- From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
- Date: Thu, 9 Jan 2014 11:54:20 +1100
- To: Randell Jesup <randell-ietf@jesup.org>
- Cc: public-webrtc <public-webrtc@w3.org>
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 00:55:08 UTC