Re: [rtcweb] draft-sipdoc-rtcweb-open-wire-protocol-00 (Open In-The-Wire Protocol for RTC-Web)

[Hadriel Kaplan <[e-mail]>]

On Oct 27, 2011, at 4:55 AM, Iñaki Baz Castillo wrote:
> 
> Hi, just wondering if somebody has read this informational draft. I'd
> would really appreciate if folks agree with the conclusiones and
> requirements exposed in the document or want to discuss about them.
> The draft tries to clarify the scope of RTC-Web, and IMHO that's an
> important milestone before we can move on.

I read it as well, and agreed with its statements and conclusions.  What I don't know is whether you wrote it for the purpose of becoming a WG document, or simply as an FYI-type draft.  It has some requirements in it, which I think can be moved into draft-ietf-rtcweb-use-cases-and-requirements (assuming we get consensus to do so).  I thought the requirements were like self-obvious, but some people may not agree, so it's great that you documented them and we can debate them.

One process-based concern about making requirement 4 a WG requirement: you can't actually do SIP over Websocket with a "pure SIP network" until we get Websocket into a SIP-extending RFC as a new transport type.  I wouldn't want to hold up WebRTC docs becoming RFCs, waiting for the DISPATCH and probable SIPCORE process to make a websocket SIP transport into a RFC.  I *want* to add Websocket as a SIP transport type, but it's not actually as trivial as one would think.

One other minor nit: the draft naming scheme doesn't follow IETF conventions.  The convention is to have the main author/editor's last name as the word after "draft-".

-hadriel