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

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

From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
Date: Thu, 9 Jan 2014 11:54:20 +1100
Message-ID: <CAHp8n2n3irpDVJO9eQuM_cmsrTTsuTO5qMfdiMZeJ04XwBGLyg@mail.gmail.com>
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

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