|
|
|
|
expect(source.getStreamProfile()).toEqual({
|
|
hasVideo: true,
|
|
hasH265Video: true,
|
|
hasAudio: true,
|
|
hasAACAudio: false,
|
|
});
|
|
});
|
|
|
|
it('should report an empty profile before connection', () => {
|
|
const { source } = setup();
|
|
source.start();
|
|
|
|
expect(source.getStreamProfile()).toEqual({
|
|
hasVideo: false,
|
|
hasH265Video: false,
|
|
hasAudio: false,
|
|
hasAACAudio: false,
|
|
});
|
|
});
|
|
});
|
|
});
|