| Network Working Group | H. T. . Alvestrand |
| Internet-Draft | |
| Intended status: Standards Track | June 05, 2011 |
| Expires: December 07, 2011 |
Overview: Real Time Protocols for Brower-based Applications
draft-alvestrand-rtcweb-overview-00
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 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 07, 2011.
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The Internet was, from very early in its lifetime, considered a possible veichle 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 are, 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 possiblitiy of making those interfaces available in a standardized way within the browser.
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 auxillary 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 auxillary 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 functionallity 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.
Datagram transport is the subject of a separate draft, "A Datagram Transport for the RTC-Web profile".[I-D.alvestrand-dispatch-rtcweb-datagram] The basic approach is to use ICE as a setup mechanism, and to specify mechanisms to use ICE over connections that utilize UDP and TCP if needed to support a basic datagram-passing function with adequate security. In order to deal with complex NAT/firewall situations, relaying using TURN MUST be supported.
For octet-stream transport, TCP is used. (QUESTION: Do we need a TCP relay specification? The use of TURN over TCP and TLS is specified in the TURN RFC - is it suitable?)
(The role of Web Sockets [I-D.ietf-hybi-thewebsocketprotocol] needs to be clarified.)
The data transport MUST behave reasonably in the presence of congested networks; this is usually interpreted as reducing the send rate when congestion is encountered. TCP, when correctly implemented, does this automatically; this is not the case with UDP, and the RTP framing specification does not contain a congestion control component.
Determining an useful congestion handling mechanism is a high priority for work with this specification suite.
Usually when designing data transport for media, one separates out the functions of bandwith estimation (which is a determinant for which codec and which codec parameters to use) and congestion management (reacting to events that change the available bandwidth, such as congestion or media change, in an appropriate manner). The totality of these features MUST ensure that an implementation of the RTCWEB suite is able to coexist on a network with other users, including TCP-based data transfers, without starving them of resources, and without letting itself be starved.
RTP [RFC3550]and SRTP [RFC3711]. The RTP/SAVP profile, defined as part of SRTP, is supported, and "extended RTCP", RTP/SAVPF [RFC4585], with its secured version RTP/SAVPF [RFC5124]is used in order to support codec functionality that depends on this RTP profile, such as
The implementation of SRTP used MUST support encryption using AES-CM with MIC, on both RTP and RTCP channels. <TODO: Add pointer to appropriate profile here> (Note that like for all mandatory-to-implement, there is no requirement that these protocols be used, just that it is possible to negotiate them.)
[OPEN ISSUE; We need to specify a securable format of passing data that is not RTP. One proposal has been to use DTLS over DCCCP, although specifying a "data codec" and using SRTP has been proposed too.]
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.
NOTE IN DRAFT: The particular codecs named are NOT A DECISION. They are included to illustrate possible choices, and to check with the group that the references given are necessary and sufficient for the purpose of specifying an interoperable codec suite.
In audio, the OPUS codec[I-D.ietf-codec-opus] MUST be supported. For ease of interoperability with gateways to older equipment, G.711 U-law, audio/PCMU, defined in RFC 1890 [RFC1890] section 4.4.12, is also mandatory to implement. There is no third mandatory to implement.
In video, the VP8 codec [I-D.westin-payload-vp8] MUST be supported.
The Theora codec is also freely available. H.264/AVC and H.264/SVC [I-D.ietf-avt-rtp-svc] are widely enough used that it gives a wider range of communications partners if they are supported.
The overall set of data formats and parameters, and the identifiers that allow the partners to bind data streams to application-level entities, form a session description. It is vital that the communicating parties have the same session description, and that the session description can be updated while the connection is in progress.
This specification is silent on the definition of connection management protocols. It envisions that implementors will make a choice on whether to implement connection management protocols as a downloadable component, as a browser plug-in, or as a frontend/backend split, where a part of the protocol machinery is downloaded into the browser and uses some mechanism (for instance WebSockets) to communicate back to a backend implementing the rest of the connection management protocol.
XMPP, and its Jingle component, has proved a versatile tool in building interoperable communities, and so has SIP. This suite requires that the browser support establishing and describing connections using a data format for session description capable of representing the information needed by these two protocols, such as one that can be one-to-one transformed into SDP. The exact specification of this API is done elsewhere <insert reference when available>; this API is powerful enough that all interesting parameters of the transport mechanisms specified above are settable, and clear enough that how to connect the API to the protocols is obvious.
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.
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:
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:
This specification addresses some, but not all, of these concerns, and makes some assumptions about the security considerations of other parts of the environment; it is up to the implementor to see that these security assumptions are warranted. In particular:
(there needs to be more text here)
| [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. |