Re: [rtcweb] An input for discussing congestion control (Fwd: New Version Notification for draft-alvestrand-rtcweb-congestion-00.txt)

Soo-Hyun Choi <soohyun.choi@cl.cam.ac.uk> Tue, 20 September 2011 12:34 UTC

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From: Soo-Hyun Choi <soohyun.choi@cl.cam.ac.uk>
Date: Tue, 20 Sep 2011 21:36:24 +0900
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To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] An input for discussing congestion control (Fwd: New Version Notification for draft-alvestrand-rtcweb-congestion-00.txt)
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Harald,

On Tue, Sep 20, 2011 at 20:47, Harald Alvestrand <harald@alvestrand.no> wrote:
>
> let's avoid unquantifiable terms in our discussions. "A little more careful"
> is an unquantifiable term.
>
> If we want to say "I think the algorithm must guarantee that the bandwidth
> estimate is updated within 1.5 seconds when the available bandwidth changes
> from 2 Mbit/sec to 1 Mbit/sec", let's say something like that - focus on
> what the requirements are.
>
> We can always be a little more careful. But it doesn't tell us when we're
> close enough.
>

Sorry if my previous email looked that way to you (of course, I didn't
mean to just pray for the mere hope). I was talking about some general
CC principles in the high level for real-time interactive multimedia
applications, like some of us did too in this thread.

One of the critical metrics of a delay-based CC algo is an accurate
measurement of inter-packet interval (IPI) and timely report of IPI to
the sender so it can update the send rate correctly, which we all
agreed in this email thread. Although everyone in this list seemed to
aware of this, I have highlighted again that the relatively
course-grained RTCP carrying the IPI measurement data can cause
instability at the encoder's output bitrate due to the following
reason.


> I.e., Inaccurate BW estimation could lead into providing inaccurate
> information about the available BW to the upper layer, which
> eventually result in instability at the codec's encoding rate.


So, my original question/suggestion was why not use AVPF (which is
smaller packet than the standard RTCP) to generate/report more
frequently to the sender in your algorithm. Or alternatively, I have
already designed a window-version's of TFRC that re-introduces a
TCP-like Ack-clocking, so-called TFWC (TCP-Friendly Window-based CC),
which performs reasonably well (as good as TFRC) for the similar
situations (RTC environment).

I would like to ask to this list (or Harald) if suggesting alternative
CC algo idea could fit into the RTCWEB Charter's scope - if so, I
could elaborate more by writing an I-D or so. In the mean time, you
could check my document at http://tfwc.hackerslab.eu/ if you are
interested.

Kind regards,
Soo-Hyun