Files
meet/src/frontend/src/features/rooms/components/Conference.tsx
T
lebaudantoine 21c57bffb4 🧑‍💻(devx) add a WebRTC stats and network throttling devtool
Introduce an in-app devtool that monitors WebRTC statistics in
real time and lets developers simulate various network scenarios,
including constraining the uplink and downlink bandwidth.

Makes it much easier to reproduce and investigate connectivity or
quality issues locally without depending on external tools.

The code was AI generated, and might contain some smell.
It's only enabled in dev, and not included in the production
build. Feel free to enhance it as needed.
2026-09-01 22:05:53 +02:00

308 lines
10 KiB
TypeScript

import { useEffect, useMemo, useRef, useState } from 'react'
import { useQuery } from '@tanstack/react-query'
import { useTranslation } from 'react-i18next'
import {
LiveKitRoom,
usePersistentUserChoices,
} from '@livekit/components-react'
import {
ConnectionError,
ConnectionErrorReason,
DisconnectReason,
MediaDeviceFailure,
Room,
type RoomOptions,
VideoPresets,
} from 'livekit-client'
import { getMediaDeviceFailure } from '@/features/rooms/livekit/utils/mediaPermissions'
import { keys } from '@/api/queryKeys'
import { queryClient } from '@/api/queryClient'
import { Screen } from '@/layout/Screen'
import { QueryAware } from '@/components/QueryAware'
import { ErrorScreen } from '@/components/ErrorScreen'
import { fetchRoom } from '../api/fetchRoom'
import type { ApiRoom } from '../api/ApiRoom'
import { useCreateRoom } from '../api/createRoom'
import { InviteDialog } from './InviteDialog'
import { VideoConference } from '../livekit/prefabs/VideoConference'
import { css } from '@/styled-system/css'
import { BackgroundProcessorFactory } from '../livekit/components/blur'
import { LocalUserChoices } from '@/stores/userChoices'
import {
captureEvent,
captureMediaEvent,
reportError,
} from '@/features/analytics/telemetry'
import { useConfig } from '@/api/useConfig'
import { isFireFox } from '@/utils/livekit'
import { useIsMobile } from '@/utils/useIsMobile'
import { navigateTo } from '@/navigation/navigateTo'
import { PictureInPictureConference } from '@/features/pip/components/PictureInPictureConference'
import { notifyAutoMutedOnJoin } from '@/features/notifications/utils'
import { useSnapshot } from 'valtio'
import { userPreferencesStore } from '@/stores/userPreferences'
import { userStore } from '@/stores/user'
import { WatchMediaDeviceErrors } from './WatchMediaDeviceErrors'
import { MeetDevtools } from '@/features/devtools'
import { VOICE_AUDIO_CONSTRAINTS } from '@/features/rooms/livekit/utils/constants'
export const Conference = ({
roomId,
initialRoomData,
mode = 'join',
}: {
roomId: string
mode?: 'join' | 'create'
initialRoomData?: ApiRoom
}) => {
const { data: apiConfig } = useConfig()
const { userChoices: userConfig } = usePersistentUserChoices() as {
userChoices: LocalUserChoices
}
const { username } = useSnapshot(userStore)
useEffect(() => {
void captureMediaEvent('visit-room', { slug: roomId })
}, [roomId])
const fetchKey = [keys.room, roomId]
const [isConnectionWarmedUp, setIsConnectionWarmedUp] = useState(false)
const userPreferencesSnap = useSnapshot(userPreferencesStore)
const {
mutateAsync: createRoom,
status: createStatus,
isError: isCreateError,
} = useCreateRoom({
onSuccess: (data) => {
queryClient.setQueryData(fetchKey, data)
},
})
const {
status: fetchStatus,
isError: isFetchError,
data,
} = useQuery({
queryKey: fetchKey,
staleTime: 6 * 60 * 60 * 1000, // By default, LiveKit access tokens expire 6 hours after generation
initialData: initialRoomData,
queryFn: () =>
fetchRoom({
roomId: roomId as string,
username: username,
}).catch((error) => {
if (error.statusCode == '404') {
createRoom({ slug: roomId, username })
}
}),
retry: false,
})
const roomOptions = useMemo((): RoomOptions => {
return {
adaptiveStream: true,
dynacast: true,
publishDefaults: {
videoCodec: 'vp9',
},
videoCaptureDefaults: {
deviceId: userConfig.videoDeviceId ?? undefined,
resolution: userConfig.videoPublishResolution
? VideoPresets[userConfig.videoPublishResolution].resolution
: undefined,
},
audioCaptureDefaults: {
deviceId: userConfig.audioDeviceId ?? undefined,
...VOICE_AUDIO_CONSTRAINTS,
},
audioOutput: {
deviceId: userConfig.audioOutputDeviceId ?? undefined,
},
}
// do not rely on the userConfig object directly as its reference may change on every render
}, [
userConfig.videoDeviceId,
userConfig.videoPublishResolution,
