majestic lite: is WebRTC intended for 8 MB NOR devices? (+37% since April)

Is WebRTC intended for the `lite` majestic variant on 8 MB NOR?

Not a bug report — a sizing question, with measurements, from someone who may
simply be holding it wrong.

majestic.<family>.lite has grown 37% since April:

2026-04-22    685,672 B
2026-08-15    811,176 B   (+125,504)
2026-08-29    938,548 B   (+127,372)

Section-level, the last step is .text +86,084 and .rodata +34,712. The new
dynamic symbols identify it fairly clearly:

mbedtls_ssl_conf_dtls_srtp_protection_profiles
mbedtls_ssl_get_dtls_srtp_negotiation_result
mbedtls_ssl_conf_dtls_cookies
mbedtls_ssl_conf_export_keys_ext_cb
evhttp_send_reply_chunk_with_cb, evdns_base_load_hosts, ...

plus strings /ws/webrtc, Adapt bitrate to WebRTC viewers. So: DTLS-SRTP and
libevent-based signalling — a WebRTC stack.

The question. On a gk7202v300 with 8 MB NOR, the rootfs partition is 5632k.
For devices whose video transport is external (ours is a separate agent speaking
WebTransport/QUIC; others use RTSP into an NVR, or RTMP), that WebRTC stack is
dead weight — but it is not optional: BR2_OPENIPC_MAJESTIC defaults to "lite",
we are already on it, and majestic is a prebuilt binary so we cannot configure the
feature out. The published alternatives are ultimate (larger) and fpv (a
reduced build we cannot use — we need the full audio and encoder pipeline).

Is WebRTC in lite deliberate for 8 MB devices, or would a variant without it
make sense — something like lite-nowebrtc? I am happy to be told this is the
intended direction and that 8 MB NOR is simply reaching end of life; that is a
perfectly good answer and I would rather know than guess.

For context, not as a complaint: we absorbed it. Upstream's base actually
shrank 464K in the same window, but every byte came from files we already
trimmed locally (libnl*, jshn, jsonfilter, libblobmsg_json,
libjson_script), so it bought us nothing while majestic's growth cost us in
full. We have since reclaimed 64k by shrinking the kernel partition (the uImage
measured 1,883,236 / 1,883,236 / 1,883,228 across three consecutive nightlies, so
1920k was over-provisioned). That gives us roughly one more cycle at the current
rate.

Asking because if the answer is "a no-WebRTC lite is reasonable", that likely
helps every 8 MB NOR device, not just ours.

Your measurements are right, and this is a good question to have asked.

Same numbers here, against today's nightly — note gk7202v300 builds the
gk7205v200 majestic family, so we are looking at the same binary:

lite       938,684 B
fpv        515,920 B
ultimate 1,439,200 B

And yes, that is WebRTC: DTLS-SRTP, ICE, and the cloud signalling path. The
mbedtls dynamic symbol count goes 28 → 88 between fpv and lite, which is the
DTLS-SRTP surface arriving.

Is it deliberate? Yes. The three flavours are curated by what the camera is
for, not by how much flash it has — that is now written down in
Majestic Streamer → Lite, Ultimate and FPV. Lite is the everyday
build: audio, talkback, RTMP, WebRTC. FPV is the latency-first build with audio,
WebRTC, SIP and RTMP compiled out. There is no size tier, and that is on purpose.

Would a lite-nowebrtc make sense? No, and the reason is worth explaining,
because it is about direction rather than bytes. WebRTC is no longer an optional
extra sitting next to the others. It is the WebUI's default preview transport —
MSE and MJPEG are the fallback path now, not the other way round — and it is how
cloud.enabled reaches a camera without port-forwarding. The optimisation work
queued behind it is aimed squarely at cellular and at OpenIPC Cloud, where
WebRTC's congestion control and NAT traversal are the entire point. A build
without it would get steadily less of what we are actually working on, and Lite
is published for every supported SoC — so a fourth flavour that had to track
Lite's coverage is the expensive one to add, not FPV's.

Is 8 MB NOR reaching end of life? Not today, and there is no plan to drop it.
For what it is worth, upstream is not comfortable on your board either — nightly
2026-08-30, gk7202v300 lite:

rootfs.squashfs  5,013,504 B / 5120 KB cap  →  224 KB headroom
uImage           1,883,212 B / 2048 KB cap  →  209 KB headroom

So you are not an outlier, and the caps are enforced per build, which means 8 MB
cannot silently stop working — it would fail a build first. What I will not
promise is that every future feature gets compiled out to keep it fitting. If
there is a hardware refresh anywhere in your roadmap, 16 MB is cheap insurance.
Your kernel-partition reclaim is sound reasoning, incidentally; make BOARD=<x> size-report will show you where the rest of it is going.

On getting the build you actually want. You describe a device with its own
transport agent and a private trim of the rootfs, which reads like a product you
ship rather than a hobby camera. If that is right, then the route to a Lite
without WebRTC already exists and it is not a public flavour: Majestic is
Prosperity Public License 3.0.0 — free forever for personal and
noncommercial use, thirty-day commercial trial — and commercial licenses include
custom builds with feature selection. That is a better answer than
lite-nowebrtc would be, because you would get exactly the set you want, with
the full audio and encoder pipeline intact, rather than a compromise shared with
every other 8 MB device. business@openipc.org if you want to talk about it.

If it is not right and this is noncommercial, say so and ignore that paragraph —
the rest of the answer stands either way.

I was wondering what your thoughts on using WebTransport for majestic over WebRTC? QUIC is quite stable now and HTTP3 is a much lighter technology to use, not to mention is great for Client Server exchanges

Thank you — that is a much better answer than the one I was fishing for, and the
reasoning about direction rather than bytes is the part I needed to hear.

On the licensing paragraph: it does not apply. This is a hobby project. Three
boards on a bench, no product, nothing sold or planned to be. The "own transport
agent and private rootfs trim" reads commercial because I have been over-engineering
a home camera for months, not because anyone is paying for it. I would rather say
that plainly than let an ambiguity sit in a public thread.

So lite with WebRTC is simply what I get, and that is fine. I will stop trying to
shrink my way out of it.

Two corrections I am taking away, both mine:

fpv was on my shortlist until you said audio is compiled out. I had assumed it
was lite-minus-latency-features. Audio is load-bearing here — the camera does
on-device audio classification — so that would have been a slow, confusing
discovery.
I had my kernel headroom backwards. I reclaimed 64k from the kernel partition
(1920k → 1856k) after verifying my uImage fits, leaving 17,308 B spare. But you
size against a 2048k cap and are at 1,883,212 — so you have ~214k of room you may
legitimately grow into, and roughly 196k of that would not fit my partition. My
reclaim is safe against today's kernel and fragile against yours. It fails a build
gate rather than a board, but I would not have known to look without your numbers.

make BOARD=<x> size-report is now on my list; I had not been running it.

