Internet DRAFT - draft-vvv-webtransport-overview
draft-vvv-webtransport-overview
Network Working Group V. Vasiliev
Internet-Draft Google
Intended status: Standards Track November 2, 2019
Expires: May 5, 2020
The WebTransport Protocol Framework
draft-vvv-webtransport-overview-01
Abstract
The WebTransport Protocol Framework enables clients constrained by
the Web security model to communicate with a remote server using a
secure multiplexed transport. It consists of a set of individual
protocols that are safe to expose to untrusted applications, combined
with a model that allows them to be used interchangeably.
This document defines the overall requirements on the protocols used
in WebTransport, as well as the common features of the protocols,
support for some of which may be optional.
Status of This Memo
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provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on May 5, 2020.
Copyright Notice
Copyright (c) 2019 IETF Trust and the persons identified as the
document authors. All rights reserved.
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to this document. Code Components extracted from this document must
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
1.1. Background . . . . . . . . . . . . . . . . . . . . . . . 2
1.2. Conventions and Definitions . . . . . . . . . . . . . . . 4
2. Common Transport Requirements . . . . . . . . . . . . . . . . 5
3. Session Establishment . . . . . . . . . . . . . . . . . . . . 6
4. Transport Features . . . . . . . . . . . . . . . . . . . . . 6
4.1. Datagrams . . . . . . . . . . . . . . . . . . . . . . . . 6
4.2. Streams . . . . . . . . . . . . . . . . . . . . . . . . . 7
4.3. Protocol-Specific Features . . . . . . . . . . . . . . . 7
4.4. Bandwidth Prediction . . . . . . . . . . . . . . . . . . 8
5. Buffering and Prioritization . . . . . . . . . . . . . . . . 8
6. Transport Properties . . . . . . . . . . . . . . . . . . . . 8
7. Security Considerations . . . . . . . . . . . . . . . . . . . 9
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9
9. References . . . . . . . . . . . . . . . . . . . . . . . . . 10
9.1. Normative References . . . . . . . . . . . . . . . . . . 10
9.2. Informative References . . . . . . . . . . . . . . . . . 10
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 11
1. Introduction
The WebTransport Protocol Framework enables clients constrained by
the Web security model to communicate with a remote server using a
secure multiplexed transport. It consists of a set of individual
protocols that are safe to expose to untrusted applications, combined
with a model that allows them to be used interchangeably.
This document defines the overall requirements on the protocols used
in WebTransport, as well as the common features of the protocols,
support for some of which may be optional.
1.1. Background
Historically, web applications that needed a bidirectional data
stream between a client and a server could rely on WebSockets
[RFC6455], a message-based protocol compatible with the Web security
model. However, since the abstraction it provides is a single
ordered stream of messages, it suffers from head-of-line blocking
(HOLB), meaning that all messages must be sent and received in order
even if they are independent and some of them are no longer needed.
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This makes it a poor fit for latency-sensitive applications which
rely on partial reliability and stream independence for performance.
One existing option available to Web developers are WebRTC data
channels [I-D.ietf-rtcweb-data-channel], which provide a WebSocket-
like API for a peer-to-peer SCTP channel protected by DTLS. In
theory, it is possible to use it for the use cases addressed by this
specification. However, in practice, its user in non-browser-to-
browser settings has been quite low due to its dependency on ICE
(which fits poorly with the Web model) and userspace SCTP (which has
very few implementations available).
An alternative design would be to layer WebSockets over HTTP/3
[I-D.ietf-quic-http] in a manner similar to how they are currently
layered over HTTP/2 [RFC8441]. That would avoid head-of-line
blocking and provide an ability to cancel a stream by closing the
corresponding WebSocket object. However, this approach has a number
of drawbacks, which all stem primarily from the fact that
semantically each WebSocket is a completely independent entity:
o Each new stream would require a WebSocket handshake to agree on
application protocol used, meaning that it would take at least one
RTT to establish each new stream before the client can write to
it.
o Only clients can initiate streams. Server-initiated streams and
other alternative modes of communication (such as the QUIC
DATAGRAM frame [I-D.pauly-quic-datagram]) are not available.
o While the streams would normally be pooled by the user agent, this
is not guaranteed, and the general process of mapping a WebSocket
to a server is opaque to the client. This introduces
unpredictable performance properties into the system, and prevents
optimizations which rely on the streams being on the same
connection (for instance, it might be possible for the client to
request different retransmission priorities for different streams,
but that would be much more complex unless they are all on the
same connection).
