On Fri, Mar 18, 2011 at 5:04 AM, Elwell, John <john.elwell@siemens-enterprise.com> wrote:
Ted,
I think there needs to be some mention of interop with existing real-time applications using IETF protocols (such as SIP, RTP). Something along the lines of: "Work should take account of browser-based applications that require to interoperate with existing applications using IETF protocols such as SIP and RTP and commonly deployed audio and video codecs. Solutions that require mediation functions to achieve interoperation might be acceptable, provided such functions are not unduly complex, expensive or detrimental to performance."
So, my personal read is that these work items: "4. Define which RTP functions and extensions shall be supported in the client and their usage for real-time media, including media adaptation to ensure congestion safe usage. 5. Define what functionalities in the solution, such as media codecs, security algorithms, etc., can be extended and how the extensibility mechanisms works. 6. Define a set of media formats that must or should be supported by a client to improve interoperability." and this general statement which follows the work items: "This work will be done primarily by using already defined protocols or functionalities. " Those seem to cover the same ground as your "take account of". If your aim is to get to something more concrete, like "you must be able to build a browser-based SIP softphone using the tools identified by this group", I think you need to unpack that a bit more. Some of the presence aspects of that toolkit, for example, may not be identified here. To put this another way, my personal read is that we're aiming to create the toolkit needed to build real-time media applications in web contexts. Any specific application, including a SIP softphone, would be a use of the toolkit; those will be covered by something like draft-holmberg-rtcweb-ucreqs-00.txt as a use case, rather than in the charter. Again, just my personal take on this. regards, Ted Hardie
John
-----Original Message----- From: rtc-web-bounces@alvestrand.no [mailto:rtc-web-bounces@alvestrand.no] On Behalf Of Ted Hardie Sent: 17 March 2011 16:18 To: rtc-web@alvestrand.no Subject: [RTW] New proposed charter text. Please review before the BoF.
Name: Real-Time Communication in WEB-browsers (RTCWeb)
There are a number of proprietary implementations that provide direct interactive rich communication using audio, video, collaboration, games, etc. between two peers' web-browsers. These are not interoperable, as they require non-standard extensions or plugins to work. There is a desire to standardize the basis for such communication so that interoperable communication can be established between any compatible browsers. The goal is to enable innovation on top of a set of basic components. One core component is to enable real-time media like audio and video, a second is to enable datagram and byte stream data transfer directly between clients.
This work will be done in collaboration with the W3C. The IETF WG will produce architecture and requirements for selection and profiling of the on the wire protocols. The architecture needs to be coordinated with W3C. The IETF WG work will identity state information and events that need to be exposed in the APIs as input to W3C. The W3C will be responsible for defining APIs to ensure that application developers can control the components.
The security goals and requirements will be developed by the WG. The security model needs to be coordinated with the W3C. The work will also consider where support for extensibility is needed. RTP functionalities, media formats, security algorithms are example of things that commonly needs extensions, additions or replacement, and thus some support for negotiation between clients is required.
The WG will perform the following work: 1. Define the communication model in detail, including how session management is to occur within the model. 2. Define a security model that describes the security goals and how the communication model can achieve these goals. 3. Define how NAT and Firewall traversal is to occur. 4. Define which RTP functions and extensions that shall be supported in the client and their usage for real-time media, including media adaptation to ensure congestion safe usage. 5. Define what functionalities in the solution, such as media codecs, security algorithms, etc., that can be extended and how the extensibility mechanisms works. 6. Define a set of media formats that must or should be supported by a client to improve interoperability. 7. Define how non RTP datagram and byte stream data communication between the clients can be done securely and in a congestion safe way. 8. Provide W3C input for the APIs that comes from the communication model and the selected components and protocols that are part of the solution.
This work will be done primarily by using already defined protocols or functionalities. If there is identification of missing protocols or functionalities, such work can be requested to be done in another working group with a suitable charter or by requests for chartering it in this WG or another WG. 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, Location services, IM & Presence, Resource Priority.
The products of the working group will support security and keying as required by BCP 61 and be defined for IPv4, IPv6, and dual stack deployments. The Working Group will consider the possibility of defining a browser component that implements an existing session negotiation and management protocol. 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.
Milestones:
Aug 2011 Architecture and Security and Threat Model sent to W3C
Aug 2011 Use cases and Scenarios document sent to W3C
Sept 2011 Architecture and Security and Threat Model to IESG as Informational
Sept 2011 Use cases and Scenarios for RTCWeb document sent to IESG as Informational
Dec 2011 RTCWeb and Media format specification(s) to IESG as PS
Dec 2011 Information elements and events APIs Input to W3C
Apr 2012 API to Protocol mapping document submitted to the IESG as Informational (if needed) _______________________________________________ RTC-Web mailing list RTC-Web@alvestrand.no http://www.alvestrand.no/mailman/listinfo/rtc-web