The part I actually want to ask about

You mentioned WebRTC's congestion control and NAT traversal being the point for
cellular and Cloud. That is adjacent to something I know well.

My day job is transport: WebTransport and QUIC, in depth. The agent on these
cameras is a Rust binary that publishes H.264/H.265 and Opus over WebTransport to a
cloud relay, with a datagram path for intercom, and it has been running on 8 MB
gk7202v300 hardware for months — so I have practical numbers on what QUIC costs on
a 64 MB, single-core ARMv7 camera, not just opinions.

Where that might be useful to OpenIPC, if any of it is interesting:

WebRTC and WebTransport share the hard parts — congestion control, pacing,
loss recovery, MTU discovery. If the Cloud work is tuning any of that, I am happy
to review, test on real constrained hardware, or contribute.
WebTransport as a publish path is genuinely simpler than WebRTC when the
camera talks to your infrastructure rather than to a browser peer: one QUIC
connection, no ICE, no SFU, no DTLS-SRTP. It is not a WebRTC replacement — it does
not do browser-to-camera P2P — but for camera-to-cloud it removes most of the
moving parts. Whether that is worth a flag in majestic is your call entirely; I
mention it as something I could prototype rather than something I think you are
missing.
Congestion control on cellular is where I would expect to be most useful.
BBR-ish behaviour on a link that is simultaneously lossy and buffer-bloated is
a place where good defaults matter more than features.

Entirely understood if none of this fits your roadmap, or if a transport
contribution from someone who has been here about three weeks is premature. But I
have enjoyed the work — the DVP gate landing in #2276 came out of exactly this kind
of exchange — and I would rather ask where help is wanted than turn up with an
unsolicited PR.

Thanks for saying it plainly — and no harm done. I wrote that paragraph conditional
precisely because I could not tell, and "over-engineering a home camera for months"
is a description I recognise from the inside. Hobby it is.

It also changes the answer in your favour, because nothing is stopping you stripping
the image down to what you actually run.

The thing you should delete is not WebRTC

You have been trying to shrink the streamer, which is the one part you cannot
change. Meanwhile you are shipping a web interface that, by your own description,
nothing in your setup uses — your agent is the interface.

I went and measured this properly rather than estimating, because uncompressed
byte counts are misleading here and I did not want to hand you another wrong number
after you already corrected one of mine. Real squashfs, your board, nightly
2026-08-30, against the 5120 KB cap:

gk7202v300 lite	rootfs.squashfs	headroom
as shipped	5,013,504	224 KB
WebUI removed	4,882,432	352 KB
majestic → fpv	4,755,456	476 KB
both	4,624,384	604 KB

Dropping the WebUI is worth 131,072 B — exactly 128 KB of flash. That is half of
everything the entire FPV downgrade would buy you, and you keep audio, WebRTC, RTMP
and SIP. Given audio is load-bearing for your classifier, it is the whole of what you
can actually collect.

Same measurement on hi3516ev200, to check the number was not an artefact of one
board: 4,837,376 → 4,706,304. Also 131,072 B. It is one squashfs block, which is why
it lands so round.

The method, since you will want to check it rather than take my word:

fakeroot -s fr.state unsquashfs -d root rootfs.squashfs.gk7202v300
# ... edit root/ ...
fakeroot -i fr.state mksquashfs root out.sq -noappend -b 131072 -comp xz

Rebuilding the unmodified tree that way reproduces the shipped image byte-exactly
on both boards — 5,013,504 and 4,837,376 — so the deltas above are real flash, not a
compression estimate.

One incidental finding from doing this: between the 08-30 nightly and the 08-31
publish, majestic grew 12,580 bytes of binary, which landed as 4,096 more bytes of
flash. In a day. Your instinct that this is worth watching is correct.

How to turn it off: one line in your board defconfig.
gk7202v300_lite_defconfig has BR2_PACKAGE_MAJESTIC_WEBUI=y at line 63. haserl
is selected by that symbol, so it leaves with it.

What survives — the part worth checking before you pull the trigger:

Majestic's HTTP API, RTSP, ONVIF, snapshots, HLS, the WebSocket endpoints. All
of that lives in the majestic binary, not in majestic-webui. Your agent keeps
everything it talks to.
First-boot claiming. /setup.html and POST /setup are served by majestic
itself — I checked, there is no setup* anywhere under /var/www. A camera with
the WebUI removed can still be claimed from a browser, and openipc-claim over
SSH is untouched regardless.

What you lose is the browser UI: settings pages, preview, log viewer. If your agent
covers that, it is 128 KB you are carrying for nothing.

One caveat on the fpv row above, so you do not read more into it than it holds:
that is WebRTC and audio and SIP and RTMP removed together. Majestic ships
prebuilt, so I cannot isolate WebRTC on its own — 252 KB is an upper bound on the
whole bundle, not a price tag for WebRTC.

Why the protocol set is what it is

WebRTC is the only sub-second path a browser gives you natively. No plugin, no
polyfill, hardware-accelerated decode, congestion control running end to end to the
actual viewer. Nothing else in a browser does that, which is why it became the
default preview transport rather than one option among several.

MSE over WebSockets is the other half, and it is deliberate. It works in every
browser and every awkward network, costs the camera almost nothing, and buys a
second or two of latency. MSE is the floor; WebRTC is the ceiling.

Those two are what we can afford, and that is a flash decision rather than a taste
one — every protocol is permanently resident in a binary with roughly half a megabyte
of flash budget on the smallest boards we still ship. There is no third slot. Picking
one would mean deleting one of those two, and neither can go.

For the shrinking hobby specifically

You will get more out of this than out of size-report:
https://openipc.github.io/firmware-explorer/

It renders each nightly × platform from the sizes.<plat>.json sidecar the build
publishes: a treemap of packages by bytes, a sortable package table that expands to
per-file detail, a module table flagging which .kos are autoloaded from
/etc/modules versus sitting there on demand — that filter is a good compaction-
candidate finder — a panel showing what the finalize hooks already stripped, and a
drift view that diffs any two builds for a platform sorted by byte delta.

The drift view is the one for you. It would have shown you the majestic step as a
single row, and it will show you the next one before it costs you an evening of
readelf.

On WebTransport and QUIC

Straight answer, since you asked straight: not as a third transport in majestic,
and the reason is the flash paragraph above rather than anything about the design.
You are right that camera-to-cloud over one QUIC connection removes most of the
moving parts — no ICE, no SFU, no DTLS-SRTP — and if the budget were different that
would be an interesting argument to have. It is not, so it is not.

