W3C home > Mailing lists > Public > public-webrtc@w3.org > July 2013

RE: setting bandwidth

From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
Date: Mon, 29 Jul 2013 22:15:46 +1000
Message-ID: <CAHp8n2=kus3SV60DRcnY4CFT-g2jbb_0ROh8FdkovTzax70vLA@mail.gmail.com>
To: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
Cc: public-webrtc <public-webrtc@w3.org>, Harald Tveit Alvestrand <harald@alvestrand.no>
Also I wonder if it is a chrome bug or intended per the spec that a webrtc
video stream in Chrome is limited to 2Mbps no matter how much bandwidth is
available? After this discussion it seems more like a Chrome bug...?

Silvia.
 On 29 Jul 2013 19:05, "Charles Eckel (eckelcu)" <eckelcu@cisco.com> wrote:

>  I was just catching up on this thread, and I found it hard to follow
> because I think a few different things are being mixed together.****
>
> 1) negotiation of bandwidth****
>
> This is the result of some API/signaling interaction, not the real-time
> performance of the network. Failure to negotiate some minimum required
> bandwidth could result in an application wanting to consider the
> negotiation as failed (e.g. failure to add a media stream)****
>
> 2) current bandwidth utilization****
>
> This is something that seems valuable to be able to query through the
> stats API.****
>
> 3) packet loss/delay****
>
> Packet loss or delay beyond some thresholds is what is going to result in
> poor video quality. This is something for which a callback to the app seems
> appropriate.****
>
> With video, the bandwidth is high when there is a scene change or lots of
> motion. The bandwidth may drop considerably when the image is fairly
> static. Having a callback when the bandwidth falls below some minimum does
> not seem very useful. What is more useful is a callback when specific
> packet loss or delay thresholds are crossed.****
>
> ** **
>
> Cheers,****
>
> Charles****
>
> ** **
>
> *From:* Harald Alvestrand [mailto:harald@alvestrand.no]
> *Sent:* Monday, July 29, 2013 6:16 AM
> *To:* public-webrtc@w3.org
> *Subject:* Re: setting bandwidth****
>
> ** **
>
> On 07/29/2013 12:58 AM, cowwoc wrote:****
>
>  On 28/07/2013 5:06 PM, Harald Alvestrand wrote:****
>
>  On 07/28/2013 10:43 PM, cowwoc wrote:****
>
>
>     Look, when I try to upload a very large file over TCP I see the upload
> speed peg at close to 100% of capacity seemingly immediately. Can't we take
> the same mechanism as TCP and layer it on top of what WebRTC uses today?
> I'd like to avoid drilling down into the specific implementation at this
> time. All, I'm asking is whether this is technically doable and if not, why.
>
>     Let's not lose track of the goal of this discussion, which is to
> enable users to specify the initial/minimum bandwidth usage so chat
> sessions don't start out with blurry/choppy video at 1080p. If you have an
> alternative for achieving this, I'm all ears.****
>
>
> That may be YOUR goal in this discussion. It is certainly not a
> formulation I'll sign up for as a WG goal.
>
> In the IETF, the first goal is the continued survival of the Internet; all
> other desires are subordinate to that.
>
> We know from bitter experience (google "congestion collapse") that badly
> designed congestion control, when deployed simultaneously by a large
> fraction of the nodes on the Internet, brings problems that can render the
> Net unusable, and which - mark this - do NOT show up in tests done with
> only a few users.
> The IETF congestion people are conservative, and have good reason to be.
>
> I think we can agree that there is a clear desire to start up chat
> sessions with video that is pleasing to the user.
>
> I don't think we have any kind of consensus that letting the browser obey
> any suggestion for initial bandwidth that comes from the user, no matter
> what it knows (or knows it doesn't know) about the network conditions
> between the sender and the recipient, is a reasonable way to achieve that.
>
> Harald****
>
>
> Harald,
>
>      The only reason I brought up the example of HTTP upload or WebSockets
> is to ask what we can learn anything from how other technologies handle
> this problem. I'm only interested in the "what" (instant-on clear/smooth
> video chat sessions) not the "how". To that end, do you have any ideas?***
> *
>
>
> My best idea at the moment is to start the video without showing it,
> waiting for a few tenths of a second to let the congestion manager ramp up,
> and then show it.
>
> That depends on having a bandwidth manager that can ramp up in a few RTTs
> rather than multiple seconds, of course, but the relevant WG is the IETF
> RMCAT WG rather than this one.
>
> This group needs to focus on how it exposes controls for things that can
> live within the constraints set by the environment it has to live in.
>
> ****
>
Received on Monday, 29 July 2013 12:16:13 UTC

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