This document gives an overview 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.
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 (Bradner, S., “Key words for use in RFCs to Indicate Requirement Levels,” March 1997.) [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/.
Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as “work in progress.”
This Internet-Draft will expire on May 15, 2011.
Copyright (c) 2010 IETF Trust and the persons identified as the document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.
2. Functionality groups
3. Data transport
4. Data framing and securing
5. Data formats
6. Connection management
7. Presentation and control
8. Local system support functions
9. IANA Considerations
10. Security Considerations
12. Normative References
§ Author's Address
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 <insert reference to work with the W3C Device API WG and friends> 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 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 implementaiton; 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 is the subject of a separate draft, "A Datagram Transport for the WEBM profile".[datagrams] (Alvestrand, H., “A Datagram Transport for the RTC-Web profile,” November 2010.) 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?) (QUESTION: Do octet-streams need to be specified to be Web Sockets [I‑D.ietf‑hybi‑thewebsocketprotocol] (Fette, I., “The WebSocket protocol,” October 2010.)
RTP [RFC3550] (Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, “RTP: A Transport Protocol for Real-Time Applications,” July 2003.)and SRTP [RFC3711] (Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, “The Secure Real-time Transport Protocol (SRTP),” March 2004.). The RTP/SAVP profile, defined as part of SRTP, is supported, and "extended RTCP", RTP/SAVPF [RFC4585] (Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, “Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF),” July 2006.), with its secured version RTP/SAVPF [RFC5124] (Ott, J. and E. Carrara, “Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF),” February 2008.)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.)
[QUESTION: Should we say anything about use / support for TLS and/or DTLS?]
The intent of this specification is to allow each communcations 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.
In audio, the OPUS codec[I‑D.ietf‑codec‑opus] (Valin, J. and K. Vos, “Definition of the Opus Audio Codec,” October 2010.) MUST be supported. <NOTE IN DRAFT: NOT SETTLED> For ease of interoperability with gateways to older equipment, G.711 U-law, audio/PCMU, defined in RFC 1890 (Schulzrinne, H., “RTP Profile for Audio and Video Conferences with Minimal Control,” January 1996.) [RFC1890] section 4.4.12, is also mandatory to implement. There is no third mandatory to implement.
In video, the VP8 codec [vp8] (, “The VP8 Codec Specification (not yet published),” December 2010.) MUST be supported.
The Theora codec is also freely available. H.264/AVC and H.264/SVC [I‑D.ietf‑avt‑rtp‑svc] (Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis, “RTP Payload Format for Scalable Video Coding,” November 2010.) are widely enough used that it gives a wider range of communcations partners if they are supported.
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 Jabber 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 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 partis 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 in <insert reference here when available>.
These do not need to be specified, but it is good to speak about available components here.
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 two 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,” draft-ietf-avt-rtp-svc-24 (work in progress), November 2010 (TXT).|
|[I-D.ietf-codec-opus]||Valin, J. and K. Vos, “Definition of the Opus Audio Codec,” draft-ietf-codec-opus-00 (work in progress), October 2010 (TXT).|
|[I-D.ietf-hybi-thewebsocketprotocol]||Fette, I., “The WebSocket protocol,” draft-ietf-hybi-thewebsocketprotocol-03 (work in progress), October 2010 (TXT).|
|[RFC1890]||Schulzrinne, H., “RTP Profile for Audio and Video Conferences with Minimal Control,” RFC 1890, January 1996 (TXT).|
|[RFC2119]||Bradner, S., “Key words for use in RFCs to Indicate Requirement Levels,” BCP 14, RFC 2119, March 1997 (TXT, HTML, XML).|
|[RFC3550]||Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, “RTP: A Transport Protocol for Real-Time Applications,” STD 64, RFC 3550, July 2003 (TXT, PS, PDF).|
|[RFC3711]||Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, “The Secure Real-time Transport Protocol (SRTP),” RFC 3711, March 2004 (TXT).|
|[RFC4585]||Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, “Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF),” RFC 4585, July 2006 (TXT).|
|[RFC5124]||Ott, J. and E. Carrara, “Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF),” RFC 5124, February 2008 (TXT).|
|[datagrams]||Alvestrand, H., “A Datagram Transport for the RTC-Web profile,” November 2010.|
|[vp8]||“The VP8 Codec Specification (not yet published),” December 2010.|
|Harald T. Alvestrand|