The WebTransport protocol framework avoids all of those issues by
letting applications create a single transport object that can
contain multiple streams multiplexed together in a single context
(similar to SCTP, HTTP/2, QUIC and others), and can be also used to
send unreliable datagrams (similar to UDP).
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1.2. Conventions and Definitions
The keywords "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in BCP
14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown here.
WebTransport is a framework that aims to abstract away the underlying
transport protocol while still exposing a few key transport-layer
aspects to application developers. It is structured around the
following concepts:
Transport session: A transport session is a single communication
context established between a client and a server. It may
correspond to a specific transport-layer connection, or it may be
a logical entity within an existing multiplexed transport-layer
connection. Transport sessions are logically independent from one
another even if some sessions can share an underlying transport-
layer connection.
Transport protocol: A transport protocol (WebTransport protocol in
contexts where this might be ambiguous) is an instantiation of
WebTransport over a given transport-layer protocol.
Datagram: A datagram is a unit of transmission that is treated
atomically.
Stream: A stream is a sequence of bytes that is reliably delivered
to the receiving application in the same order as it was
transmitted by the sender. Streams can be of arbitrary length,
and therefore cannot always be buffered entirely in memory. It is
expected for transport protocols and APIs to provide partial
stream data to the application before the stream has been entirely
received.
Message: A message is a stream that is sufficiently small that it
can be fully buffered before being passed to the application.
WebTransport does not define messages as a primitive, since from
the transport perspective they can be simulated by fully buffering
a stream before passing it to the application. However, this
distinction is important to highlight since some of the similar
protocols and APIs (notably WebSocket [RFC6455]) use messages as a
core abstraction.
Transport property: A transport property is a specific behavior that
may or may not be exhibited by a transport. Some of those are
inherent for all instances of a given transport protocol (TCP-
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based transport cannot support unreliable delivery), while others
can vary even within the same protocol (QUIC connections may or
may not support connection migration).
Server: A WebTransport server is an application that accepts
incoming transport sessions.
Client: A WebTransport client is an application that initiates the
transport session and may be running in a constrained security
context, for instance, a JavaScript application running inside a
browser.
User agent: A WebTransport user agent is a software system that has
an unrestricted access to the host network stack and can create
transports on behalf of the client.
2. Common Transport Requirements
Since clients are not necessarily trusted and have to be constrained
by the Web security model, WebTransport imposes certain requirements
on any specific transport protocol used.
Any transport protocol used MUST use TLS [RFC8446] or a semantically
equivalent security protocol (for instance, DTLS
[I-D.ietf-tls-dtls13]). The protocols SHOULD use TLS version 1.3 or
later, unless they aim for backwards compatibility with legacy
systems.
Any transport protocol used MUST require the user agent to obtain and
maintain explicit consent from the server to send data. For
connection-oriented protocols (such as TCP or QUIC), the connection
establishment and keep-alive mechanisms suffice. For other
protocols, a mechanism such as ICE [RFC8445] can be used.
Any transport protocol used MUST limit the rate at which the client
sends data. This SHOULD be accomplished via a feedback-based
congestion control mechanism (such as [RFC5681] or
[I-D.ietf-quic-recovery]).
Any transport protocol used MUST support simultaneously establishing
multiple sessions between the same client and server.
Any transport protocol used MUST prevent the clients from
establishing transport sessions to network endpoints that are not
WebTransport servers.
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Any transport protocol used MUST provide a way for servers to filter
clients that can access it by checking the initiating origin
[RFC6454].
Any transport protocol used MUST provide a way for a server endpoint
location to be described using a URI [RFC3986]. This enables
integration with various Web platform features that represent
resources as URIs, such as Content Security Policy [CSP].
3. Session Establishment
WebTransport session establishment is most often asynchronous,
although in some transports it can succeed instantaneously (for
instance, if a transport is immediately pooled with an existing
connection). A session MUST NOT be considered established until it
is secure against replay attacks. For instance, in protocols
creating a new TLS 1.3 session [RFC8446], this would mean that the
user agent MUST NOT treat the session as established until it
received a Finished message from the server.
In some cases, the transport protocol might allow transmitting data
before the session is established; an example is TLS 0-RTT data.
