| Network Working Group | H. T. . Alvestrand |
| Internet-Draft | |
| Intended status: Standards Track | June 20, 2011 |
| Expires: December 22, 2011 |
Overview: Real Time Protocols for Brower-based Applications
draft-alvestrand-rtcweb-overview-01
This document gives an overview and context of a protocol suite intended for use with real-time applications that can be deployed in browsers - "real time communication on the Web".
It intends to serve as a starting and coordination point to make sure all the parts that are needed to achieve this goal are findable, and that the parts that belong in the Internet protocol suite are fully specified and on the right publication track.
This work is an attempt to synthesize the input of many people, but makes no claims to fully represent the views of any of them. All parts of the document should be regarded as open for discussion, unless the RTCWEB chairs have declared consensus on an item.
This document is a candidate to become a work item of the RTCWEB working group.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119 [RFC2119].
This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- Drafts is at http://datatracker.ietf.org/drafts/current/.
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This Internet-Draft will expire on December 22, 2011.
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The Internet was, from very early in its lifetime, considered a possible vehicle for the deployment of real-time, interactive applications - with the most easily imaginable being audio conversations (aka "Internet telephony") and videoconferencing.
The first attempts to build this were dependent on special networks, special hardware and custom-built software, often at very high prices or at low quality, placing great demands on the infrastructure.
As the available bandwidth has increased, and as processors and other hardware has become ever faster, the barriers to participation have decreased, and it is possible to deliver a satisfactory experience on commonly available computing hardware.
Still, there are a number of barriers to the ability to communicate universally - one of these is that there is, as of yet, no single set of communication protocols that all agree should be made available for communication; another is the sheer lack of universal identification systems (such as is served by telephone numbers or email addresses in other communications systems).
Development of The Universal Solution has proved hard, however, for all the usual reasons. This memo aims to take a more building-block-oriented approach, and try to find consensus on a set of substrate components that we think will be useful in any real-time communications systems.
The last few years have also seen a new platform rise for deployment of services: The browser-embedded application, or "Web application". It turns out that as long as the browser platform has the necessary interfaces, it is possible to deliver almost any kind of service on it.
Traditionally, these interfaces have been delivered by plugins, which had to be downloaded and installed separately from the browser; in the development of HTML5, much promise is seen by the possibility of making those interfaces available in a standardized way within the browser.
This memo specifies a set of building blocks that can be made accessible and controllable through a Javascript API interface in a browser, and which together form a necessary and sufficient set of functions to allow the use of interactive audio and video in applications that communicate directly between browsers across the Internet.
Other efforts, for instance the W3C WebRTC, Web Applications and Device API working groups, focus on making standardized APIs and interfaces available, within or alongside the HTML5 effort, for those functions; this memo concentrates on specifying the protocols and subprotocols that are needed to specify the interactions that happen across the network.
The goal of the RTCWEB protocol specification is to specify a set of protocols that, if all are implemented, will allow the implementation to communicate with another implementation using audio, video and auxiliary data sent along the most direct possible path between the participants.
This document is intended to serve as the roadmap to the RTCWEB specifications. It defines terms used by other pieces of specification, lists references to other specifications that don't need further elaboration in the RTCWEB context, and gives pointers to other documents that form part of the RTCWEB suite.
By reading this document and the documents it refers to, it should be possible to have all information needed to implement an RTCWEB compatible implementation.
The total RTCWEB/WEBRTC effort consists of two pieces:
Together, these two specifications aim to provide an environment where Javascript embedded in any page, viewed in any compatible browser, when suitably authorized by its user, is able to set up communication using audio, video and auxiliary data, where the browser environment does not constrain the types of application in which this functionality can be used.
The protocol specification does not assume that all implementations implement this API; it is not intended to be possible by observing the bits on the wire whether they come from a browser or from another device implementing this specification.
The goal of cooperation between the protocol specification and the API specification is that for all options and features of the protocol specification, it should be clear which API calls to make to exercise that option or feature; similarly, for any sequence of API calls, it should be clear which protocol options and features will be invoked. Both subject to constraints of the implementation, of course.
The "Mission statement of the IETF" [RFC3935] states that "The benefit of a standard to the Internet is in interoperability - that multiple products implementing a standard are able to work together in order to deliver valuable functions to the Internet's users."
Communication on the Internet frequently occurs in two phases:
There are often many choices that can be made for communicative functionality; the history of the Internet is rife with the proposal, standardization, implementation, and success or failure of many types of options, in all sorts of protocols.
The goal of having a mandatory to implement function set is to prevent negotiation failure, not to preempt or prevent negotiation.
The presence of a mandatory to implement function set serves as a strong changer of the marketplace of deployment - in that it gives a guarantee that, as long as you conform to a specification, and the other party is willing to accept communication at the base level of that specification, you can communicate successfully.
The alternative - that of having no mandatory to implement - does not mean that you cannot communicate, it merely means that in order to be part of the communications partnership, you have to implement the standard "and then some" - that "and then some" usually being called a profile of some sort; in the version most antithetical to the Internet ethos, that "and then some" consists of having to use a specific vendor's product only.
