One document matched: draft-alvestrand-dispatch-rtcweb-protocols-00.txt
Network Working Group H. Alvestrand
Internet-Draft Google
Intended status: Standards Track November 11, 2010
Expires: May 15, 2011
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 [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
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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 . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Functionality groups . . . . . . . . . . . . . . . . . . . . . 4
3. Data transport . . . . . . . . . . . . . . . . . . . . . . . . 5
4. Data framing and securing . . . . . . . . . . . . . . . . . . . 5
5. Data formats . . . . . . . . . . . . . . . . . . . . . . . . . 5
6. Connection management . . . . . . . . . . . . . . . . . . . . . 6
7. Presentation and control . . . . . . . . . . . . . . . . . . . 6
8. Local system support functions . . . . . . . . . . . . . . . . 7
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 7
10. Security Considerations . . . . . . . . . . . . . . . . . . . . 7
11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 8
12. Normative References . . . . . . . . . . . . . . . . . . . . . 8
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 9
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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.
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2. Functionality groups
The functionallity groups that are needed can be specified, more or
less from the bottom up, as:
o Data transport: TCP, UDP and the means to set up secure
connections between entities.
o Data framing: RTP and other data formats that serve as containers.
o 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.
o 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.
o 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.
o 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
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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] 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]
4. Data framing and securing
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.)
[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.
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In audio, the OPUS codec[I-D.ietf-codec-opus] 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 [RFC1890] section 4.4.12, is also mandatory to implement. There
is no third mandatory to implement.
In video, the VP8 codec [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 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>.
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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:
o Echo cancellation should be good enough that feedback (defined as
a rising volume of sound with no local sound input) does not
occur.
o If remote control of camera is offered, the APIs should be
available to figure out who's controlling the camera, and possibly
deny permission.
o 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:
o 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.
o 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
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measures are taken.
o Part of that last part is securing that the participants are who
they say they are (when appropriate).
o
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:
o 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".
o 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
[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-23 (work in progress),
October 2010.
[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.
[I-D.ietf-hybi-thewebsocketprotocol]
Fette, I., "The WebSocket protocol",
draft-ietf-hybi-thewebsocketprotocol-03 (work in
progress), October 2010.
[RFC1890] Schulzrinne, H., "RTP Profile for Audio and Video
Conferences with Minimal Control", RFC 1890, January 1996.
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[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[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.
[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.
[datagrams]
Alvestrand, H., "A Datagram Transport for the RTC-Web
profile", November 2010.
[vp8] "The VP8 Codec Specification (not yet published)",
December 2010.
Author's Address
Harald T. Alvestrand
Google
Kungsbron 2
Stockholm, 11122
Sweden
Email: harald@alvestrand.no
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