[R-C] [Iccrg] draft-alvestrand-rtcweb-congestion-01 - A Google Congestion Control Algorithm for Real-Time Communication on the World Wide Web

[Wesley Eddy]

On 11/1/2011 10:34 AM, Lars Eggert wrote:
> This may be of interest to the RG:
> http://wiki.tools.ietf.org/html/draft-alvestrand-rtcweb-congestion-01
>


Here are my comments on this (also copied to the authors and the
mailing list for RTCWEB congestion control):

(1)
I think the model tries to correct for some effects later, but this
part only makes sense if all frames are the same size, and none of
the timestamps are subject to jitter being introduced by OS scheduling,
MAC layer (e.g. wireless) issues, etc:
"""
    We define the (relative) inter-arrival time, d(i) as

      d(i) = t(i)-t(i-1)-(T(i)-T(i-1))
"""

(2)

This part doesn't quite make sense to me, but I think it's just in
the wording and not in the math:
"""
    Since the time ts to send a frame of size L over a path with a
    capacity of C is

      ts = L/C


    we can model the inter-arrival time as

               L(i)-L(i-1)
      d(i) = -------------- + w(i) = dL(i)/C+w(i)
                   C
"""
That's not a model of the inter-arrival time, it's a model of the
difference between the delta timestamps and delta reception times
based on difference in packet length.


(3)
In the definition of w(i), do you also intend to account for other
sources of delays in the end systems' software?  Why do we think that
white noise is the right model?  This seems like a big assumption,
unless there's actually data that shows it holds under a very wide
range of settings.

(4)
It should probably be understood clearly that the i values are sample
numbers, but that the generation of samples comes from the behavior of
the plant, and they aren't taken at the filter/controller regularly.
This seems to be quite different than when the document says:
"""
    In this document we make the assumption that the rate control
    subsystem is executed periodically and that this period is constant.
"""

(5)
The algorithm at the receiver side is clueless about packet losses,
and seems like it would see them only when the loss period ends (when
a packet survives) and then would see them as really long inter-arrival
times rather than losses?  (or did I miss something really important?)


(6)
The rules about 2% and 10% in the sender-side algorithm seem ... odd.
Unless the report interval is sufficiently long in relation to the frame
rate, the percentages may not work well.  These parameters have to be
balanced, or the algorithm will be fragile.

(7)
Shouldn't the sender estimate of available bandwidth be able to go lower
than the TFRC equation output if the delay has been increasing (though
packets are not yet being lost)?

(8)
Startup is a big topic that isn't yet addressed.




-- 
Wes Eddy
MTI Systems