Where your expertise lands is the part you identified yourself. Congestion
control, pacing, loss recovery and MTU discovery are shared between the two, and that
is where the Cloud and cellular work actually is. BBR-ish behaviour on a link that is
lossy and buffer-bloated at the same time is exactly the problem, and "good defaults
matter more than features" is the correct instinct.

The scarce thing there is not opinions — it is numbers from real constrained
hardware, and you have months of them.

So: please post the numbers. Concretely, what I would find most useful —

CPU cost of the QUIC path under sustained publish. What fraction of that single
ARMv7 core goes to transport rather than to the encoder, at what bitrate.
Resident memory of the Rust agent, and what it peaks at when the link degrades.
What the congestion controller does when the cell link goes bad — how fast it
backs off, how fast it recovers, whether it oscillates.
Anything that surprised you about QUIC on a 64 MB device. Handshake cost, crypto
without AES acceleration, buffer sizing.

Nobody has published what this costs on this class of chip. That write-up is worth
something on its own, whether or not a line of it ever becomes a majestic patch, and
it is the input that would make a congestion-control conversation concrete instead of
theoretical. An issue here is a fine place for it, or the wiki if it grows.

I am not going to promise you a roadmap slot in a GitHub thread. But three weeks in
with #2276 already merged is not "premature", and if you want to go deeper on the
Cloud side than an issue comment supports, d.ilyin@openipc.org.

Leaving this open for follow-ups; the WebRTC question itself is answered.

Correction to my last comment. I said I could not isolate WebRTC from the prebuilt
binaries and that the 252 KB fpv delta was an upper bound on the whole bundle. That
was true of the published tarballs, but it was the wrong place to stop — I can build
majestic both ways. So I did.

hi3516ev200, lite, Release, stripped, same toolchain and the same dependency tree,
with exactly one build option changed. The ENABLE_RTC=ON build comes out at
946,188 bytes, the same size as the published lite, so this is the shipped
configuration rather than an approximation of it.

WebRTC costs 82,376 bytes of binary, and 44 KB of flash

section	RTC=ON	RTC=OFF	delta
`.text`	626,278	573,934	+52,344
`.rodata`	244,820	224,012	+20,808
`.data`	7,476	3,172	+4,304
dynamic linking¹	—	—	+3,696
`.bss`	65,400	65,396	+4
binary	946,188	863,812	+82,376

¹ .dynstr +1,264, .dynsym +800, .plt +616, .rel.plt +408, .got +204,
.hash +200, .rel.dyn +104, .gnu.version +100, .data.rel.ro +56.

In the rootfs, which is the number that decides whether your image fits — same
squashfs method as before, baseline reproducing the shipped image byte-exactly:

hi3516ev200 lite	rootfs.squashfs	headroom
RTC=ON (as shipped)	4,841,472	392 KB
RTC=OFF	4,796,416	436 KB

45,056 bytes. 44 KB of flash — 8.7% of the binary, 0.9% of the rootfs.

A footnote while I had the two builds side by side: 8,440 of those 82,376 bytes are
the SoC crypto engine path behind the SRTP transform, which is compiled in on
hi3516ev200 and hi3516cv500 only. It makes no difference to the squashfs at all —
it is smaller than the 128 KB block granularity.

What that changes

The advice in my last comment holds, and gets stronger. Ranked by what each is
actually worth on your board:

	flash
The WebUI you are not using	128 KB
Everything `fpv` removes (WebRTC + audio + SIP + RTMP)	252 KB
WebRTC alone	44 KB

WebRTC is about a sixth of the fpv delta, and the WebUI is worth nearly three times
WebRTC on its own. If you had somehow got the lite-nowebrtc you originally asked
for, it would have bought you 44 KB — less than a third of what deleting a web
interface you never open buys you, and you would have given up the browser path
permanently to get it.

So the thing worth deleting was never the streamer. That was the answer I gave you
before I had the number; I am glad the number agrees, because it could easily have
gone the other way and I would have owed you a different comment.

One correction to your figures

You attributed .text +86,084 and .rodata +34,712 to the WebRTC step. WebRTC
itself is .text +52,344 and .rodata +20,808 — so roughly 60% of that step was the
WebRTC stack landing, and the other 40% was unrelated work that happened to land in
the same window. Your identification was right; the attribution was a bit generous to
WebRTC. Worth knowing if you are extrapolating a growth rate from that step, because
it means the per-feature cost is lower and the general drift is higher than it looked.

Still very much interested in the QUIC numbers whenever you get to them.

The QUIC numbers, as asked. All measured today on a GK7202V300 bench board; where
something is inferred rather than measured I say so.

Hardware. GK7202V300, ARMv7l Cortex-A7, one core, USER_HZ 100. CPU features:
half thumb fastmult vfp edsp neon vfpv3 tls vfpv4 idiva idivt vfpd32 lpae evtstrm —
no aes, no pmull, no sha1/sha2. The ARM crypto extensions are ARMv8-A;
nothing on this class of chip has them. /proc/crypto shows aes-generic.

One correction to the premise, and it matters for the memory numbers: these are sold
as 64 MB boards, but MemTotal is 35,268 kB. The rest is reserved for the ISP/MMZ
before Linux sees it.

Stack. Rust agent: wtransport 0.7.2 → quinn 0.11.11 / quinn-proto 0.11.17,
rustls 0.23.43 with the ring provider. No TransportConfig override, so this is
quinn's defaults — Cubic, default windows. Only keep_alive_interval (3s) and
max_idle_timeout (30s) are set. Publish is H.265 640×360@12 + AAC to a cloud relay;
the same binary also serves 1080p H.264 over WebTransport on the LAN.

Wire format, because it turns out to be the dominant cost: every access unit —
each video AU, each audio frame — is sent as its own unidirectional QUIC stream.

1. CPU: what fraction of the core is transport

25-second windows, utime+stime from /proc/<pid>/stat, single core so percentages
are of the whole machine. The publish-only row is n=3, reproduced within ±0.2.

state	agent	majestic	wlan0 TX
agent stopped — encoder only	—	19.5%	0
cloud publish only (74 kbit/s media)	8.4%	21.9%	~105 kbit/s
+ 1 LAN viewer, substream	12.5%	23.8%	192 kbit/s
+ 1 LAN viewer, 1080p H.264	20.1%	24.4%	1292 kbit/s

At a realistic load the split is roughly 20% encoder, 20% transport — transport is
not a rounding error next to the encoder, it is the same order of magnitude.

majestic costs ~2.4 points just to serve the loopback RTSP tap (19.5% → 21.9% with
no network client involved). If you are counting the price of an agent-style
architecture, that is part of it and it is not obvious.

Solving for the marginal cost using the two viewer deltas as a 2×2 (per-stream α,
per-byte β):

α ≈ 0.156% of core per stream/second
β ≈ 0.0046% of core per kbit/s — i.e. 4.6% of the core per Mbit/s

Those two terms come out equal at ~1.35 Mbit/s for a 40 stream/s feed. Below that,
stream count dominates the bytes. A 12 fps substream at 74 kbit/s spends more core
on stream setup/teardown than on its own payload.

