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

From: Alex Gouaillard <alex.gouaillard@temasys.com.sg>
Date: Thu, 9 Jan 2014 09:40:54 +0800
Message-ID: <CA+Gsrjd-3Z00xN-MqnYKPCno_Zdh-4EtQ64r_ProjUTxZtk15w@mail.gmail.com>
To: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
Cc: Randell Jesup <randell-ietf@jesup.org>, public-webrtc <public-webrtc@w3.org>
@ 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 01:41:21 UTC

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