[Cake] mo bettah open source multi-party videoconferncing in an age of bloated uplinks?

Fri Mar 27 16:32:23 EDT 2020

I don't know to what extent the freeswitch guys would be interested in
this thread. I'd like find a good list or forum to talk about the
state of the art in videoconferencing ? , the ietf rmcat and webrtc
lists are mostly dead. hangouts, jitsi, zoom, etc, seem to be pretty
good products
nowadays (at least in my fq_codel'd environment), but solid info on
how to make them better in the home and for online tele-learning

On Fri, Mar 27, 2020 at 12:00 PM David P. Reed <dpreed at deepplum.com> wrote:
>
> Congestion control for real-time video is quite different than for streaming. Streaming really is dealt with by a big enough (multi-second) buffering, and can in principle work great over TCP (if debloated).

Your encoder still has to adjust to the available bandwidth. The
facebook streaming application did this beautifully through my very
limited highly shared 5mbit uplink - adjusting quickly to a parallel
rrul test in particular by skipping some frames. then lowering the
frame rate and quality, but an early attempt of mine to merely reflect
rtmp streams did not, neither an attempt with "obs studio".

there was about 30 sec of delay in the facebook test - I figure some
of this is tuned to visible uplink buffer sizes (still seconds over
cell), but also to give the riaa a shot at censoring the audio. (a
commercial song crept into - over a mic! - which was detected as
infringing on one attempt which automatically muted the audio and
keyed a nastygram from fb)

I'm going to poke into obs studios underlying code (rtsp anyone?0 at
some point, and really - udp with a head dropping aqm is the best
thing for transporting video, IMHO.

> UDP congestion control MUST be end-to-end and done in the application layer, which is usually outside the OS kernel. This makes it tricky, because you end up with latency variation due to eh OS's process scheduler that is on the order of magnitude of the real-time requirements for air-to-air or light-to-light response (meaning the physical transition from sound or picture to and from the transducer).

We are so far from that point! encoder latencies today are in the
100+ms range. I always liked the opus codec because it can get down to
2.7ms encoding latencies, and a doubled frame rate camera 8ms.... but
video encoding rates Im out of date on. (?)

One long deferred piece of webrtc/rmcat research I always meant to do
was audio and video on separate ports in the stream,
and using that 2.7m opus clock and depending on fq at the bottleneck
to provide better congestion control information by treating the
smaller audio packets as a clock signal. Due to lack of port space and
a widespread perception that fq isn't out there, most
videoconferencing streams multiplex everything over the same port.
With ipv6 in place, well, port space is no longer a problem.

>
> This creates a godawful mess when trying to do an app. Whether in WebRTC (peer to peer UDP) or in a Linux userspace app, the scheduler has huge variance in delay.

I figure the bounding scheduler latency is still well manageable below
a single 60fps frame.

> Now getting rid of bloat currently requires TCP to respond to congestion signalling. UDP in the kernel doesn't do that, and it doesn't tell userspace much either (you can try to detect packet drops in userspace, but coding that up is quite hard because the schdulers get in the way of measurement, and forget about ECN being seen in userspace)

ECN in userspace is easy on udp, except that most api's tend to
abstract into a file handle style abstraction and a single return of
data, not control information, and the api for getting tos options
ugly. APIs that can return data and info (data, packetheader) =
getudp_someway() probably exist for more modern languages like go, but
rarely c or c++. Totally out of date on this, last I looked at the
google congestion congtrol code bae was in mozilla... 8 years ago!

As for doing udp semi-efficiently in batches...

sendmmsg, recvmmsg is a rather underused kernel api. And ugly as sin.
With some major limitations.

>
> This is OS architecture messiness, not a layer 2 or 3 issue.

To me the nightmare starts with most cpu context switch latencies
being 1000s of clocks nowadays.

>
> I've thought about this a lot. Here's my thoughts:
>
> I hate putting things in the kernel! It's insecure. But what this says is that for very historical and stupid reasons (related to the ideas of early timesharing systems like Unix and Multics) folks try to make real-time algorithms look like ordinary "processes" whose notion of controlling temporal behavior is abstracted away.

On the whole, with the rise of quic - in particular quic, as multiple
userspace libs have been emerging - we've got good bases to move
forward with more stuff in userspace.

>
> So:
> 1. We really should rethink how timing-sensitive algorithms are expressed, and it isn't gonna be good to base them on semaphores and threads that run at random rates. That means a very different OS conceptual framework. Can this share with, say, the Linux we know and love - yes, the hardware can be shared. One should be able to dedicate virtual processors that are not running Linux processes, but instead another computational model (dataflow?).

Linux switched to an EDF model for networking in 5.0

> An example of this (though clunky and unsupported by good tools) is in FreeBSD, it's called *netgraph*. It's a structured way to write reactive algorithms that are demand or arrival driven. It also has some security issues, and since it is heavily based on passing mbufs around it's really quirky. But I have found it useful for the kind of things that need to get done in teleconferencing voice and video.

Neat.

