On Sun, Oct 10, 2010 at 5:44 PM, Silvia Pfeiffer
<silviapfeiffer1@gmail.com> wrote:
It seems to me that works nicely for 1-to-1 RTC. It wouldn't even require introduction of RTP/RTSP but can simply be pushing encoded a/v packets over the network as fast as possible on the given URL, which would then be a UDP URL?
The URLs I refer to are really just identifiers to identify resources internally inside the browser - for the network connectivity the plan is to use ICE to interactively determine how packets should be routed to the desired endpoint in the presence of NATs and firewalls.
Regarding RTP, we'll want to use RTP for the on-the-wire protocol, so that we get all the goodness of RTP - multiplexing multiple streams and/or formats, detection of losses, lipsync, etc.
I guess for multi-peer RTC a different approach on the network would be required or would it be a full mesh?
Conferencing is a key use case, and I think this design can support either centralized or distributed conferencing models. The nodes will connect to one another using the same wire protocol, but the overall topology of the nodes will be up to the application.
Silvia.On Mon, Oct 11, 2010 at 10:43 AM, Justin Uberti
<juberti@google.com> wrote:
My idea is that we will tie these various pieces together like a filter chain, i.e.
<device> -> [encoder] -> [transport] -> [decoder] -> <video>
where the connections between pieces are made by passing URLs around. (i.e. opening a device yields a URL for a stream, which is supplied to the encoder; at the other end, streams coming from the decoder are identified by a URL, which can then be passed directly to a <video> tag or WebGL texture.)
Other combinations are of course possible, such as direct access to [transport], in the case of a web real-time game, or combining <device> -> encoder -> websocket, for doing live (non-realtime) broadcasts.
We're still figuring out the right interfaces for encoder/decoder; for transport, hopefully the
draft I proposed can serve as a reasonable starting point.
--justinOn Sun, Oct 10, 2010 at 4:26 PM, Stefan Håkansson LK
<stefan.lk.hakansson@ericsson.com> wrote:
That's right, a lot of things remain regarding protocols
and other stuff. But IMHO <device>, StreamAPIs and <audio> and
<video> should be part of the puzzle!
--Stefan
Hi Stefan,
I have seen those, thanks. That's actually the
reason why I asked: because I have already seen it work with the <device>
element and I wondered what the remaining challenges were. It seems there is
lots of discussion about protocols and
codecs.
Cheers,
Silvia.
On Mon, Oct 11, 2010 at 9:49 AM, Stefan Håkansson LK
<stefan.lk.hakansson@ericsson.com>
wrote:
Silvia,
BR,
Stefan
On 10/09/10 03:17, Silvia Pfeiffer wrote:
Thanks for the general invite!
I wonder: has
the HTML5 device element been looked at (http://dev.w3.org/html5/html-device/) and what are the
problems with that solution?
We're the ones who have to look -
and some on the list have been closely involved with writing the
<device> spec. It would be surprising to me if they are not part of the
solution - but just part.
As far as I know, it's still not clear how to
tie a <device> to a media stream - given that media streams aren't
defined yet, this is not very surprising :-)
Cheers,
Silvia.
On Sat, Oct 9, 2010 at 5:28 PM, David Singer
<singer@apple.com> wrote:
Cool,
thanks Harald
as I said during the day, I'd like to separate (as
much as possible) "why is real-time communications on the internet hard?"
(which is true, but a subject the IETF, the ITU, and others are also
grappling with) from "what is interesting/challenging about real-time
communications *in the web*?" -- which I take to mean in pages shown by a
browser.
On Oct 8, 2010, at 4:24 , Harald Alvestrand
wrote:
> Cullen (I think) has changed the permissions on the
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Multimedia and Software Standards, Apple
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