[RTW] Does RTC-WEB need to pick a signaling protocol?

[Erik Lagerway]

I understand the argument and agree that pure http might be the right course
at some point in the future but the fact remains, SIP rules in
communications "today", it will do so for years to come. It might be wise
for us to move forward on a standard that is understood and appreciated by
not only the developer community but also the business community paying the
bills.

Yes, we can build anew atop http, all it takes is time and money.

An http effort will take a while, I think we all know that. How long did it
take for SIP to displace H.323? I think the unanimous answer is "too long".

We have a standard that exists today, one that we are all very familiar
with, one that works! It would seem a shame not to leverage all we (and our
employers) have invested, which is considerable.

This is a major event in communications, we need to consider this carefully
but we also owe it to "standards" to do it quickly, which has not been the
norm it seems.

*Erik Lagerway | hookflash | m. 604.562.8647*


On Sat, Jan 29, 2011 at 6:35 AM, Jonathan Rosenberg <jdrosen at jdrosen.net>wrote:

> I'm starting a separate thread on this, since I don't want to confound it
> with the charter discussion. This is a topic that should be resolved within
> the group itself, and here are my thoughts on it.
>
> If one asks the question on whether it is actually NECESSARY to require
> that a browser implement something like SIP in order to enable voip
> natively, the answer is definitively NO. The browser already provides a tool
> for exchanging messaging of arbitrary content between the browser and a
> server - its called HTTP (and websockets). Through client-side Javascript
> that comes from the server, an application can craft arbitrary protocol
> messaging of its own design between the client and the server. As an obvious
> example, in order to read mail on Gmail, the browser doesn't need to have an
> implementation of IMAP or POP; Gmail's Javascript implements the client side
> of a protocol of Google's design, and it talks to a web server which
> implements the server side of that protocol. The protocol is then then
> carried over HTTP.
>
> As such, if we take our charter here to define only what is truly REQUIRED
> of a browser, in order to enable voip without a plugin, then we do NOT need
> to pick a signaling protocol. All we need are the things which are truly
> impossible or grossly unsuitable for HTTP, and that is the real-time media
> path only. There need only be APIs for pushing in, and extracting out, the
> data that must be exchanged through HTTP-based signaling - and those are
> things like IP addresses and codec selections.
>
> That said, even if one asks the question of whether it is a good idea for
> us to pick something, I think the answer is no. The enormous benefit of the
> web model is its ability for innovation and velocity. Standardization is not
> needed for communications within the domain of the provider; new features
> can be developed and deployed as quickly as they can be conceived. This is
> something which, despite our best efforts here at IETF over the years, we
> have failed to achieve. I think it is critical that we allow web-based voip
> to innovate with the same kind of pace we've seen in the web overall.
>
> One of the arguments made on the list about why we should pick something,
> is that building their own signaling protocols and messaging is "hard" for a
> tiny web developer that just wants to add a bit of voice to their app. In
> such a case, I fully expect that within weeks or months of specification and
> implementation of RTC-WEB stuff in browsers, smart people will develop
> Javascript libraries which do all of this "hard work", along with PHP and
> many other server-side libraries with sit on the other side. None of it
> requires standardization, and we can let the open source community and the
> marketplace innovate on whatever solutions are needed.
>
> Thanks,
> Jonathan R.
> --
> Jonathan D. Rosenberg, Ph.D.                   SkypeID: jdrosen
> Chief Technology Strategist                    Mobile: +1 (732) 766-2496
> Skype                                          SkypeIn: +1 (408) 465-0361
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>
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