Overview: Real Time Protocols for Brower-based Applications
draft-alvestrand-dispatch-rtcweb-protocols-00

Abstract

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.

Requirements Language

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].

Status of this Memo

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 Notice

Copyright (c) 2010 IETF Trust and the persons identified as the document authors. All rights reserved.

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Table of Contents

1.  Introduction
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
11.  Acknowledgements
12.  Normative References
§  Author's Address

1.  Introduction

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.

2.  Functionality groups

The functionallity groups that are needed can be specified, more or less from the bottom up, as:

• Data transport: TCP, UDP and the means to set up secure connections between entities.
• Data framing: RTP and other data formats that serve as containers.
• Data formats: Codec specifications, format specifications and functionality specifications for the data passed between systems. Audio and video codecs, as well as formats for data and document sharing, belong in this category.
• Connection management: Setting up connections, agreeing on data formats, changing data formats during the duration of a call; SIP and Jabber/XMPP belong in this category.
• Presentation and control: What needs to happen in order to ensure that interactions behave in a non-surprising manner. This can include floor control, screen layout, voice activated image switching and other such functions - where part of the system require the cooperation between parties. Cisco/Tandberg's TIP was one attempt at specifying this functionality.
• Local system support functions: These are things that need not be specified uniformly, because each participant may choose to do these in a way of the participant's choosing, without affecting the bits on the wire in a way that others have to be cognizant of. Examples in this category include echo cancellation (some forms of it), local authentication and authorization mechanisms, OS access control and the ability to do local recording of conversations.

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.

3.  Data transport

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.)

4.  Data framing and securing

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?]

5.  Data formats

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.

6.  Connection management

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.

7.  Presentation and control

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>.

8.  Local system support functions

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:

• Echo cancellation should be good enough that feedback (defined as a rising volume of sound with no local sound input) does not occur.
• If remote control of camera is offered, the APIs should be available to figure out who's controlling the camera, and possibly deny permission.
• Automatic gain control, if present, should normalize a speaking voice into <whatever dB metrics makes sense here - most important that we have one only>

9.  IANA Considerations

This document makes no request of IANA.

Note to RFC Editor: this section may be removed on publication as an RFC.

10.  Security Considerations

Security of the web-enabled real time communications comes in two pieces:

• Security of the components: The browsers, and other servers involved. The most target-rich environment here is probably the browser; the aim here should be that the introduction of WEBM components introduces no additional vulnerability.
• Security of the communication channels: It should be easy for a participant to reassure himself of the security of his communication - by verifying the crypto parameters of the links he himself participates in, and to get reassurances from the other parties to the communication that they promise that appropriate measures are taken.
• Part of that last part is securing that the participants are who they say they are (when appropriate).

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:

• We assume that the ICE security mechanism is a necessary and sufficient criterion for accepting that a connection attempt is from a communications partner. This means that we trust the randomness of ICE "usernames" and the security of ICE "passwords".
• We assume that the SRTP key exchange mechanisms and security profiles specified provide an adequate level of protection for audio and video media.

(there needs to be more text here)

11.  Acknowledgements

12. Normative References

Author's Address