That is the most useful thing I have learned running QUIC here, and it is a design
lesson rather than a tuning one: one-stream-per-frame is clean, gives you per-frame
framing for free, and lets a lagging reader drop whole frames — but on this class of
CPU you pay for it per frame, not per byte. If I were starting again I would put
frames on a small number of long-lived streams and frame them myself.

Caveat on the 8.4% baseline: that is the whole agent, which also does RTSP demux
and runs an on-device audio classifier. It is not 8.4% of pure transport. The α and
β figures are clean, because they are deltas with everything else held constant.

2. Memory


`VmRSS` steady	4.3–4.5 MB
`VmHWM`	6.0 MB
`VmSize`	15.3 MB

Under a degraded link VmHWM did not move. During sustained radio saturation RSS
oscillated between 4,336 and 5,532 kB and the high-water mark stayed pinned at exactly
6,000 kB — the same value it had before the test. Whatever else QUIC costs here, it
does not balloon its buffers when the link goes bad. On a 34 MB box that was what I
was most worried about, and it turned out to be a non-issue.

A trap for anyone measuring this: ps reports %MEM from VSZ on busybox. It
showed the agent at 43% of the box. Real resident is 12%.

3. Handshake

Full QUIC + WebTransport establishment, camera as the TLS server (so the camera
does the server-side signature), over WiFi, software crypto only:

trial 1: session=0.065s config=0.111s first-video=1.058s FIRST-KEYFRAME=1.103s
trial 2: session=0.082s config=0.134s first-video=0.201s FIRST-KEYFRAME=0.201s
trial 3: session=0.066s config=0.122s first-video=0.183s FIRST-KEYFRAME=0.183s
trial 4: session=0.054s config=0.116s first-video=0.168s FIRST-KEYFRAME=0.168s
trial 5: session=0.070s config=0.168s first-video=0.272s FIRST-KEYFRAME=0.272s
trial 6: session=0.092s config=0.131s first-video=0.204s FIRST-KEYFRAME=0.204s

54–92 ms, median ~68 ms. I expected worse without AES acceleration. Handshake cost
is not an argument against QUIC on this hardware.

(Trial 1's 1.06s first-frame is a cold replay ring waiting on a real IDR, not
transport. Warm joins are 168–272 ms.)

4. Crypto without AES acceleration

No openssl on the image, so I cross-compiled a benchmark against the same ring
0.17 the QUIC stack actually uses, sealing 1200-byte packets (QUIC packet sized, not
bulk):

AEAD	throughput	per packet
AES-128-GCM	72.8 Mbit/s	131.9 µs
AES-256-GCM	58.7 Mbit/s	163.4 µs
ChaCha20-Poly1305	166.0 Mbit/s	57.8 µs

ChaCha20-Poly1305 is 2.3× AES-128-GCM on this chip. NEON carries ChaCha20; AES has
nothing to stand on and falls back to constant-time software. This inverts the usual
server-side ordering, where AES-NI makes AES-GCM the obvious default.

The actionable part: in TLS 1.3 the server picks the suite. A camera talking to
a cloud that prefers AES-GCM pays 2.3× for record protection and has no say in it.
Anywhere OpenIPC controls both ends — Cloud especially — preferring
ChaCha20-Poly1305 for these SoCs is free performance. I would expect the same to hold
for SRTP in the WebRTC path, though I have not measured that.

But scale it before acting on it: at 1.3 Mbit/s, AES-128-GCM costs ~1.8% of the
core and ChaCha20 ~0.8%. Against a 20% transport total, crypto is not the
bottleneck — the per-stream state machine work is, by roughly an order of magnitude.
I went in assuming crypto would dominate on a chip with no AES. It does not, and that
surprised me more than any other result here.

5. Behaviour when the link degrades — two caveats first

The image has no tc, so I could not use netem. I degraded the link physically
instead: three cameras streaming 1080p simultaneously on one 40 MHz channel (RSSI
−60/−63/−68, negotiated 54–150 Mbps). That is real contention, but it is airtime
starvation, not the lossy-and-bufferbloated cellular case you actually care about.

More important: what I can observe is my application's send-window-exhaustion
signal, which sits downstream of quinn's congestion control. I have not instrumented
quinn's cwnd/RTT directly, so I cannot tell you what Cubic is doing underneath —
only what the application feels.

With that said, the signal oscillates, reproducibly:

19:54:12  uplink pressure   → throttle
19:54:35  pressure cleared  → restore      (23s)
19:54:38  uplink pressure   → throttle     (3s later)

An earlier independent run gave the same shape with a 1.5s re-entry. So: it backs off
promptly, holds while contended, and re-enters pressure within 1.5–3 s of clearing
— because my clear condition is a 5-second quiet window and the link re-saturates
almost immediately once the bitrate is restored. That is my hysteresis being too
optimistic rather than obviously a CC pathology, but it is exactly the "does it
oscillate" question, and on a contended link the answer is yes, on a 1.5–3 s cycle.

Instrumented cwnd/RTT/loss traces off this hardware are a small patch to my agent if
that would be more useful than my application's shadow of the controller — including
against a real cellular modem rather than contended WiFi.

Summary

Transport is ~1× the encoder on this class of chip, not a rounding error.
Per-stream cost dominates per-byte cost below ~1.35 Mbit/s. Protocol shape
matters more than bitrate at camera bitrates; anything that opens a stream or
channel per frame will hurt here.
Prefer ChaCha20-Poly1305 wherever OpenIPC controls the server side. 2.3× on the
crypto line, free.
Memory is not the problem. 4.3 MB resident, 6.0 MB peak, stable under
degradation, on a box with 34 MB.
The 64 MB on the box is 35,268 kB to Linux.

The crypto benchmark is ~40 lines if it is worth having in-tree — that table for every
SoC OpenIPC ships would be more useful than my one row of it.

This is the best data anyone has put on this class of hardware, and the caveats are
the part that makes it usable — you flagged every place the measurement stops short of
the claim, which is rarer than it should be. Some of it I could check against our own
bench, so here is what corroborates, what changes, and the one experiment I think you
should run before acting on your own headline.

The per-stream result needs one more experiment before you redesign

Your α/β solve is internally consistent — I checked the arithmetic on all of it and it
holds, including the 1.35 Mbit/s crossover and the 1.8%/0.8% crypto scaling. And the
strong form of the conclusion does not even need the fit: 87 kbit/s costs 4.1 points
and 1100 kbit/s costs 7.6, so a 12.6× increase in bytes bought a 1.85× increase in
cost. Whatever dominates at these bitrates, it is not the bytes. That much is solid.