The following terms are used in this document, and as far as possible across the documents specifying the RTCWEB suite, in the specific meanings given here. Other terms are used in their commonly used meaning.
The list is in alphabetical order.
NOTE: Where common definitions exist for these terms, those definitions should be used to the greatest extent possible.
TODO: Extend this list with other terms that might prove slippery.
The functionality groups that are needed can be specified, more or less from the bottom up, as:
Within each functionality group, it is important to preserve both freedom to innovate and the ability for global communication. Freedom to innovate is helped by doing the specification in terms of interfaces, not implementation; any implementation able to communicate according to the interfaces is a valid implementation. Ability to communicate globally is helped both by having core specifications be unencumbered by IPR issues and by having the formats and protocols be fully enough specified to allow for independent implementation.
One can think of the three first groups as forming a "media transport infrastructure", and of the three last groups as forming a "media service". In many contexts, it makes sense to use a common specification for the media transport infrastructure, which can be embedded in browsers and accessed using standard interfaces, and "let a thousand flowers bloom" in the "media service" layer; to achieve interoperable services, however, at least the first five of the six groups need to be specified.
Data transport refers to the sending and receiving of data over the network interfaces, the choice of network-layer addresses at each end of the communication, and the interaction with any intermediate entities that handle the data, but do not modify it (such as TURN relays).
It includes necessary functions for congestion control: When not to send data.
The data transport protocols used by RTCWEB are described in <WORKING GROUP DRAFT "TRANSPORTS">.
The interactions with intermediate boxes, such as firewalls, relays and NAT boxes, is described in <WORKING GROUP DRAFT "PEER TO PEER CONNECTIVITY">.
SRTP [RFC3550] is used for transport of all real-time media.
The detailed considerations for usage of functions from RTP and SRTP, as well as for non-media real-time data, are given in <WORKING GROUP DRAFT "MEDIA TRANSPORTS">.
The intent of this specification is to allow each communications event to use the data formats that are best suited for that particular instance, where a format is supported by both sides of the connection. However, a minimum standard is greatly helpful in order to ensure that communication can be achieved. This document specifies a minimum baseline that will be supported by all implementations of this specification, and leaves further codecs to be included at the will of the implementor.
The mandatory to implement codecs, as well as any profiling requirements for both mandatory and optional codecs, is described in <WORKING GROUP DRAFT "MEDIA PROCESSING">.
The methods, mechanisms and requirements for setting up, negotiating and tearing down connections is a large subject, and one where it is desirable to have both interoperability and freedom to innovate.
The particular choices made for RTCWEB are described in <WORKING GROUP DRAFT "SIGNALING AND NEGOTIATION">.
The most important part of control is the user's control over the browser's interaction with input/output devices and communications channels. It is important that the user have some way of figuring out where his audio, video or texting is being sent, for what purported reason, and what guarantees are made by the parties that form part of this control channel. This is largely a local function between the browser, the underlying operating system and the user interface; this is being worked on as part of the W3C API effort. <INSERT REFERENCE TO W3C API SPEC WHEN AVAILABLE>
These are characterized by the fact that the quality of these functions strongly influences the user experience, but the exact algorithm does not need coordination. In some cases (for instance echo cancellation, as described below), the overall system definition may need to specify that the overall system needs to have some characteristics for which these facilities are useful, without requiring them to be implemented a certain way.
Local functions include echo cancellation, volume control, camera management including focus, zoom, pan/tilt controls (if available), and more.
Certain parts of the system SHOULD conform to certain properties, for instance:
The requirements on RTCWEB systems in this category are found in <WORKING GROUP DRAFT "MEDIA PROCESSING">.
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an RFC.
Security of the web-enabled real time communications comes in several pieces:
The security analysis, and the requirements derived from that analysis, is contained in <WORKING GROUP DRAFT "SECURITY">.
The number of people who have taken part in the discussions surroounding this draft are too numerous to list, or even to identify. The ones below have made special, identifiable contributions; this does not mean that others' contributions are less important.
Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus Westerlund and Joerg Ott, who offered techincal contributions on various versions of the draft.
Thanks to Justin Uberti and Simon Leinen for document review.
| [I-D.ietf-avt-rtp-svc] | Wenger, S, Wang, Y, Schierl, T and A Eleftheriadis, "RTP Payload Format for Scalable Video Coding", Internet-Draft draft-ietf-avt-rtp-svc-27, February 2011. |
| [RFC3935] | Alvestrand, H., "A Mission Statement for the IETF", BCP 95, RFC 3935, October 2004. |
This section may be deleted by the RFC Editor when preparing for publication.
Added section "On interoperability and innovation"
Added data confidentiality and integrity to the "data framing" layer
Added congestion management requirements in the "data transport" layer section
Changed need for non-media data from "question: do we need this?" to "Open issue: How do we do this?"
Strengthened disclaimer that listed codecs are placeholders, not decisions.
More details on why the "local system support functions" section is there.
Added section on "Relationship between API and protocol"
Added terminology section
Mentioned congestion management as part of the "data transport" layer in the layer list
Removed most technical content, and replaced with pointers to drafts as requested and identified by the RTCWEB WG chairs.
Added content to acknowledgements section.
Added change log.
Spell-checked document.