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

Jonathan Rosenberg jdrosen at jdrosen.net
Sat Jan 29 15:35:52 CET 2011


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
jdrosen at skype.net                              http://www.skype.com
jdrosen at jdrosen.net                            http://www.jdrosen.net




More information about the RTC-Web mailing list