But "per stream" is not the only thing that varied. The 2×2 attributes every
non-byte cost to stream count, and the experiment cannot separate QUIC stream
lifecycle from everything else that happens once per access unit — RTSP demux, the
copy out of the tap, your classifier's per-frame path, allocator churn. Those all scale
with AU rate and none of them go away when you multiplex frames onto a long-lived
stream. If they are the bulk of α, your redesign buys you the framing you write
yourself and very little CPU.

The clean experiment is one binary with two modes, same feed, same bytes, same frame
rate, changing only the mapping: N frames → N streams, versus N frames → 1 stream with
your own length prefix. If α collapses, your instinct was right and it is worth the
rewrite. If it barely moves, the cost is per-AU and lives somewhere else entirely.
That is a couple of hours and it converts a design intuition into a measurement.

One input missing from the table, and it is the one the fit depends on: the actual
AU rates for each row. Your constants imply roughly 24 streams/s for the substream
viewer and 16/s for the 1080p one — the substream opening more streams per second
than the main one is exactly what makes the system solvable rather than degenerate, so
if those rates are not right the constants move a lot. Worth publishing the per-row
frame rates alongside the bitrates.

You can have the 2.3× today, without anyone else changing anything

You are right that in TLS 1.3 the server picks the suite. But it picks from what the
client offered, and the client controls that list — in rustls, restrict the provider's
cipher_suites to TLS13_CHACHA20_POLY1305_SHA256 and no cloud that wants to talk to
you can choose AES-GCM, because you never offered it.

So for your own agent this is not a request to anyone; it is a two-line change and you
keep the 2.3×. The general recommendation for OpenIPC Cloud stands and is well made,
but you do not have to wait for it.

It will not transfer to SRTP, and the reason is worth knowing

DTLS-SRTP negotiates protection profiles from a fixed list — AES-128-CM with
HMAC-SHA1 (RFC 5764) and the AES-GCM profiles (RFC 7714). There is no deployed
ChaCha20 SRTP profile. Both ends must support whatever is chosen, so unlike TLS there
is no list you can trim to force the fast cipher. The WebRTC path is stuck with AES on
a chip that has no AES.

What we do about it instead is the SoC crypto engine. Measured on gen-4 HiSilicon
parts, per 1100-byte packet, software → engine:

86.4 µs → 38.4 µs
87.6 µs → 42.0 µs

A bit over 2× — the same order as your ChaCha win, arrived at from the other
direction. Note this lands close to your software AES-128-GCM figure of 131.9 µs per
1200-byte packet, on a different vendor's silicon, which is a nice independent check
that software AES on this generation is simply expensive.

Two things that will interest you given your "scale it before acting on it" instinct.
A gen-3 part measured slower through its engine than in software, so we do not build
it there. And it is enabled on two boards only — Goke, including your GK7202V300, is
not one of them. Your chip has neither AES acceleration nor an engine we use.

Memory: reproduced on our bench, with one correction that matters

Our lab GK7205V200 — same family as your board — reports MemTotal 35,232 kB
against your 35,268. Within 36 kB, so your board and ours are configured the same way.

That configuration is not the shipped default. osmem falls back to 32M when
unset, which gives about 27 MB of userspace; you are on 40M, as our bench is. Anyone
budgeting from your numbers on a stock image has roughly 8 MB less than you did, which
makes your 4.3 MB resident a materially bigger slice of the box than it looks here.
Your result is still the reassuring one — it just deserves that asterisk.

An incidental echo: majestic on that same board sits at VmRSS 4,704 kB / VmSize
15,000 kB. Your agent is 4.3–4.5 MB / 15.3 MB. Two unrelated processes landing on the
same footprint is musl's arena behaviour showing through in both.

Your ps warning is a good one and matches our experience — VmRSS from
/proc/<pid>/status is the only number worth reading on these boards.

On `tc`: confirmed absent, and here is the way around it

Not an oversight you can work around on-box. CONFIG_TC is off in our busybox config,
there is no iproute2 package in the tree, and the only thing shipped is busybox ip,
which does addresses, links and routes but no qdiscs. There is nothing on the camera
that can shape.

The practical answer is to move the shaping off the camera: put it behind a Linux box
acting as its AP or as a bridge, and run netem on that interface. You get real loss,
delay and queue depth under your control, applied to the camera's actual uplink, and
you can reproduce bufferbloat properly rather than borrowing airtime contention as a
stand-in. Your contended-WiFi result is honest about being the wrong shape; a bridge
in the middle gets you the right one for a few minutes of setup.

And yes — instrumented cwnd/RTT/loss traces would be much more useful than the
application's shadow of the controller, especially against a real modem. That is the
gap between "my hysteresis is too optimistic" and "here is what Cubic does on a
cellular link", and only the second one tells us anything we can act on.

The benchmark: yes, and it is worth more than one row

A per-SoC AEAD table is the most useful thing in your whole message, because it feeds a
decision we currently make by hand. Hardware crypto is switched on for two boards
because someone measured those two boards; every other SoC we ship is an assumption.
A tool anyone can run, producing a row per chip, turns that into data.

Probing and identification tooling lives in ipctool rather than here, and
that is the natural home for something you run on a camera to find out what it is
capable of — with the resulting table on the wiki. Worth opening the
conversation there; point at this thread for the motivation.

One finding of yours that is about us, not you

19.5% → 21.9% for majestic to serve a loopback RTSP tap with no network client is
the number I did not expect, and it is the one I am taking away from this. 2.4 points
of a single core to move frames to a local consumer is worth understanding on its own
terms, independently of anything QUIC.

Thank you for measuring all of this properly. Very little of what reaches us has error
bars on it.

Three of these I could act on immediately, and one of them turned into a correction
to your ChaCha suggestion that I think is worth having on the record.

The missing input: per-row AU rates

You are right that the fit depends on these and I should have published them. Measured
by the client, over the same 26 s windows:

row	wlan0 TX	streams/s	of which video	audio
+ 1 LAN viewer, substream (video1, 12 fps cap)	192 kbit/s	23.8	8.2	~15.6
+ 1 LAN viewer, 1080p (video0, 25 fps)	1292 kbit/s	40.0	24.6	~15.4

One correction: you inferred ~24/s for the substream and ~16/s for the 1080p viewer,
and read the substream opening more streams than the main one as the thing that makes
the system solvable. It is the other way round — 23.8 vs 40.0, the substream opens
fewer. Solving with the measured rates gives α = 0.156 %core per stream/s and
β = 0.0046 %core per kbit/s, which is what I quoted; your arithmetic check on the
constants was right, the rates behind them just were not stated.

