[R-C] Proposal comments (Re: RTCWEB Congestion Control Standardization)

[Varun Singh]

Hi,

Comments inline

On Fri, Oct 28, 2011 at 00:04, Harald Alvestrand <harald at alvestrand.no> wrote:
> Commenting on Randell's proposal, not on the alternatives, so changing
> subject....
>
> On 10/26/2011 01:57 PM, Randell Jesup wrote:
>>
>> I vote for 2, and provide a sample implementation and let people innovate.
>>  I would also support trying to standardize the sample implementation under
>> 3 in parallel, with the understanding it will take
>> A Long Time.
>>

One thing about 1. (TFRC) is that it is fair, but has rate
oscillations that degrade call quality etc. While I agree 1 is
sub-optimal, it depends on what we mean by fairness and how much do we
emphasize it. Else I agree with the process of starting on 2 now and 3
in parallel for the long term.

>> Here's a set of proposed stab at requirements for rtcweb implementations:
>>
>> As part of rtcweb, congestion control must be addressed:
>>
>>    1.  All WebRTC media and data streams MUST be
>>        congestion-controlled.
>>
>>    2.  The congestion algorithms used MUST cause WebRTC streams
>>            to act fairly with TCP and other congestion-controlled
>>            flows, such as DCCP and TFRC, and other WebRTC flows.  Note
>>            that WebRTC involves multiple data flows which "normally"
>>            would be separately congestion-controlled.
>
> I'd use "reasonably fairly" to reduce (slightly) the chances of getting lost
> in the definition of "fair" and the number of angels who can dance on the
> head of a pin.

I agree with Harald on this.

>>
>>    3.  In order to support better overall user experiences and to
>>        allow applications to have better interaction with
>>        congestion control, a new AVPF feedback message [ insert
>
> Suggest moving the part before the comma to an intro paragraph. It's an
> overall goal.
>>
>>        name here] shall be defined to allow reporting of total
>>        predicted bandwidth for receiving data, as opposed to
>
> "reporting of a recipient's estimate of available bandwidth for receiving
> data"

using data is a bit confusing because data else where in the text
means data channel.

"reporting of a recipient's estimate of available bandwidth for
receiving the combined media and data streams"


Just to confirm:
the receiver calculates the rate per stream and then adds them up:
CT_combined = R_audio  + R_video + R_data
and the audio and video channels calculate the rate as described in
the proposal.

The sender uses the receiver CT_combined and the current sending rate
of each channel to re-allocate the distribution. Is the aim to try and
get the distribution similar to what the receiver envisioned or is the
sender free to do whatever?

>>
>>        TMMBR, which requests a sending rate for a single SSRC
>>        flow.  [ This is roughly equivalent to b=CT:xxx ]
>>
>>        We may want to give the estimation algorithm the option to
>>        not include or exclude the data-channel bandwidth, but it
>>        SHOULD include that.
>>

Is the data sent using the same congestion control mechanism?

>>    4.  In order to facilitate better operation of
>>        bandwidth-estimation algorithms on the receiving side, the
>>        sending side MAY include a transmit-time RTP header
>>        extension (TBD) to some or all media streams. Note that
>>        this will add about 12 bytes to each RTP packet.
>
> 8 bytes in the proposal I'm editing in another window....

I agree with the MAY as it is yet to be proven useful.

>>
>>        An optimization may be to only include these timestamps if
>>        they deviate by more than [ some amount TBD from the
>>        running average and from the number of bytes preceding it
>>            with the same timestamp ].  This is based on the fact that
>>            for many devices, the sample->send interval is fairly
>>        consistent at the levels of accuracy needed here, and so
>>        significant bandwidth savings can be made.
>
> would make more sense to me to only include it if it varied more than N
> microseconds from (the time specified by the RTP timestamp for the frame + a
> constant delay). Suggest omitting this for what you propose to the larger
> group.
>>
>>    5.  The receiver SHOULD attempt to minimize the number of
>>        bandwidth reports when there is little or no change, while
>>        reporting quickly when there is a significant change.
>>

Will we propose an algorithm or lower or upper-bound for this? like:
not quicker than once per RTT even if the 5% rule allows it. Or send
an early report when loss, inter packet delay, etc exceeds a given
threshold.

>>    6.  Congestion control MUST work even if there are no media
>>        channels, or if the media channels are inactive in one or
>>        both directions.
>
> What does "work" mean if there is no data?

MUST be enabled?

>>
>>    7.  The congestion control algorithm SHOULD attempt to keep the
>>            total bandwidth controlled so as to minimize the media-
>>            stream end-to-end delays between the participants.
>
> Not sure I understand this. If I understand it, suggest to rewrite as
>
>   7. The congestion control algorithm SHOULD attempt to minimize the
> media-stream
>       end-to-end delays between the participants, by controlling bandwidth
> appropriately.

The receiver doesn't know the end-to-end delay, the RTT is calculated
at the sender. So is the sender making this decision or the receiver?

>>
>>    8.  When receiving a [ insert new AVPF message here ], the
>>        sender shall attempt to comply with the overall bandwidth
>>        requirements by adjusting parameters it can control, such
>>        as codec bitrates and modes, and how much data is sent on
>>        the data channels.
>
> Suggest "shall" -> "may".

Does the application know apriori how much data would be sent on the
data channel? because it is likely that the sender would want to use
all of the signaled bandwidth for audio and video. (Especially if the
data channel is used for IM, which may be used sporadically).

>>
>> Not part of our IETF requirements, at the JS level bandwidth changes
>> should be reported to the application along so that it has the option to
>> make changes that we can't make automatically, such as removing or adding a
>> stream, or controlling the parameters of a stream (frame rate, etc).  Note
>> that if the application doesn't do anything, the automatic adaptation will
>> still occur.
>>
> There may be adjustments that need communicating with the other end too
> (renegotiation). It's reasonable to assume that these are only performed
> when requested through the JS layer.
>
>
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