diff --git a/src/utils/signaling.js b/src/utils/signaling.js
index cbd039a4373..b9d5a369ac9 100644
--- a/src/utils/signaling.js
+++ b/src/utils/signaling.js
@@ -25,6 +25,7 @@ import CancelableRequest from './CancelableRequest.ts'
 import Encryption from './e2ee/encryption.js'
 import { convertToUnix } from './formattedTime.ts'
 import { messagePleaseTryToReload } from './talkDesktopUtils.ts'
+import { DEFAULT_MAX_STREAM_BITS } from './webrtc/simplewebrtc/simulcastBitrates.ts'
 
 const actorStore = useActorStore(pinia)
 
@@ -71,6 +72,7 @@ function Base(settings) {
 	this.signalingConnectionTimeout = null
 	this.signalingConnectionWarning = null
 	this.signalingConnectionError = null
+	this.maxStreamBits = DEFAULT_MAX_STREAM_BITS
 }
 
 Signaling.Base = Base
@@ -1297,6 +1299,16 @@ Signaling.Standalone.prototype.joinResponseReceived = function(data, token) {
 	}
 
 	this._rejoinRoomAfterInvalidSession = null
+
+	// Apply the room-specific max bitrate provided in server response
+	// Falls back to the default value when a room has no configured limit
+	const totalBps = data.room?.bandwidth?.maxstreambitrate
+	if (typeof totalBps === 'number' && totalBps > 0) {
+		this.maxStreamBits = totalBps
+	} else {
+		this.maxStreamBits = DEFAULT_MAX_STREAM_BITS
+	}
+
 	this.signalingRoomJoined = token
 	if (this.pendingJoinCall && token === this.pendingJoinCall.token) {
 		const pendingJoinCallResolve = this.pendingJoinCall.resolve
diff --git a/src/utils/webrtc/simplewebrtc/peer.js b/src/utils/webrtc/simplewebrtc/peer.js
index 2f592176c44..7fa0b2f2446 100644
--- a/src/utils/webrtc/simplewebrtc/peer.js
+++ b/src/utils/webrtc/simplewebrtc/peer.js
@@ -8,6 +8,7 @@ import adapter from 'webrtc-adapter'
 import webrtcSupport from 'webrtcsupport'
 import WildEmitter from 'wildemitter'
 import { isSafari } from '../../browserCheck.ts'
+import { getSimulcastMaxBitrates } from './simulcastBitrates.ts'
 
 /**
  * @param {object} stream the stream object.
@@ -356,6 +357,17 @@ function mungeSdpForSimulcasting(sdp) {
 }
 /* eslint-enable */
 
+Peer.prototype._getMaxBitrates = function() {
+	// Read the live per-stream limit from the signaling connection so a room
+	// switch or reconnect is reflected, instead of the value frozen at
+	// SimpleWebRTC construction time.
+	const maxStreamBits = this.parent?.config?.connection?.maxStreamBits
+	if (maxStreamBits) {
+		return getSimulcastMaxBitrates(maxStreamBits)
+	}
+	return this.maxBitrates
+}
+
 Peer.prototype.offer = function(options) {
 	const sendVideo = this.sendVideoIfAvailable && this.type !== 'screen'
 	if (sendVideo && this.enableSimulcast && adapter.browserDetails.browser === 'firefox') {
@@ -368,22 +380,23 @@ Peer.prototype.offer = function(options) {
 			if (!parameters) {
 				parameters = {}
 			}
+			const maxBitrates = this._getMaxBitrates()
 			parameters.encodings = [
 				{
 					rid: 'h',
 					active: true,
-					maxBitrate: this.maxBitrates.high,
+					maxBitrate: maxBitrates.high,
 				},
 				{
 					rid: 'm',
 					active: true,
-					maxBitrate: this.maxBitrates.medium,
+					maxBitrate: maxBitrates.medium,
 					scaleResolutionDownBy: 2,
 				},
 				{
 					rid: 'l',
 					active: true,
-					maxBitrate: this.maxBitrates.low,
+					maxBitrate: maxBitrates.low,
 					scaleResolutionDownBy: 4,
 				},
 			]
diff --git a/src/utils/webrtc/simplewebrtc/simplewebrtc.js b/src/utils/webrtc/simplewebrtc/simplewebrtc.js
index 9af8778cdf6..11710329604 100644
--- a/src/utils/webrtc/simplewebrtc/simplewebrtc.js
+++ b/src/utils/webrtc/simplewebrtc/simplewebrtc.js
@@ -6,6 +6,7 @@
 import mockconsole from 'mockconsole'
 import webrtcSupport from 'webrtcsupport'
 import WildEmitter from 'wildemitter'
+import { getSimulcastMaxBitrates } from './simulcastBitrates.ts'
 import WebRTC from './webrtc.js'
 