Note the audio stream rate is nearly identical in both rows (AAC at 1024 samples /
16 kHz ≈ 15.6/s regardless of video), so almost all of the stream-rate delta is video
frames. That is convenient for the experiment you propose: it isolates cleanly.

Your experiment is the right one and I have not run it yet

I am not going to pretend otherwise. Your objection is correct and it is the one that
matters: the 2×2 attributes everything that happens once per access unit to "stream
count", and RTSP demux, the copy out of the tap, the classifier's per-frame path and
allocator churn all scale with AU rate and survive multiplexing. If those dominate α,
the rewrite buys framing I have to write myself and no CPU.

One binary, two modes, same feed, N→N vs N→1 with my own length prefix, is exactly the
right shape and I will run it. I would rather report that honestly than let the
"per-stream" framing harden into a design decision on the strength of an experiment
that cannot separate the two. I will post the result either way — including if it says
my instinct was wrong, which is the outcome that would save me the rewrite.

The ChaCha20 change does not work on QUIC, and the reason is structural

This is the one I want to flag, because your reasoning is right for TLS and I
implemented it verbatim before finding out it does not carry over.

Restricting the rustls provider to TLS13_CHACHA20_POLY1305_SHA256 does not fail to
negotiate — the endpoint refuses to construct at all:

panicked at wtransport-0.7.2/src/config.rs:1085:
CipherSuite::TLS13_AES_128_GCM_SHA256 missing: NoInitialCipherSuite { specific: false }

QUIC derives Initial packet protection keys per RFC 9001 §5.2, and that is hardwired to
AES-128-GCM — before any negotiation happens, and independently of what the handshake
later selects. So a QUIC client cannot drop AES from its provider the way a TLS client
can. The suite list has to keep AES-128-GCM in it for the transport to exist.

What is available is preference: offer ChaCha20 first and AES-128-GCM second. rustls
sends them in that order and most servers honour client order, but the server still
picks, so it is a preference rather than the unilateral guarantee. I have shipped that
ordering. I could not measure it: at the publish bitrate the crypto line is ~0.14 % of
a core, an order of magnitude under my ±0.2 measurement noise, so agent CPU read
8.4/8.2/8.2 % before and after. The 1.8 % vs 0.8 % figures only separate at 1.3 Mbit/s.

Two consequences worth drawing out, since they cut against the optimistic reading of my
own crypto table:

The chip has no AES hardware, no crypto engine you build for, and cannot decline
AES for QUIC Initials. Some AES on this silicon is unavoidable.
It is a small fraction of the packets — Initials only — so the practical cost is
negligible. But "just force ChaCha20" is not available to a QUIC endpoint, and I
would not want that repeated as advice without the caveat.

Accepted without reservation

SRTP. You are right and I was speculating past my evidence. DTLS-SRTP negotiates
from a fixed profile list with no deployed ChaCha20 option, and unlike TLS there is no
list to trim. I should not have extended a TLS-shaped conclusion to it. Your engine
numbers (86.4 → 38.4 µs, 87.6 → 42.0 µs per 1100 B) landing near my 131.9 µs/1200 B
software AES figure on different silicon is a good independent check, and the gen-3
part being slower through its engine is exactly the sort of result that only shows up
if someone measures instead of assuming.

The osmem asterisk is important and I should have caught it. Confirmed on my
board — bootargs=mem=40M …, which is where 35,268 kB comes from. Anyone budgeting
from my numbers on a stock 32M image has ~8 MB less than I did, and my 4.3 MB resident
is a materially larger slice of that box. The reassuring conclusion survives, but it
deserves the caveat and the caveat is yours, not mine.

Your GK7205V200 reporting 35,232 kB against my 35,268 is a nice confirmation that the
two benches are configured the same.

tc. Thank you for confirming it is absent by construction rather than by
oversight. Shaping from a Linux box acting as AP or bridge is obviously the right
answer and I will set that up — it gets me real loss, delay and queue depth on the
camera's actual uplink instead of borrowing airtime contention as a stand-in, which I
knew was the wrong shape and said so mainly because I had nothing better.

That plus instrumented cwnd/RTT/loss is the combination worth doing properly, and I
agree it is the only version that says anything actionable about Cubic on a cellular
link. Currently I can only report what the application feels.

The benchmark, and ipctool

Agreed that a per-SoC table beats one row, and that "hardware crypto is on for two
boards because someone measured those two boards" is exactly the kind of thing a
runnable tool fixes. I will open the conversation on ipctool and point at this thread.

The benchmark is ~40 lines against ring, sealing packet-sized buffers rather than bulk,
which matters — bulk numbers flatter AES on chips where the per-call overhead dominates.
Happy to shape it to whatever ipctool wants for probe output.

On majestic's 2.4 points

Glad that one is useful, and I would not over-read it from a single board yet. It is
19.5 % → 21.9 % of one core measured as majestic's own utime+stime across a 25 s
window, with the agent stopped versus running and no network client attached — so
it is the cost of serving one loopback RTSP consumer, nothing more. n=1 board, one
encoder config (H.265 640×360@12 + AAC). If it reproduces on your bench I would trust
it; if it does not, mine is the one to doubt.

Ran it. You were right and my headline was wrong: multiplexing recovers about a
tenth of what I attributed to stream count. Posting the number that kills my own
conclusion, as promised.

Method

One binary, two modes, selected by the connect URL so nothing else differs:

A — one uni stream per access unit (what ships)
B — ?mux=1, everything on one long-lived uni stream, [u32 BE len][payload]

Same feed, same encoder, same CONFIG, same replay ring, same broadcast receiver, same
frames in the same order. Smoke test before measuring, to confirm the arms were
actually comparable rather than assuming it:

A: held 10s: 406 streams (249 video)
B: held 10s MUXED: 406 frames (249 video) on 1 stream

Identical counts. Then 11 interleaved 22-second windows on the GK7202V300, video0 at
25 fps, agent CPU from utime+stime in /proc/<pid>/stat.

Result

A (stream per AU)   23.62% +/- 0.76 of core    2079 kbit/s    597 video frames/window
B (one stream)      23.55% +/- 0.94 of core    2182 kbit/s    605 video frames/window

Scene motion moved the bitrate between runs, so regressing CPU on bitrate and arm
rather than comparing means directly:

multiplexing removes 0.70 points (SE 0.25, t = −2.75, 95% CI [−1.19, −0.20]) out of
~23.6. Small, but not zero.

Against that, the 2×2 attributed 6.95 points to "per stream" at ~45 streams/s
(α = 0.156 × 45). So:

QUIC stream lifecycle is ~10% of what I called per-stream cost
~90% is per-access-unit work that survives multiplexing — RTSP demux, the copy
out of the tap, allocator churn, the broadcast hop
even the optimistic end of the CI only recovers 17%

Your framing was exactly right: the experiment could not separate the two, and it turns
out to be almost entirely the half that multiplexing does not touch.

