Screen share was capped by WebRTC's default ~2.5 Mbps and by window capture resolution, and fixed aspect-video cards forced the page to scroll. Adds minimal [sejong-rtc] logs to diagnose ICE failures. Co-authored-by: Cursor <cursoragent@cursor.com>
247 lines
7.6 KiB
TypeScript
247 lines
7.6 KiB
TypeScript
import type { AppSocket } from './useSocket'
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import { iceSummary, rtcLog, rtcWarn } from '~/utils/rtcLog'
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type PeerKey = string
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function peerKey(remoteId: string, streamId: string) {
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return `${remoteId}:${streamId}`
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}
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/** WebRTC defaults cap screen share near 2.5 Mbps, which looks soft at 1080p60. */
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const MAX_VIDEO_BITRATE = 12_000_000
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export function useWebRTC(options: {
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socket: Ref<AppSocket | null>
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clientId: Ref<string>
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getLocalStream: () => MediaStream | null
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}) {
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const peers = new Map<PeerKey, RTCPeerConnection>()
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const remoteStreams = ref<Record<string, MediaStream>>({})
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const iceServers: RTCIceServer[] = [{ urls: 'stun:stun.l.google.com:19302' }]
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async function applySenderQuality(pc: RTCPeerConnection, key: string) {
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for (const sender of pc.getSenders()) {
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if (sender.track?.kind !== 'video') continue
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const params = sender.getParameters()
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if (!params.encodings || params.encodings.length === 0) {
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params.encodings = [{}]
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}
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params.encodings[0]!.maxBitrate = MAX_VIDEO_BITRATE
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params.encodings[0]!.maxFramerate = 60
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params.degradationPreference = 'maintain-framerate'
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try {
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await sender.setParameters(params)
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rtcLog('sender:quality', { key, maxBitrate: MAX_VIDEO_BITRATE, maxFramerate: 60 })
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} catch (e) {
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rtcWarn('sender:setParameters failed', { key, err: String(e) })
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}
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}
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}
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function getPc(remoteId: string, streamId: string) {
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const key = peerKey(remoteId, streamId)
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let pc = peers.get(key)
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if (pc) return pc
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rtcLog('pc:create', { key, iceServers })
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pc = new RTCPeerConnection({ iceServers })
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peers.set(key, pc)
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pc.onicecandidate = (ev) => {
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if (!ev.candidate || !options.socket.value) {
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if (!ev.candidate) rtcLog('ice:local gathering done', { key })
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return
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}
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rtcLog('ice:local', { key, ...iceSummary(ev.candidate.toJSON()) })
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options.socket.value.emit('webrtc:ice', {
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to: remoteId,
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streamId,
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candidate: ev.candidate.toJSON(),
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})
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}
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pc.oniceconnectionstatechange = () => {
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rtcLog('ice:state', { key, state: pc?.iceConnectionState })
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if (pc?.iceConnectionState === 'failed') {
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rtcWarn('ice:FAILED — same-LAN check firewall/AP isolation; else need TURN', { key })
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}
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}
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pc.onconnectionstatechange = () => {
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rtcLog('pc:state', { key, state: pc?.connectionState })
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if (pc?.connectionState === 'failed' || pc?.connectionState === 'closed') {
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closePeer(remoteId, streamId)
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}
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}
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pc.ontrack = (ev) => {
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const stream = ev.streams[0] || new MediaStream([ev.track])
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rtcLog('track:remote', {
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key,
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kind: ev.track.kind,
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id: ev.track.id,
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muted: ev.track.muted,
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readyState: ev.track.readyState,
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streamTracks: stream.getTracks().map((t) => t.kind),
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})
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remoteStreams.value = { ...remoteStreams.value, [streamId]: stream }
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}
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return pc
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}
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function closePeer(remoteId: string, streamId: string) {
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const key = peerKey(remoteId, streamId)
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const pc = peers.get(key)
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if (pc) {
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rtcLog('pc:close', { key })
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pc.close()
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peers.delete(key)
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}
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if (remoteStreams.value[streamId]) {
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const next = { ...remoteStreams.value }
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delete next[streamId]
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remoteStreams.value = next
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}
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}
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function closeAll() {
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rtcLog('pc:closeAll', { count: peers.size })
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for (const pc of peers.values()) pc.close()
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peers.clear()
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remoteStreams.value = {}
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}
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async function createOfferAsViewer(broadcasterId: string, streamId: string) {
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const pc = getPc(broadcasterId, streamId)
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if (pc.getTransceivers().length === 0) {
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pc.addTransceiver('video', { direction: 'recvonly' })
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pc.addTransceiver('audio', { direction: 'recvonly' })
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}
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const offer = await pc.createOffer()
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await pc.setLocalDescription(offer)