Since this data can be replayed by attackers, it MUST NOT be used
unless the client has explicitly requested 0-RTT for specific streams
or datagrams it knows to be safely replayable.
4. Transport Features
The following transport features are defined in this document. This
list is not meant to be comprehensive; future documents may define
new features for both new and already existing transports.
All transport protocols MUST provide datagrams, unidirectional and
bidirectional streams in order to make the transport protocols easily
interchangeable.
4.1. Datagrams
A datagram is a sequence of bytes that is limited in size (generally
to the path MTU) and is not expected to be transmitted reliably. The
general goal for WebTransport datagrams is to be similar in behavior
to UDP while being subject to common requirements expressed in
Section 2.
The WebTransport sender is not expected to retransmit datagrams,
though it may if it is using a TCP-based protocol or some other
underlying protocol that requires reliable delivery. WebTransport
datagrams are not expected to be flow controlled, meaning that the
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receiver might drop datagrams if the application is not consuming
them fast enough.
The application MUST be provided with the maxiumum datagram size that
it can send. The size SHOULD be derived from the result of
performing path MTU discovery.
4.2. Streams
A unidirectional stream is a one-way reliable in-order stream of
bytes where the initiator is the only endpoint that can send data. A
bidirectional stream allows both endpoints to send data and can be
conceptually represented as a pair of unidirectional streams.
The streams are in general expected to follow the semantics and the
state machine of QUIC streams ([I-D.ietf-quic-transport], Sections 2
and 3). TODO: describe the stream state machine explicitly.
A WebTransport stream can be reset, indicating that the endpoint is
not interested in either sending or receiving any data related to the
stream. In that case, the sender is expected to not retransmit any
data that was already sent on that stream.
Streams SHOULD be sufficiently lightweight that they can be used as
messages.
Data sent on a stream is flow controlled by the transport protocol.
In addition to flow controlling stream data, the creation of new
streams is flow controlled as well: an endpoint may only open a
limited number of streams until the peer explicitly allows creating
more streams.
Every stream within a transport has a unique 64-bit number
identifying it. Both unidirectional and bidirectional streams share
the number space. The client and the server have to agree on the
numbering, so it can be referenced in the application payload.
WebTransport does not impose any other specific restrictions on the
structure of stream IDs, and they should be treated as opaque 64-bit
blobs.
4.3. Protocol-Specific Features
In addition to features described above, there are some capabilities
that may be provided by an individual protocol but are not
universally applicable to all protocols. Those are allowed, but any
protocol is expected to be useful without those features, and
portable clients should not rely on them.
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A notable class of protocol-specific features are features available
only in non-pooled transports. Since those transports have a
dedicated connection, a user agent can provide clients with an
extensive amount of transport-level data that would be too noisy and
difficult to interpret when the connection is shared with unrelated
traffic. For instance, a user agent can provide the number of
packets lost, or the number of times stream data was delayed due to
flow control. It can also expose variables related to congestion
control, such as the size of the congestion window or the current
pacing rate.
4.4. Bandwidth Prediction
Using congestion control state and transport metrics, the client can
predict the rate at which it can send data. That is essential for
many WebTransport use cases; for instance, real time media
applications adapt the video bitrate to be a fraction of the
throughput they expect to be available. While not all transport
protocols can provide low-level transport details, all protocols
SHOULD provide the client with an estimate of the available
bandwidth.
5. Buffering and Prioritization
TODO: expand this outline into a full summary.
o Datagrams are intended for low-latency communications, so the
buffers for them should be small, and prioritized over stream
data.
o In general, the transport should not apply aggregation algorithms
(e.g., Nagle's algorithm [RFC0896]) to datagrams.
6. Transport Properties
In addition to common requirements, each transport can have multiple
optional properties associated with it. Querying them allows the
client to ascertain the presence of features it can use without
requiring knowledge of all protocols. This allows introducing new
transports as drop-in replacements for existing ones.
The following properties are defined in this specification:
o Stream independence. This indicates that there is no head of line
blocking between different streams.
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o Partial reliability. This indicates that if a stream is reset,
none of the data sent on it will be retransmitted. This also
indicates that datagrams will not be retransmitted.
o Pooling support. Indicates that multiple transports using this
transport protocol may end up sharing the same transport layer
connection, and thus share a congestion controller and other
contexts.
o Connection mobility. Indicates that the transport may continue
existing even if the network path between the client and the
server changes.