>
> 2. EBPF is interesting, because it is more secure, and is again focused on running code at kernel level, event-driven.  I think it would be a seriously difficult lift to get it to the point where one could program the networked media processing in BPF.

But there is huge demand for it, so people are writing way more in it
than i ever ever thought possible... or desirable.

>
> 3. One of the nice things about KVM (hardware virtualization) is that potentially it lets different low level machine models share a common machine. It occurs to me that using VIRTIO network devices and some kind of VIRTIO media processing devices, that a KVM virtual machine could be hooked up to the packet-level networking drivers in the end device, isolating the teleconferencing from the rest of the endpoint OS, and creating the right kind of near-bare--metal environment for managing the timing of network packets and the paths to the screen and audio that would be simple and clean and tightly scheduled. KVM could "own" one or more of the physical cores during the teleconference.

see also sch_etx and  van's teaching nics about time -
https://netdevconf.info/0x12/news.html?keynote-recording-is-up there
has been a lot of progress on this front in the past few years -
having applications say when they want a packet to emerge -

but the offload discussion over on the linux list I referenced has
seemingly missed this idea entirely.

>
> You can see, though, that this isn't just a "network protocol design" problem. This is only partly a network protocol issue, but one that is coupled with the architecture of the end systems.
>
> I reminisce a little bit thinking back to the 1970's and 80's when TCP/IP and UDP/IP were being designed. Sadly, it was one of the big problems of communicating between the OS community and the protocol community that the OS community couldn't think outside the "timesharing" system box, and the protocol community thought of networking like phone calls (sessions). This is where the need for control of timing and buffering got lost. The timesharing folks largely thought of networks as for reliable timeless sequential "streams" of data that had no particular urgency. The network protocol folks were focused on ARQ.
> Only a few of us cared about end-to-end latency bounds (where ends meant keyboard click or audio sample to screen display change or speaker motion).

I recently got a usb keyboard with truly annoying latencies in it.
https://danluu.com/ 's work makes me feel better about people
perpetually ignoring this. An Apple II won the benchmark...

Last year I got a voice processor, with 70+ms usb latencies for audio
- useless for overdubs. same for all the usb based audio mixers made
today.

thunderbolt based  audio gear is hard to find and expensive, I have
been scrounging old firewire based stuff. (hey, if anyone has a RME
multiface card let me know off list)

> The packet speech guys did, but most networking guys wanted to toss them under the bus as annoying. And those of us doing distributed multinode algorithms did, but the remote login and FTP guys were skeptical that would ever matter.

Yep, we're annoying. But annoyed. And: I really think it would be a
less stressed and better communicating world if we got cell phone
audio
latencies, in partiular, back below 20ms.

> It's the latency, stupid. Not the reliability, nor the consistency, nor throughput. Unless both the OS and the path are focused on minimizing latency, a vast set of applications will suck. Unfortunately, both the OS and network communities are *stuck* in a world where latency is uncontrollable, and there are no tools for getting it better.

except ours! :)

>
>
> On Friday, March 27, 2020 1:27pm, "Dave Taht" <dave.taht at gmail.com> said:
>
> > sort of an outgrowth of this convo:
> >
> > https://lwn.net/SubscriberLink/815751/786d161d06a90f0e/
> >
> > I imagine worldwide videoconferencing quality could be much better if
> > we could convince more folk to
> > finally install sqm or upgrade to a working docsis 3.1 solution, etc.
> > Maybe some rag somewhere will finally pick up on bufferbloat solutions
> > and run with it? Or we can write some articles? Or reach out to school
> > systems? Or?
> >
> > I've been fiddling with jitsi, and am about to give freeswitch a try.
> > Last I looked freeswitch's otherwise pretty nifty conference bridge
> > didn't dynamically adjust at all due to e2e signalling, but that was
> > years ago. (?)
> >
> > I have to admit that p2p multiparty videoconferencing seems more
> > plausible in a de-bufferbloated age, but
> > haven't explored what tools are available. (?)
> >
> > There's also been this somewhat entertaining convo on the ietf mbone
> > list: https://mailarchive.ietf.org/arch/msg/mboned/2thFQk_IYn38XCZBQavhUmOd6tk/
> >
> > Around me there has been this huge interest in "streaming". The user
> > agreement for these (see restream.io's) is scary - and the copyright
> > police have control... but I am very happy to report that even a
> > couple really lousy long distance fq_codel'd ath9k links work *really*
> > well (with facebook's implementation), where a non fq_codeled link
> > (ath10k) failed miserably... and setting up a reflector in nginx also
> > failed miserably.
> >
> > Anyone working on the ath10k AQL backport for openwrt as yet?
> >
> > --
> > Make Music, Not War
> >
> > Dave Täht
> > CTO, TekLibre, LLC
> > http://www.teklibre.com
> > Tel: 1-831-435-0729
> > _______________________________________________
> > Cake mailing list
> > Cake at lists.bufferbloat.net
> > https://lists.bufferbloat.net/listinfo/cake
> >
>
>

-- 
Make Music, Not War

Dave Täht
CTO, TekLibre, LLC
http://www.teklibre.com
Tel: 1-831-435-0729