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rtcLog('offer:send', {
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to: broadcasterId,
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streamId,
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mLines: (offer.sdp?.match(/^m=/gm) || []).length,
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})
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options.socket.value?.emit('webrtc:offer', {
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to: broadcasterId,
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streamId,
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sdp: offer,
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})
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}
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async function handleOffer(from: string, streamId: string, sdp: RTCSessionDescriptionInit) {
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const pc = getPc(from, streamId)
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rtcLog('offer:recv', {
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from,
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streamId,
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mLines: (sdp.sdp?.match(/^m=/gm) || []).length,
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hasLocalStream: Boolean(options.getLocalStream()),
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localTracks: options.getLocalStream()?.getTracks().map((t) => t.kind) || [],
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})
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await pc.setRemoteDescription(sdp)
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const local = options.getLocalStream()
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if (local) {
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for (const track of local.getTracks()) {
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const exists = pc.getSenders().some((s) => s.track?.id === track.id)
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if (!exists) {
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pc.addTrack(track, local)
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rtcLog('track:add local', { to: from, streamId, kind: track.kind })
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}
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}
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} else {
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rtcWarn('offer:recv but no local MediaStream to send', { from, streamId })
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}
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const answer = await pc.createAnswer()
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await pc.setLocalDescription(answer)
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await applySenderQuality(pc, peerKey(from, streamId))
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rtcLog('answer:send', {
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to: from,
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streamId,
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mLines: (answer.sdp?.match(/^m=/gm) || []).length,
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})
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options.socket.value?.emit('webrtc:answer', {
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to: from,
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streamId,
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sdp: answer,
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})
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}
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async function handleAnswer(from: string, streamId: string, sdp: RTCSessionDescriptionInit) {
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const pc = peers.get(peerKey(from, streamId))
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if (!pc) {
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rtcWarn('answer:recv but no PC', { from, streamId })
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return
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}
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rtcLog('answer:recv', {
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from,
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streamId,
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mLines: (sdp.sdp?.match(/^m=/gm) || []).length,
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})
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await pc.setRemoteDescription(sdp)
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}
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async function handleIce(from: string, streamId: string, candidate: RTCIceCandidateInit) {
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const pc = peers.get(peerKey(from, streamId))
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if (!pc) return
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rtcLog('ice:remote', { from, streamId, ...iceSummary(candidate) })
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try {
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await pc.addIceCandidate(candidate)
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} catch (e) {
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rtcWarn('ice:add failed', { from, streamId, err: String(e) })
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}
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}
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function attachLocalTracksToPeer(remoteId: string, streamId: string) {
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const local = options.getLocalStream()
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if (!local) return
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const pc = getPc(remoteId, streamId)
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for (const track of local.getTracks()) {
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const exists = pc.getSenders().some((s) => s.track?.id === track.id)
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if (!exists) pc.addTrack(track, local)
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}
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}
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function bindSocketHandlers() {
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const s = options.socket.value
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if (!s) return
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rtcLog('socket:bind webrtc handlers', { clientId: options.clientId.value })
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s.on('webrtc:offer', async (payload) => {
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if (!payload.from || !payload.sdp || !payload.streamId) return
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await handleOffer(payload.from, payload.streamId, payload.sdp)
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})
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s.on('webrtc:answer', async (payload) => {
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if (!payload.from || !payload.sdp || !payload.streamId) return
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await handleAnswer(payload.from, payload.streamId, payload.sdp)
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})
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s.on('webrtc:ice', async (payload) => {
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if (!payload.from || !payload.candidate || !payload.streamId) return
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await handleIce(payload.from, payload.streamId, payload.candidate)
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})
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s.on('peer:left', (payload) => {
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rtcLog('peer:left', payload)
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for (const key of [...peers.keys()]) {
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if (key.startsWith(`${payload.clientId}:`)) {
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const streamId = key.split(':')[1]
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closePeer(payload.clientId, streamId)
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}
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}
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})
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}
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return {
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remoteStreams,
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createOfferAsViewer,
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attachLocalTracksToPeer,
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closePeer,
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closeAll,
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bindSocketHandlers,
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}
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}
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