- From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
- Date: Mon, 29 Jul 2013 22:15:46 +1000
- To: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
- Cc: public-webrtc <public-webrtc@w3.org>, Harald Tveit Alvestrand <harald@alvestrand.no>
- Message-ID: <CAHp8n2=kus3SV60DRcnY4CFT-g2jbb_0ROh8FdkovTzax70vLA@mail.gmail.com>
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