- From: Eric Rescorla <ekr@rtfm.com>
- Date: Wed, 8 Jan 2014 20:02:50 -0800
- To: cowwoc <cowwoc@bbs.darktech.org>
- Cc: "public-webrtc@w3.org" <public-webrtc@w3.org>
On Wed, Jan 8, 2014 at 7:52 PM, cowwoc <cowwoc@bbs.darktech.org> wrote: > Remind me again, what was wrong with this approach? It doesn't enable essentially any screen sharing scenario that people want. -Ekr > Enable screensharing without a flag/plugin. > Prompt the user for permission. > Allow screensharing for a single browser tab (can't capture the general > screen or foreign processes). > Prevent pages that use screensharing from issuing requests to foreign hosts > (i.e. Same Origin policy minus any exceptions). > > Lets start with something that is fairly restrictive (but doesn't require a > flag/plugin which kills traction), enable *some* use-cases, and built up > from there. > > Gili > > > On 08/01/2014 9:03 PM, Eric Rescorla wrote: > > On Wed, Jan 8, 2014 at 5:53 PM, piranna@gmail.com <piranna@gmail.com> wrote: > > I'm not ccomparing both in the way I accept whatever of both, but instead in > the way both (plugins and flags) are equally bad ideas. Screen and > application sharing should be included and enabled on browsers by default, > and not hidden behind a flag or whatever other method. > > For the reasons described in: > http://tools.ietf.org/html/draft-ietf-rtcweb-security-05#section-4.1.1 > > The browser vendors don't think this is that great an idea. > > If you think that screen sharing should be available by default, you > should perhaps suggest some security mechanisms which would > make the threats described here less severe. > > -Ekr > > Send from my Samsung Galaxy Note II > > El 09/01/2014 02:42, "Alex Gouaillard" <alex.gouaillard@temasys.com.sg> > escribió: > > @ piranha. > > while I agree with you for social users and most of the population out > there, the difference between clicking a flag and installing a plugin > is the process required by IT teams to accept the product and deploy > it in an enterprise environment. Everything needs to validated > beforehand, including (especially?) plugins. They have a very long > list of products to screen and maintain, and are very reluctant to add > yet another one. Moreover, google's chrome start with a higher > credibility than any small or medium sized company's plugin. > > On Thu, Jan 9, 2014 at 8:54 AM, Silvia Pfeiffer > <silviapfeiffer1@gmail.com> wrote: > > On Thu, Jan 9, 2014 at 10:10 AM, Randell Jesup <randell-ietf@jesup.org> > wrote: > > On 1/7/2014 8:50 PM, Alexandre GOUAILLARD wrote: > > here are a few proposition on things that are really biting us, and how > to > (perhaps) make it easier: > > - bandwidth control > 1. It seems that the number one sdp munging cause is the now infamous > B=AS: > line to put a cap on bandwidth. Since that capacity exists in the > underlying > code, it would be great to have an API that can help us put caps, > either on > each stream, and/or on the full call. > > > yes. > > > 2. I also see that there is a "auto-mute" feature being implemented > that > depend on an arbitrary threshold. It might be interested (but > overkill?), to > give user the capacity to set that limit (currently 50k I guess) > somehow. > > > Pointer to this auto-mute implemetation? > > > 3. Additionally, and perhaps not unrelated, we would alike to be able > to > decide what happen when bandwidth goes down. Right now it feels like > the > video has the priority over the audio. We would like to be able to > explicitly set the audio priority higher than the video in the > underlying > system, as opposed to implement a stats listener, which triggers > re-negotiation (with the corresponding O/A delay) when bandwidth goes > below > a certain threshold. > > > Right now they have the same "priority", but really audio is typically > fixed, so the video reacts to changes in the apparent level of > delay/buffering. What you may be seeing is better (or less-obvious) > error > control and recovery in the video; the eye is often less sensitive to > things > like dropped frames than the ear. > > I'd love to see a trace/packet-capture/screen-scrape-recording where > you see > that apparent behavior. > > > > - call controls like mute / hold > Right now, you can mute a local stream, but it does not seem to be > possible > to let the remote peers know about the stream being muted. We ended up > implementing a specific off band message for that, but we believe that > the > stream/track could carry this information. This is more important for > video > than audio, as a muted video stream is displayed as a black square, > while a > muted audio as no audible consequence. We believe that this mute / hold > scenario will be frequent enough, that we should have a standardized > way of > doing it, or interop will be very difficult. > > > There is no underlying standard in IETF for communicating this; it's > typically at the application level. And while we don't have good ways > in > MediaStream to do this yet, I strongly prefer to send an fixed image > when > video-muted/holding. Black is a bad choice.... > > It would be nice if browsers sent an image, such as "video on hold" - > just like they provide default 404 page renderings. This is a quality > of implementation issue then. Maybe worth registering a bug on > browsers. But also might be worth a note in the spec. > > > - screen/application sharing > We are aware of the security implications, but there is a very very > strong > demand for screen sharing. Beyond screen sharing, the capacity to share > the > displayed content of a given window of the desktop would due even > better. > Most of the time, users only want to display one document, and that > would > also reduce the security risk by not showing system trays. > Collaboration > (the ability to let the remote peer edit the document) would be even > better, > but we believe it to be outside of the scope of webRTC. > > > yes, and dramatically more risky. Screen-sharing and how to preserve > privacy and security is a huge problem. Right now the temporary kludge > is > to have the user whitelist services that can request it (via extensions > typically) > > Yeah, I'm really unhappy about the screen sharing state of affairs, > too. I would much prefer it became a standard browser feature. > > Cheers, > Silvia. > > Randell > > > > - NAT / Firewall penetration feedback - ICE process feedback > Connectivity is a super super pain to debug, and the number one cause > of > concern. > 1. The 30s time out on chrome generated candidate is biting a lot of > people. > The time out is fine, but there should be an error message that > surfaces > (see 5) > 2. Turn server authentication failure does not generate an error, and > should > (see 5) > 3. ICE state can stay stuck in "checking" forever even after all the > candidate have been exhausted > 4. Not all ICE states stated in the spec are implemented (completed? > fail?) > 5. It would due fantastic to be able to access the list of candidates, > with > their corresponding status (not checked, in use, failed, ….) with the > cause > for failure > 6. In case of success, it would be great to know which candidate is > being > used (google does that with the googActive thingy) but also what is the > type > of the candidate. Right now, on client side, at best you have to go to > chrome://webrtc-internals, get the active candidate, and look it up > from the > list of candidates. When you use a TURN server as a STUN server too, > then > the look up is not an isomorphism. > > right now, the only way to understand what's going on is to have a > "weaponized" version of chrome, or a native app, that gives you access > to > the ICE stack, but we can not expect clients to deploy this, nor to > automate > it. Surfacing those in an API would allow one to: > - adapt the connection strategy on the fly in an iterative fashion on > client > side. > - report automatically the problems and allow remote debug of failed > calls, > > > > On Tue, Jan 7, 2014 at 2:15 AM, Eric Rescorla <ekr@rtfm.com> wrote: > > On Mon, Jan 6, 2014 at 10:10 AM, piranna@gmail.com <piranna@gmail.com> > wrote: > > That's not really going to work unless you basically are on a > public > IP address with no firewall. The issue here isn't the properties of > PeerConnection but the basic way in which NAT traversal algorithms > work. > > I know that the "IP and port" think would work due to NAT, but > nothing > prevent to just only need to exchange one endpoint connection data > instead of both... > > I don't know what you are trying to say here. > > A large fraction of NATs use address/port dependent filtering which > means that there needs to be an outgoing packet from each endpoint > through their NAT to the other side's server reflexive IP in order to > open the pinhole. And that means that each side needs to provide > their address information over the signaling channel. > > I strongly recommend that you go read the ICE specification and > understand the algorithms it describes. That should make clear > why the communications patterns in WebRTC are the way they > are. > > -Ekr > > > > -- > Randell Jesup -- rjesup a t mozilla d o t com > >
Received on Thursday, 9 January 2014 04:03:58 UTC