 /**
@@ -19,11 +20,7 @@ export default function SimpleWebRTC(opts) {
 		debug: false,
 		enableDataChannels: true,
 		enableSimulcast: false,
-		maxBitrates: {
-			high: 900000,
-			medium: 300000,
-			low: 100000,
-		},
+		maxBitrates: getSimulcastMaxBitrates(opts.connection?.maxStreamBits),
 		autoRequestMedia: false,
 		receiveMedia: {
 			offerToReceiveAudio: 1,
diff --git a/src/utils/webrtc/simplewebrtc/simulcastBitrates.ts b/src/utils/webrtc/simplewebrtc/simulcastBitrates.ts
new file mode 100644
index 00000000000..6913d976d7e
--- /dev/null
+++ b/src/utils/webrtc/simplewebrtc/simulcastBitrates.ts
@@ -0,0 +1,44 @@
+/**
+ * SPDX-FileCopyrightText: 2025 Nextcloud GmbH and Nextcloud contributors
+ * SPDX-License-Identifier: AGPL-3.0-or-later
+ */
+
+/**
+ * Default maximum bitrate for a single published stream
+ * when the signaling server does not advertise a limit. (1 Mbps)
+ */
+export const DEFAULT_MAX_STREAM_BITS = 1_048_576
+/**
+ * Expected maximum bitrate for a hi-quality stream (1920x1080, 30fps)
+ * Over-provisioning for a minimum "high" resolution (1280x720, 30fps)
+ */
+export const HIGH_MAX_STREAM_BITS = 2_764_800
+/**
+ * Expected maximum bitrate for a mid-quality stream (640x360, 30fps)
+ */
+export const MEDIUM_MAX_STREAM_BITS = 300_000
+/**
+ * Expected maximum bitrate for a low-quality stream (320x180, 30fps)
+ */
+export const LOW_MAX_STREAM_BITS = 100_000
+
+/**
+ * Compute per-layer simulcast maxBitrate ceilings from the publisher's total
+ * bandwidth limit. Low and medium layers use fixed, resolution-appropriate
+ * caps; only the high layer scales with the available budget.
+ *
+ * Firefox only (applied via RTCRtpSender.setParameters());
+ * Chromium/Safari use SDP munging and let REMB distribute the bitrate.
+ *
+ * @param totalBps - Total bandwidth in bps.
+ */
+export function getSimulcastMaxBitrates(totalBps: number = DEFAULT_MAX_STREAM_BITS) {
+	if (typeof totalBps !== 'number' || totalBps <= 0) {
+		totalBps = DEFAULT_MAX_STREAM_BITS
+	}
+	return {
+		high: Math.min(Math.round(0.9 * totalBps), HIGH_MAX_STREAM_BITS),
+		medium: MEDIUM_MAX_STREAM_BITS,
+		low: LOW_MAX_STREAM_BITS,
+	}
+}
diff --git a/src/utils/webrtc/simplewebrtc/webrtc.js b/src/utils/webrtc/simplewebrtc/webrtc.js
index 3b443fc33d1..bb6a8fd3938 100644
--- a/src/utils/webrtc/simplewebrtc/webrtc.js
+++ b/src/utils/webrtc/simplewebrtc/webrtc.js
@@ -27,6 +27,7 @@ export default function WebRTC(opts) {
 		},
 		enableDataChannels: true,
 		enableSimulcast: false,
+		// Overwritten by simplewebrtc.js
 		maxBitrates: {
 			high: 900000,
 			medium: 300000,
diff --git a/src/utils/webrtc/webrtc.js b/src/utils/webrtc/webrtc.js
index 91869c50964..364e98b6a64 100644
--- a/src/utils/webrtc/webrtc.js
+++ b/src/utils/webrtc/webrtc.js
@@ -628,6 +628,84 @@ export function initWebRtc(signaling, _callParticipantCollection, _localCallPart
 	}
 	window.OCA.Talk.SimpleWebRTC = webrtc
 
