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

Erik Lagerway elagerway at gmail.com
Sun Jan 30 07:59:19 CET 2011


I understand your position Jonathan and agree that it 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. I think they, as will everyone  
else that owns a SIP endpoint, will appreciate that sentiment.

Yes, we can build anew atop http, all it takes is time and money,  
money is running in shirt supply. It would be nice if we could get it  
working for those who spent hard earned dollars on SIP, sooner rather  
than later.

An http effort will take a while, we have a standard, one that you are  
very familiar with, let's use that.

-Erik

On 2011-01-29, 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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