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

[Henrik Lundin <[e-mail]>]

On Mon, Oct 3, 2011 at 12:55 PM, Ingemar Johansson S <
[e-mail]> wrote:

> **
> Hi
>
> I guess one thing that should be considered is that a mobile user may make
> handover between different radio access types with large differences in
> terms of e.g throughput. You may have scenarios such as handover from LTE to
> GPRS or perhaps you have handover from WiFi to HSPA if a user walks away
> from a café. I would expect that these scenarios will become very common in
> the future. It is also worth mention that handover between  WiFi and 3GPP is
> standardized in 3GPP, which in practice means that this will ultimately
> happen without end user interaction. What the endpoints will notice is
> potentially a large and sudden change in throughput.
>
> In cases like these thoughput may drop rapidly. Of course you can sense
> this with the outlined algoritms that signals feedback only when needed but
> that assumes that you receive packets that you can infer enough statistics
> on. Problem is that on the receiver side don't really know that you are
> about to receive a packet.
> With ACK-clocked algorithms like TCP and TFWC the sender simply stops
> sending packets when ACK's are not received anymore. Receiver based algos
> are a bit more complicated as the risk is higher that the sender will
> continue to send packets for some time even though the channel throughput
> has dropped considerably, resulting in excessive congestion somewhere along
> the path.
>
> Is this a problem ?. I don't know, I guess time will tell.
>
I guess it is a problem.


>
> /Ingemar
>
>  ------------------------------
> *From:* Henrik Lundin [mailto:[e-mail]]
> *Sent:* den 3 oktober 2011 10:13
> *To:* Jim Gettys
> *Cc:* Randell Jesup; [e-mail]
> *Subject:* Re: [rtcweb] An input for discussing congestion control (Fwd:
> New Version Notification for draft-alvestrand-rtcweb-congestion-00.txt)
>
> Sorry for my late response. (I've been away for some time.) I'd just like
> to add my two cents to the discussion on feedback latency.
>
> Frequent and rapid feedback is important to ensure stability of the CC; I
> think we all agree. However, with an algorithm similar to the suggested one,
> having a receive-side processing and analysis function, the key feature is
> that feedback _can_ be sent quickly when needed. When everything is ok,
> feedback can be quite sparse, as long as the rate increments are somewhat
> adjusted for the system response time (including the feedback latency). When
> congestion is detected by the receiver, it is important that a feedback
> message can be sent more or less immediately (say, within 100 ms). However,
> I do not see the need for constant feedback every RTT or so.
>
> /Henrik
>
>
>
>
>
> On Wed, Sep 21, 2011 at 3:43 PM, Jim Gettys <[e-mail]> wrote:
>
>>
>> On 09/21/2011 09:28 AM, Randell Jesup wrote:
>>
>> On 9/21/2011 4:23 AM, Harald Alvestrand wrote:
>>
>> I think receiver->sender reporting every RTT (or every packet, which is
>> frequently less frequent) is overkill, but that's a statement with a lot of
>> gut feeling and very few numbers behind it.
>>
>> One advantage we have in RTCWEB is that we can assume that if audio and
>> video work OK across the network, we're in a good place. We don't have to
>> worry about getting gigabyte file transfers to utilize 90% of the link -
>> even thogh we have to worry about audio and video functioning while those
>> gigabyte transfers are taking place.
>>
>>
>> Agreed.  Also, in practice the TCP flows we're competing with are rarely
>> long-lived
>> high-bandwidth flows like GB file transfers.  Normally they're flurries of
>> short-lived TCP
>> (which is important to consider since these short-lived flows can suddenly
>> cause buffering
>> without warning).
>>
>>
>> You get to deal with some of each. Both cause havoc in the face of
>> bufferbloat.  The long lived flows keep the buffers in your OS/Home
>> router/broadband gear near full, inserting lots of delay.  This includes
>> doing backups (local or remote), uploading videos, downloading videos to
>> disk, bittorrent, etc. The netalyzr data is quite grim, particularly when
>> you realise it's a lower bound on the problem (the netalyzr test is
>> sensitive to cross traffic and more importantly, tops out by the time it
>> gets to 20Mbps).
>>
>>
>> http://gettys.wordpress.com/2010/12/06/whose-house-is-of-glasse-must-not-throw-stones-at-another/
>>
>> As far as the transient bufferbloat problem caused by web traffic, and why
>> IW10 is a problem in my view at this time, see:
>>
>> http://www.ietf.org/id/draft-gettys-iw10-considered-harmful-00.txt
>>
>>
>>
>> As for 1 feedback/RTT, I agree.  And if you wanted to use one
>> feedback/RTT, I'd put the feedback in
>> a TCP header extension or design an RTP equivalent that can carry it in
>> the reverse-direction
>> media flow (when available).  But that's a different argument.
>>
>> I like timestamps, if only to make it easy to tell the user: you are
>> losing, and it's because of your broken network.
>>
>> For TCP, the TCP timestamp option is on by default in Linux, and I think
>> may be on by default on other modern systems (anyone have any data?).
>>                         - Jim
>>
>>
>> Other protocols may not be so nice.
>>
>> _______________________________________________
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>>
>>
>
>
> --
> Henrik Lundin | WebRTC Software Eng | [e-mail] | [phone]
> 41 <%2B46%2070%20646%2013%2041>
>
>
>


-- 
Henrik Lundin | WebRTC Software Eng | [e-mail] | [phone]