So I am not rewriting the wire format. It would buy ~3% of one core in exchange for
hand-rolled framing and losing the drop-on-saturation policy that keeps a slow reader
from stalling the session. One-stream-per-frame stays.

Two honest caveats about arm B

It is not a clean isolation of QUIC stream machinery. Removing the per-AU stream
also removes the per-frame tokio::spawn and the bounded-inflight drop policy — a
length-framed stream desyncs if you skip an item, so B blocks where A drops. Those three
travel together; that IS "one stream" as anyone would build it. So the measurement
answers the design question, not "what does open_uni cost in isolation". Since the
answer came back negative, the distinction does not change the decision.

One A run is excluded. It delivered 797 streams against ~974 for every other run,
with two gaps >500 ms — it dropped frames, so it was doing less work and its lower CPU
was not comparable. Worth flagging that excluding it cuts against my conclusion: it
was the cheapest A run, and dropping it makes A look more expensive, not less.

The AU rates, since they were the missing input

Both arms, per 22 s window: ~600 video frames plus AAC at ~15.6/s, so ~45 streams/s in
arm A. Arm B opens one stream for the whole session.

What I take from this

A two-point solve names a correlate, not a mechanism. The constants were arithmetically
fine — you checked them and so did I — and the strong form held too: 12.6× the bytes for
1.85× the cost really does mean bytes are not what dominates at these bitrates. What was
wrong was the leap from "not bytes" to "therefore stream lifecycle", when "per access
unit, wherever that work lives" was the honest statement of the same data.

That is the second time in a week I have published a number that was real and an
attribution that was not — the other was gating our build on sizes.<plat>.json's
uimage_bytes (1,917,812) when the uImage in the nor-lite tarball is 1,883,172. That one
nearly bought a fleet-wide repartition of boards whose recovery path I know to be
unreliable. Same shape: a proxy that nearly works produces a confident wrong answer
instead of an obvious failure.

The harness stays in the tree behind ?mux=1, off by default, so this is re-runnable in
minutes against any future claim about per-stream cost — including on other SoCs, if that
is ever useful to you.

Thank you for pushing on it rather than taking the table at face value. This is the most
useful correction I have had in a while, and it saved a rewrite.

Instrumented the controller as you suggested. It corrects what I told you: the
oscillation I reported was not the application's shadow of Cubic — Cubic was doing
nothing at all. And it turned out to be a real bug in our QoS, now fixed.

What I added

wtransport exposes the inner quinn connection behind a quinn feature, so
Connection::quic_connection().stats().path gives real PathStats: cwnd, smoothed
rtt, congestion_events, lost_packets, current_mtu, black_holes_detected.
Sampled once a second, logged as per-interval deltas — cumulative totals integrate
away exactly the back-off/recovery behaviour worth seeing.

Steady state: Cubic never leaves its initial window

Cloud publish, H.265 640×360@12 + AAC, ~74 kbit/s of media, ~44 pkt/s:

rtt=27-35ms  cwnd=12000  mtu=1452  lost=0  loss=0.00%  cong_events=0  blackholes=0

cwnd is pinned at 12000 = 10 × 1200 = quinn's initial window, and never moves in
either direction. At this bitrate the connection is application-limited, never
cwnd-limited, so Cubic has no reason to grow the window and no loss to shrink it.

That is worth stating plainly, because it undercuts the premise of my own offer:
in normal operation on this hardware, the congestion controller is untested. Any
claim about Cubic's behaviour here — mine included — is currently unsupported.

The saturation run, and the bug it exposed

I then saturated the radio (three cameras streaming 1080p on one 40 MHz channel) so our
QoS would fire, expecting to catch the controller reacting:

QoS: uplink pressure -> video0 600kbps
  ... 10 consecutive samples, ALL: cwnd=12000 lost=0 cong_events=0, rtt flat 28-35ms
QoS: pressure cleared -> 1200kbps   23:46:20.641
QoS: uplink pressure -> 600kbps     23:46:21.651   <- one second later

The path was completely healthy the entire time. No loss, no congestion event, no
RTT inflation, cwnd untouched.

The cause was ours. That pressure signal fired on sem.available_permits() == 0 — our
sender's own 8-permit concurrency semaphore — not on anything the network reported. So
we were halving the local video bitrate because our own bounded window filled, most
plausibly from CPU contention holding permits longer while the agent was at ~23% of the
single core. I have not proven that causation; what is proven is that the link was not
congested when the signal fired.

So my "my hysteresis is too optimistic" was wrong twice over: the hysteresis was fine,
and it was watching a signal with no relationship to congestion.

Fixed

Pressure now comes only from evidence the transport actually reports — any of:

congestion_events increased (Cubic itself reacted)
packets lost in the interval
smoothed RTT > 2× the minimum observed RTT — queue building ahead of us, which on
a buffered link shows up before loss. min-RTT is learned from observation rather than
configured, so it adapts to whatever path a given camera has.

Re-ran the identical saturation test after the change: zero pressure events, zero
throttling, RTT 26.5–30.5 ms against a tracked minimum of 26.0. The permit-count
trigger is deleted, with a comment explaining what the trace showed so nobody restores it.

What this means for the question you actually asked

I still cannot tell you what Cubic does on a bad link, and now I can show why rather
than assert it: we never stress the controller, so there is nothing to observe. The
contended-WiFi stand-in does not stress QUIC at all — it stresses our CPU.

That needs a genuinely degraded path, which means your suggestion: netem on a Linux box
acting as AP or bridge, since tc is absent by construction on these images. The
instrumentation is in place and ships dark behind a flag file, so when I have that
bridge the traces come for free — cwnd, RTT, loss and congestion events per second
through a controlled loss/delay/queue profile, and against a real modem after that.

Two smaller things from the same trace, in case they are useful: PLPMTUD settles at
1452 on this path, and black_holes_detected stayed 0 across every run.

Got a shaper in front of a camera and ran the traces. Cubic numbers from this hardware
below, plus the bug the experiment found in my own code — which is the more useful half.

Setup, and its honest limits

No Linux bridge to hand, so I used what was: macOS dnctl/pfctl dummynet on the
receiving machine, shaping the camera's LAN WebTransport path. The camera is the
sender there, so its quinn/Cubic drives a link whose loss and delay I control.

Three caveats, all of which narrow what this proves:

It is the LAN path, not the cloud uplink. Same quinn, same Cubic, same silicon —
different connection.
dummynet loss is uniform-random. Real wireless and cellular loss is bursty, and
loss-based controllers behave differently under bursts.
Still not a cellular modem. Nothing here reproduces RAN scheduling or handover.

Instrumentation is quinn's own PathStats sampled every 2 s, counters as per-interval
deltas.

