[RTW] The charter formerly know as RTC-WEB take 3

[Cullen Jennings]

Clearly you could put a SIP and/or Jingle stack in a browser and have it control the RTP. Ignore any SIP vs Jingle issues for a second and just assume we had agreement on one or both of them. Some people think this is a good idea, some people think it is a bad idea. The charters were constructed such that the charter would not determine this and the working group could have that argument inside the working group. I really doubt that we could get consensus at this point in time to either rule it out of scope or say that the solutions had to do this. As people develop proposal around the details of the API and how this will all work, I think consensus could start to gather around if this is a good idea or not. 

Perhaps the charter should explicitly say this but that is why it seems so mute on this topic, it is. 

Cullen

On Jan 21, 2011, at 7:58 , Xavier Marjou wrote:

> Hi,
> 
> Something strikes me. So far RTC-Web is known as "an effort to achieve
> a standardized infrastructure in Web browsers on which real-time
> interactive communication between users of the World Wide Web can be
> achieved."
> So what about the selection or definition of a protocol mechanism to
> establish a media session and negotiate media properties? Are they in
> scope or out of scope? (nothing is mentioned about it in the last
> proposal)
> 
> Cheers,
> Xavier
> 
> On Tue, Jan 18, 2011 at 5:58 AM, Cullen Jennings <fluffy at cisco.com> wrote:
> >
> > In my dispatch co-chair role, I tried to take all the comments I had seen on the list about this charter and see if I could address them in a new version of the charter. I probably messed up in some places. There were some conversation that did not seem to be converging so I did not make any changes for theses. Have a read and if you think something needs to be changed, propose text changes along with the reasons why and we will keep the evolving this charter.
> >
> > Thanks
> > Cullen
> >
> > ----------------------------------------------------------------------------------
> >
> > Version: 3
> >
> > Possible Names:
> >
> > RTCWEB
> > WEBRTC
> > STORM: Standardized Transport Oriented for Realtime Media
> > BURN: Browsers Using Realtime Media
> > WAVE: Web And Voice/Video Enablement
> > WAVVE: Web And Voice Video Enablement
> > REALTIME
> > WEBCOMM
> > WREALTIME
> > WEBTIME
> > WEBFLOWS
> > BRAVO  Browser Realtime Audio and VideO
> > COBWEB COmmuication Between WEBclients
> > WHEELTIME
> >
> >
> >
> > Body:
> >
> > Many implementations have been made that use a Web browser to support
> > direct, interactive communications, including voice, video,
> > collaboration, and gaming.  In these implementations, the web server
> > acts as the signaling path between these applications, using locally
> > significant identifiers to set up the association.  Up till now, such
> > applications have typically required the installation of plugins or
> > non-standard browser extensions.  There is a desire to standardize this
> > functionality, so that this type of application can be run in any
> > compatible browser and allow for high-quality real-time communications
> > experiences within the browser.
> >
> > Traditionally, the W3C has defined API and markup languages such as HTML
> > that work in conjunction with with the IETF over the wire protocols such
> > as HTTP to allow web browsers to display media that does not have real
> > time interactive constraints with another human.
> >
> > The W3C and IETF plan to collaborate together in their traditional way
> > to meet the evolving needs of browsers. Specifically the IETF will
> > provide a set of on the wire protocols, including RTP, to meet the needs
> > on interactive communications, and the W3C will define the API and
> > markup to allow web application developers to control the on the wire
> > protocols. This will allow application developers to write applications
> > that run in a browser and facilitate interactive communications between
> > users for voice and video communications, collaboration, and gaming.
> >
> > This working group will select and define a minimal set of protocols
> > that will enable browsers to:
> >
> > * have interactive real time voice and video pairwise between browsers
> >  or other devices using RTP
> >
> > * have interactive real time application data for collaboration and
> >  gaming pairwise between browsers
> >
> > Fortunately very little development of new protocol at IETF is required
> > for this, only selection of existing protocols and selection of minimum
> > capabilities to ensure interoperability. The following protocols are
> > candidates for including in the profile set:
> >
> > 1) RTP/ RTCP
> >
> > 2) a baseline audio codec for high quality interactive audio. Opus will
> >   be one of the codecs considered
> >
> > 3) a baseline audio codec for PSTN interoperability. G.711 and iLBC will
> >   be some of the codecs considered
> >
> > 4) a baseline video codec. H.264 and VP8 will be some of the codecs
> >   considered
> >
> > 5) Diffserv based QoS
> >
> > 6) NAT traversal using ICE
> >
> > 7) media based DTMF
> >
> > 8) support for identifying streams purpose using semantics labels
> >   mappable to the labels in RFC 4574
> >
> > 9) Secure RTP and keying
> >
> > 10) support for IPv4, IPv6 and dual stack browsers
> >
> > Please note the above list is only a set of candidates that the WG may
> > consider and is not list of things that will be in the profile the set.
> >
> > The working group will cooperate closely with the W3C activity that
> > specifies a semantic level API that allows the control and manipulation
> > of all the functionality above. In addition, the API needs to
> > communicate state information and events about what is happening in the
> > browser that to applications running in the browser. These events and
> > state need to include information such as: receiving DTMF in the RTP,
> > RTP and RTCP statistics, and the state of DTLS/SRTP handshakes. The
> > output of this WG will form input to the W3C group that specifies the
> > API.
> >
> > The working group will follow BCP 79, and adhere to the spirit of BCP
> > 79. The working group cannot explicitly rule out the possibility of
> > adopting encumbered technologies; however, the working group will try to
> > avoid encumbered technologies that require royalties or other
> > encumbrances that would prevent such technologies from being easy to use
> > in web browsers.
> >
> > The following topics will be out of scope for the initial phase of the
> > WG but could be added after a recharter: RTSP, RSVP, NSIS, LOST,
> > Geolocation, IM & Presence, NSIS, Resource Priority. RTP Payload formats
> > will not be done in this WG.
> >
> > Milestones:
> >
> > May 2011 Main alternatives identified in drafts
> >
> > Aug 2011 WG draft with text reflecting agreement of what the profile set
> >         should be
> >
> > Sept 2011 Scenarios specification to IESG as Informational
> >
> > Nov 2011 Documentation specifying mapping of protocol functionality to
> >         W3C-specified API produced. This is an input to W3C API work.
> >
> > Dec 2011 Profile specification to IESG as PS
> >
> > Apr 2012 Mapping W3C defied API to IETF protocols to IESG as
> >         Informational. This depends on the W3C API work.
> >
> >
> >
> > _______________________________________________
> > RTC-Web mailing list
> > RTC-Web at alvestrand.no
> > http://www.alvestrand.no/mailman/listinfo/rtc-web
> 


Cullen Jennings
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