+	// debug helper to inspect publisher simulcast layers
+	// (configured ceilings + live per-layer sent bitrate/res).
+	// Call OCA.Talk.debugPublisherStats() to start, again to stop.
+	window.OCA.Talk.debugPublisherStats = function() {
+		if (window.__pubStats) {
+			clearInterval(window.__pubStats)
+			window.__pubStats = null
+			console.log('[pub] stopped')
+			return
+		}
+		const rtc = window.OCA?.Talk?.SimpleWebRTC?.webrtc
+		if (!rtc) {
+			console.warn('No SimpleWebRTC instance')
+			return
+		}
+		const publishers = rtc.peers.filter((p) => p.pc && p.pc.getSenders().some((s) => s.track && s.track.kind === 'video'))
+		if (!publishers.length) {
+			console.warn('No publishing video peer (are you sending video?)')
+			return
+		}
+		publishers.forEach((p) => p.pc.getSenders().forEach((s) => {
+			if (s.track && s.track.kind === 'video') {
+				console.log(
+					'[pub]',
+					p.id,
+					'configured:',
+					(s.getParameters().encodings || []).map((e) => ({ rid: e.rid, maxBitrate: e.maxBitrate })),
+				)
+			}
+		}))
+		const prev = new Map()
+		const history = new Map() // key -> rolling window of recent kbps samples
+		const WINDOW = 10 // ticks to average over (~20s at the 2s interval)
+		const avg = (arr) => (arr.length ? Math.round(arr.reduce((a, b) => a + b, 0) / arr.length) : '…')
+		window.__pubStats = setInterval(async () => {
+			for (const p of publishers) {
+				const rows = []
+				let totalCur = 0
+				let totalAvg = 0
+				;(await p.pc.getStats()).forEach((r) => {
+					if (r.type === 'outbound-rtp' && r.kind === 'video') {
+						const key = p.id + ':' + (r.rid ?? r.ssrc)
+						const p0 = prev.get(key)
+						const kbps = (p0 && r.timestamp > p0.ts)
+							? Math.round(8 * (r.bytesSent - p0.bytes) / (r.timestamp - p0.ts))
+							: null
+						prev.set(key, { bytes: r.bytesSent, ts: r.timestamp })
+						const h = history.get(key) || []
+						if (kbps !== null) {
+							h.push(kbps)
+							while (h.length > WINDOW) {
+								h.shift()
+							}
+							history.set(key, h)
+							totalCur += kbps
+							totalAvg += avg(h)
+						}
+						rows.push({
+							rid: r.rid ?? '-',
+							res: (r.frameWidth || '?') + 'x' + (r.frameHeight || '?'),
+							fps: r.framesPerSecond ?? '-',
+							cur_kbps: kbps ?? '…',
+							avg_kbps: avg(h),
+							active: r.active ?? '-',
+							limit: r.qualityLimitationReason ?? '-',
+						})
+					}
+				})
+				if (rows.length) {
+					console.log('[pub]', p.id, new Date().toLocaleTimeString(),
+						`— TOTAL cur ${totalCur} kbps / avg ${totalAvg} kbps (window ${WINDOW})`)
+					console.table(rows)
+				}
+			}
+		}, 2000)
+		console.log('[pub] polling every 2s, averaging last', WINDOW, 'ticks — call OCA.Talk.debugPublisherStats() again to stop')
+	}
+
 	signaling.on('pullMessagesStoppedOnFail', function() {
 		// Force leaving the call in WebRTC; when pulling messages stops due
 		// to failures the room is left, and leaving the room indirectly