[Galene] Re: Congestion control and WebRTC [was: Logging]

["Toke Høiland-Jørgensen" <toke@toke.dk>]

Juliusz Chroboczek <jch@irif.fr> writes:

>> No, I don't think multiplexing more streams over the same five-tuple is
>> a good idea if it can be avoided. If the bottleneck does per-flow
>> queueing (like FQ-CoDel), you'd want each video flow to be scheduled
>> separately I think.
>
> Tere's a tradeoff here.  Using port multiplexing gives more information to
> middleboxes, but using SSID multiplexing reduces the amount of ICE
> negotation -- adding a new track to an already established flow requires
> zero packet exchanges after negotiation (you just start sending data with
> a fresh SSID), while adding a new flow for port multiplexing requires
> a new set of ICE probes, which might take a few seconds in the TURN case.

So in this instance a new flow happens when a new user joins and their
video flow has to be established to every peer?

>> Another couple of ideas for packet-level optimisations that may be worth
>> trying (both originally articulated by Dave Taht):
>
> Why is Dave not here?

I dunno; why aren't you here, Dave? :)

>> In the presence of an FQ-CoDel'ed bottleneck it may be better to put
>> audio and video on two separate 5-tuples: That would cause the audio
>> stream to be treated as a 'sparse flow' with queueing priority and fewer
>> drops when congested.
>
> Uh-huh.  I'll send you a patch to do that, in case you find the time to
> test it.

Sounds good, thanks!

>> (As an aside, is there a reference for the codec constraints in
>> browsers? And is it possible to tweak codec parameters, say to burn some
>> bandwidth to enable really high-fidelity audio for special use cases? Or
>> is Opus so good that it doesn't matter?)
>
> A typical laptop microphone has rather poor frequency response, so Opus at
> 48kbit/s is as good as the original.  It's just not worth reducing the
> audio rate upon congestion, it's the video rate that gets reduced.

Right, I see. Looking at the commit that introduced codec support, it
looks pretty straight-forward to crank up the bitrate; maybe I'll
experiment with that a bit (but not using my laptop's microphone).

> As to the video rate, you've got plenty of exciting knobs.
>
> 1. Congestion control.  As implemented in modern browsers, WebRTC uses two
> congestion controllers: a fairly traditional loss-based controller, and an
> interesting delay-based one.  This is described here:
>
>   https://tools.ietf.org/html/draft-ietf-rmcat-gcc-02
>
> Unlike some other video servers, that mere forward congestion indictions
> from the receivers to the sender, Galène terminates congestion control on
> both legs.  currently obeys congestion indication from the receivers, and
> implements the loss-based congestion controller for data received from the
> sender.
>
>   https://github.com/jech/galene/blob/master/rtpconn/rtpconn.go#L956
>
> We do not currently implement the delay-based controller, which causes
> collapse if the sender is on a low-rate bufferbloated network.  That is
> why Galène's client limits the rate to 700kbit/s by default (in the
> « Send » entry in the side menu).

Right, but the browsers do?

> Implementing the delay-based controller is number 1 on my wishlist.  Your
> help would be greatly appreciated.

Can't promise any hacking time, unfortunately, at least not short-term.
Happy to test out stuff, though :)

> 2. Sender-side tweaks.  The sender has a number of knobs they can tweak,
> notably maximum bitrate (separately for each track), and hints about
> whether to prefer framerate or image quality upon congestion.  The sender
> can also pick the webcam resolution, and they can request downscaling
> before encoding.

Ah, hence the "blackboard mode" - gotcha!

> 3. SVC.  The technology that excites me right now is scalable video coding
> (SVC), which I believe will make simulcast obsolete.  With VP8, the
> sender can request that some frames should not be used as reference for
> intra prediction; these « discardable » frames can be dropped by the
> server without causing corruption.  VP9 implements full scalability:
> temporal scalability, as in VP8, spatial scalability, where the codec
> generates a low resolution flow and a high resolution flow that uses the
> low resolution flow for intra prediction, and quality scalability, where
> the codec generates frames with varying quality.
>
>   https://en.wikipedia.org/wiki/Scalable_Video_Coding
>
> I'm currently planning to skip simulcasting, which I feel is an
> obsolescent technology, and experiment with SVC instead.  Implementing the
> delay-based controller is a higher prioerity, though.

Uh, hadn't heard about that before; neat!

>> Another packet-based optimisation that could be interesting to try out
>> is to mark packets containing video key frames as ECN-capable.
>
> Keyframes can be huge (120 packets is not unusual), it wouldn't be
> resonable to mark such a burst as ECN-capable without actually reacting to
> CE.  And if we drop part of the keyframe, we'll NACK the missing packets
> and recover 20ms + 1RTT later.

Hmm, right, okay I see what you mean...

>>> Ideally you'd also actually respond to CE markings,
>> RFC 6679.  I don't know if it's implemented in browsers.
>
> It is not.

Ah, too bad :(

-Toke