7. Security Considerations
Providing untrusted clients with a reasonably low-level access to the
network comes with risks. This document mitigates those risks by
imposing a set of common requirements described in Section 2.
WebTransport mandates the use of TLS for all protocols implementing
it. This has a dual purpose. On one hand, it protects the transport
from the network, including both potential attackers and ossification
by middleboxes. On the other hand, it protects the network elements
from potential confusion attacks such as the one discussed in
Section 10.3 of [RFC6455].
One potential concern is that even when a transport cannot be
created, the connection error would reveal enough information to
allow an attacker to scan the network addresses that would normally
be inaccessible. Because of that, the user agent that runs untrusted
clients MUST NOT provide any detailed error information until the
server has confirmed that it is a WebTransport endpoint. For
example, the client must not be able to distinguish between a network
address that is unreachable and one that is reachable but is not a
WebTransport server.
WebTransport does not support any traditional means of HTTP-based
authentication. It is not necessarily based on HTTP, and hence does
not support HTTP cookies or HTTP authentication. Since it requires
TLS, individual transport protocols MAY expose TLS-based
authentication capabilities such as client certificates.
8. IANA Considerations
There are no requests to IANA in this document.
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9. References
9.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[RFC3986] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
Resource Identifier (URI): Generic Syntax", STD 66,
RFC 3986, DOI 10.17487/RFC3986, January 2005,
<https://www.rfc-editor.org/info/rfc3986>.
[RFC6454] Barth, A., "The Web Origin Concept", RFC 6454,
DOI 10.17487/RFC6454, December 2011,
<https://www.rfc-editor.org/info/rfc6454>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>.
[RFC8446] Rescorla, E., "The Transport Layer Security (TLS) Protocol
Version 1.3", RFC 8446, DOI 10.17487/RFC8446, August 2018,
<https://www.rfc-editor.org/info/rfc8446>.
9.2. Informative References
[CSP] W3C, "Content Security Policy Level 3", November 2019,
<https://www.w3.org/TR/CSP/>.
[I-D.ietf-quic-http]
Bishop, M., "Hypertext Transfer Protocol Version 3
(HTTP/3)", draft-ietf-quic-http-23 (work in progress),
September 2019.
[I-D.ietf-quic-recovery]
Iyengar, J. and I. Swett, "QUIC Loss Detection and
Congestion Control", draft-ietf-quic-recovery-23 (work in
progress), September 2019.
[I-D.ietf-quic-transport]
Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed
and Secure Transport", draft-ietf-quic-transport-23 (work
in progress), September 2019.
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[I-D.ietf-rtcweb-data-channel]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
Channels", draft-ietf-rtcweb-data-channel-13 (work in
progress), January 2015.
[I-D.ietf-tls-dtls13]
Rescorla, E., Tschofenig, H., and N. Modadugu, "The
Datagram Transport Layer Security (DTLS) Protocol Version
1.3", draft-ietf-tls-dtls13-33 (work in progress), October
2019.
[I-D.pauly-quic-datagram]
Pauly, T., Kinnear, E., and D. Schinazi, "An Unreliable
Datagram Extension to QUIC", draft-pauly-quic-datagram-04
(work in progress), October 2019.
[RFC0896] Nagle, J., "Congestion Control in IP/TCP Internetworks",
RFC 896, DOI 10.17487/RFC0896, January 1984,
<https://www.rfc-editor.org/info/rfc896>.
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
<https://www.rfc-editor.org/info/rfc5681>.
[RFC6455] Fette, I. and A. Melnikov, "The WebSocket Protocol",
RFC 6455, DOI 10.17487/RFC6455, December 2011,
<https://www.rfc-editor.org/info/rfc6455>.
[RFC8441] McManus, P., "Bootstrapping WebSockets with HTTP/2",
RFC 8441, DOI 10.17487/RFC8441, September 2018,
<https://www.rfc-editor.org/info/rfc8441>.
[RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
Connectivity Establishment (ICE): A Protocol for Network
Address Translator (NAT) Traversal", RFC 8445,
DOI 10.17487/RFC8445, July 2018,
<https://www.rfc-editor.org/info/rfc8445>.
Author's Address
Victor Vasiliev
Google
Email: vasilvv@google.com
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