Baseline, and why the cloud path told us nothing

	cloud publish (~74 kbit/s)	LAN viewer (~1.5–2 Mbit/s), unshaped
cwnd	12,000, never moves	80,564 → 237,380
RTT	27–35 ms	10.8–40.8 ms
congestion events	0	0 (in a clean 68 s run)

The cloud figure is the one I reported before: cwnd pinned at quinn's initial window
because at 74 kbit/s we are application-limited and never fill it. On the LAN path the
same stack grows to 237,380 and parks — still no loss, but now demonstrably cwnd-active.

2% loss + 50 ms

cwnd   min 6,339   max 22,768   mean 13,616      (vs ~237,000 unshaped)
rtt    63.4 – 96.9 ms, mean 76.4
loss   34/34 intervals, mean 2.59%, peak 6.20%
cong   166 events over 68 s  (~2.4/s)

cwnd collapses ~17× and stays in a continuous sawtooth: 20,776 → 17,347 → 9,334 →
9,614 → 13,011 → 15,591 → 14,269 → 9,825 → 12,179 → 15,245. Multiplicative decrease on
loss, cubic regrowth, repeatedly, with no sign of collapse to a floor or of oscillating
pathologically. Measured loss runs above the configured 2% because retransmissions are
dropped at the same rate.

Application-level outcome: the viewer still received 1,236 video frames in 70 s (~17.6
fps against 25 nominal) with 5 stalls >500 ms. Degraded, session intact.

200 ms delay, `plr 0` — the bufferbloat case

I expected Cubic to ignore the latency and fill the buffer. It half did:

cwnd   min 26,974  max 85,896  mean 49,923
rtt    217.5 – 264.2 ms, mean 231.5
loss   11/34 intervals, mean 0.61%   <-- with plr 0
cong   17 events

cwnd settled ~3.7× higher than the lossy case but well below the unshaped 237,000 — and
not because it saw the delay. With plr 0 there was still 0.61% loss, because the
pipe's finite queue overflows once 200 ms of BDP is in flight. Every reaction traces to
that loss. Cubic did not respond to latency; it responded to the drops that latency
eventually caused.

Which is the expected result for a loss-based controller, but worth having measured: on
a path that buffers deeply enough to delay without dropping, this stack has no signal at
all until the buffer finally overflows. That is the argument for delay-sensitivity on
cellular, made with numbers rather than assertion.

The bug this found in my own code

Worth more than the traces. I had added a bufferbloat trigger to our QoS: raise pressure
when smoothed RTT exceeds 2× the minimum observed. The minimum was all-time,
per-connection.

A session that connects through an already-bloated path learns its minimum from that
path:

lan-cc rtt=249.3ms min=249.3ms    <- connected through 200ms of shaping
lan-cc rtt=226.1ms min=219.2ms    <- ratio 1.03x; the 2x test can never fire

So the trigger was dead precisely on the paths it existed for — a camera on a
permanently bloated uplink would report nothing wrong, forever. Fixed by making the
minimum a 300 s rolling window (gated on ≥30 samples, so a session's first seconds
do not make every path look clean against itself). Same connection, shaper removed
mid-session:

lan-cc id=5 rtt=226.1ms min=219.2ms   <- before
        ── shaping removed, no reconnect ──
lan-cc id=5 rtt=16.5ms  min=16.5ms    <- re-baselined

Threshold goes from a meaningless 438 ms back to 33 ms.

The residual limit cannot be fixed this way and I am not going to pretend otherwise:
a path bloated for the entire window is indistinguishable from a genuinely distant relay
using RTT alone. This detects bloat that develops, not bloat that was always there. I
considered an absolute ceiling and rejected it — 250 ms is pathological on my LAN and
normal on satellite, so a fixed threshold relocates the wrong answer rather than removing
it. If there is a standard trick for this I would rather hear it than invent one.

Also worth stating: across every run here, shaped or not, every congestion event came
from loss. The RTT arm of that trigger is correctly baselined now but has still never
fired in anger on a real path.

What would make this actually answer your question

A real modem. What I have says "here is Cubic on this silicon under synthetic uniform
loss on a LAN". Cellular adds burst loss, RAN scheduling delay and handovers, and I would
not extrapolate from these numbers to that. If a cellular-attached board is something
OpenIPC has access to, the instrumentation is a flag file away from producing the same
traces there, and I am happy to run whatever profile is useful.

Closing: the question of whether WebRTC belongs in lite was answered (yes, by design), with measured costs. The QUIC/WebTransport work is your own project. Thanks for publishing the data.

majestic lite: is WebRTC intended for 8 MB NOR devices? (+37% since April) #2344

Description

Is WebRTC intended for the lite majestic variant on 8 MB NOR?

Activity

widgetii commented on Aug 31, 2026

bneigher commented on Aug 31, 2026

bneigher commented on Aug 31, 2026

The part I actually want to ask about

widgetii commented on Aug 31, 2026

The thing you should delete is not WebRTC

Why the protocol set is what it is

For the shrinking hobby specifically

On WebTransport and QUIC

widgetii commented on Aug 31, 2026

WebRTC costs 82,376 bytes of binary, and 44 KB of flash

What that changes

One correction to your figures

bneigher commented on Aug 31, 2026

1. CPU: what fraction of the core is transport

2. Memory

3. Handshake

4. Crypto without AES acceleration

5. Behaviour when the link degrades — two caveats first

Summary

widgetii commented on Sep 1, 2026

The per-stream result needs one more experiment before you redesign

You can have the 2.3× today, without anyone else changing anything

It will not transfer to SRTP, and the reason is worth knowing

Memory: reproduced on our bench, with one correction that matters

On tc: confirmed absent, and here is the way around it

The benchmark: yes, and it is worth more than one row

One finding of yours that is about us, not you

bneigher commented on Sep 1, 2026

The missing input: per-row AU rates

Your experiment is the right one and I have not run it yet

The ChaCha20 change does not work on QUIC, and the reason is structural

Accepted without reservation

The benchmark, and ipctool

On majestic's 2.4 points

bneigher commented on Sep 1, 2026

Method

Result

Two honest caveats about arm B

The AU rates, since they were the missing input

What I take from this

bneigher commented on Sep 2, 2026

What I added

Steady state: Cubic never leaves its initial window

The saturation run, and the bug it exposed

Fixed

What this means for the question you actually asked

bneigher commented on Sep 4, 2026

Setup, and its honest limits

Baseline, and why the cloud path told us nothing

2% loss + 50 ms

200 ms delay, plr 0 — the bufferbloat case

The bug this found in my own code

What would make this actually answer your question

widgetii commented on Oct 4, 2026

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Is WebRTC intended for the `lite` majestic variant on 8 MB NOR?

On `tc`: confirmed absent, and here is the way around it

200 ms delay, `plr 0` — the bufferbloat case