userConfig.audioDeviceId,
userConfig.audioOutputDeviceId,
])
const room = useMemo(() => new Room(roomOptions), [roomOptions])
useEffect(() => {
/**
* Warm up connection to LiveKit server before joining room
* This prefetch helps reduce initial connection latency by establishing
* an early HTTP connection to the WebRTC signaling server
*
* It should cache DNS and TLS keys.
*/
const prepareConnection = async () => {
if (!apiConfig || isConnectionWarmedUp) return
await room.prepareConnection(apiConfig.livekit.url)
if (isFireFox() && apiConfig.livekit.enable_firefox_proxy_workaround) {
try {
const wssUrl =
apiConfig.livekit.url
.replace('https://', 'wss://')
.replace(/\/$/, '') + '/rtc'
/**
* FIREFOX + PROXY WORKAROUND:
*
* Issue: On Firefox behind proxy configurations, WebSocket signaling fails to establish.
* Symptom: Client receives HTTP 200 instead of expected 101 (Switching Protocols).
* Root Cause: Certificate/security issue where the initial request is considered unsecure.
*
* Solution: Pre-establish a WebSocket connection to the signaling server, which fails.
* This "primes" the connection, allowing subsequent WebSocket establishments to work correctly.
*
* Note: This issue is reproducible on LiveKit's demo app.
* Reference: livekit-examples/meet/issues/466
*/
const ws = new WebSocket(wssUrl)
// 401 unauthorized response is expected
ws.onerror = () => ws.readyState <= 1 && ws.close()
} catch (e) {
console.debug('Firefox WebSocket workaround failed.', e)
}
}
setIsConnectionWarmedUp(true)
}
prepareConnection()
}, [room, apiConfig, isConnectionWarmedUp])
const isMobile = useIsMobile()
const hasAutoMutedRef = useRef(false)
/*
* Ensure stable WebSocket connection URL. This is critical for legacy browser compatibility
* (Firefox <124, Chrome <125, Edge <125) where HTTPS URLs in WebSocket() constructor
* may fail - the force_wss_protocol flag allows explicit WSS protocol conversion
*/
const serverUrl = useMemo(() => {
const livekit_url = apiConfig?.livekit.url
if (!livekit_url) return
if (apiConfig?.livekit.force_wss_protocol) {
return livekit_url.replace('https://', 'wss://')
}
return livekit_url
}, [apiConfig?.livekit])
const { t } = useTranslation('rooms')
if (isCreateError) {
// this error screen should be replaced by a proper waiting room for anonymous user.
return (
<ErrorScreen
title={t('error.createRoom.heading')}
body={t('error.createRoom.body')}
/>
)
}
// Some clients (like DINUM) operate in bandwidth-constrained environments
// These settings help ensure successful connections in poor network conditions
const connectOptions = {
maxRetries: 5, // Default: 1. Only for unreachable server scenarios
peerConnectionTimeout: 60000, // Default: 15s. Extended for slow TURN/TLS negotiation
}
return (
<QueryAware status={isFetchError ? createStatus : fetchStatus}>
<Screen header={false} footer={false}>
<LiveKitRoom
room={room}
serverUrl={serverUrl}
token={data?.livekit?.token}
connect={isConnectionWarmedUp}
audio={userConfig.audioEnabled}
video={
userConfig.videoEnabled && {
processor: BackgroundProcessorFactory.fromProcessorConfig(
userConfig.processorConfig
),
}
}
connectOptions={connectOptions}
className={css({
backgroundColor: 'primaryDark.50 !important',
})}
onError={(e) => {
const failure = getMediaDeviceFailure(e)
if (failure && failure !== MediaDeviceFailure.Other) return
// connect() was aborted by a disconnect() before the join completed
if (
e instanceof ConnectionError &&
e.reason === ConnectionErrorReason.Cancelled
) {
void captureEvent('connection-cancelled')
return
}
reportError('livekit_room_error', e, {
path: 'connect_publish',
})
}}
onConnected={async () => {
if (!apiConfig) return
if (
userPreferencesSnap.is_auto_mute_large_room_enabled &&
!hasAutoMutedRef.current &&
userConfig.audioEnabled &&
room.numParticipants > apiConfig.auto_mute_on_join_threshold
) {
hasAutoMutedRef.current = true
await room.localParticipant.setMicrophoneEnabled(false)
notifyAutoMutedOnJoin()
}
}}
onDisconnected={(e) => {
const metadata = {
room_id: roomId,
}
switch (e) {
case DisconnectReason.CLIENT_INITIATED:
navigateTo(
'feedback',
{},
{
state: { ...metadata },
}
)
return
case DisconnectReason.DUPLICATE_IDENTITY:
case DisconnectReason.PARTICIPANT_REMOVED:
navigateTo(
'feedback',
{},
{
state: {
reason: e,
...metadata,
},
}
)
return
}
}}
>
<WatchMediaDeviceErrors />
<VideoConference />
{!isMobile && <InviteDialog mode={mode} />}
<PictureInPictureConference />
<MeetDevtools />
</LiveKitRoom>
</Screen>
</QueryAware>
)
}