W3C home > Mailing lists > Public > public-webrtc@w3.org > January 2014

Re: What is missing for building "real" services?

From: Alex Gouaillard <alex.gouaillard@temasys.com.sg>
Date: Thu, 9 Jan 2014 13:19:23 +0800
Message-ID: <CA+Gsrje76pv4f7O2Xw-2Rp-2vZCfQ-C70cXn_riRrxnRfDTnJQ@mail.gmail.com>
To: Eric Rescorla <ekr@rtfm.com>
Cc: cowwoc <cowwoc@bbs.darktech.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
importance of the interest:
We, and the 50 Millions pool of clients we already serve, want this
scenario (full screen sharing), even though we would prefer the
version where only the display of a given (potentially masked on the
origin computer own desktop) window is shared. I cannot speak for
others, but I remember seeing quite a few hints of interest on the
mailing list, and some experiments with chrome screen sharing seem
pretty popular out there. Some of the video conferencing product we
used to sell (*cough*vidyo*cough*) and others already propose this
functionality and it was the main sales point. Many of
not-yet-customers have expressed utmost interest in the use case
described below for either education purpose, or in hospital
environment (regulation are different for tablets, and iPad with a
specific casing are allow in surgery. What was only prototyping in
Research Units when I was at Harvard Medical School (2008-ish) is now
a reality in "standard" hospital and radiology Units as well.

use case / scenario:
The most usual case is sharing a presentation, table, or text document
as a stream in a multi stream (document display + self video + self
audio + potentially other stuff) call. Screen sharing allows to share
the document, but then, you don't see yourself. Sharing separate
window content (as in desktop composition's window), would allow a
better presentation experience for the sender, who who be able to see
the document he is sending (as a local stream), and himself, basically
mirroring what the remote peer could see.

On Thu, Jan 9, 2014 at 12:02 PM, Eric Rescorla <ekr@rtfm.com> wrote:
> 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 05:19:51 UTC

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