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WEBTRANS                                                     V. Vasiliev
Internet-Draft                                                    Google
Intended status: Standards Track                           17 April 2020
Expires: 19 October 2020


                  The WebTransport Protocol Framework
                    draft-ietf-webtrans-overview-00

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.

Note to Readers

   Discussion of this draft takes place on the WebTransport mailing list
   (webtransport@ietf.org), which is archived at
   <https://mailarchive.ietf.org/arch/search/?email_list=webtransport>.

   The repository tracking the issues for this draft can be found at
   <https://github.com/ietf-wg-webtrans/draft-ietf-webtrans-overview/
   issues>.  The web API draft corresponding to this document can be
   found at <https://wicg.github.io/web-transport/>.

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 https://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 19 October 2020.



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Copyright Notice

   Copyright (c) 2020 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 (https://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.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
     1.1.  Background  . . . . . . . . . . . . . . . . . . . . . . .   3
     1.2.  Conventions and Definitions . . . . . . . . . . . . . . .   4
   2.  Common Transport Requirements . . . . . . . . . . . . . . . .   5
   3.  Session Establishment . . . . . . . . . . . . . . . . . . . .   6
   4.  Transport Features  . . . . . . . . . . . . . . . . . . . . .   6
     4.1.  Datagrams . . . . . . . . . . . . . . . . . . . . . . . .   7
     4.2.  Streams . . . . . . . . . . . . . . . . . . . . . . . . .   7
     4.3.  Protocol-Specific Features  . . . . . . . . . . . . . . .   8
     4.4.  Bandwidth Prediction  . . . . . . . . . . . . . . . . . .   8
   5.  Buffering and Prioritization  . . . . . . . . . . . . . . . .   8
   6.  Transport Properties  . . . . . . . . . . . . . . . . . . . .   9
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .   9
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  10
   9.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  10
     9.1.  Normative References  . . . . . . . . . . . . . . . . . .  10
     9.2.  Informative References  . . . . . . . . . . . . . . . . .  10
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  12

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.





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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.
   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 use 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:

   *  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.

   *  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.

   *  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).




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   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).

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.










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   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-
      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.  STUN Consent
   Freshness [RFC7675] is another example of the mechanism satisfying
   this requirement.





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   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.

   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.




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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
   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.



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   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.

   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.

   *  Datagrams are intended for low-latency communications, so the
      buffers for them should be small, and prioritized over stream
      data.

   *  In general, the transport should not apply aggregation algorithms
      (e.g., Nagle's algorithm [RFC0896]) to datagrams.







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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:

   *  Stream independence.  This indicates that there is no head of line
      blocking between different streams.

   *  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.

   *  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.

   *  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.



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   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.

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>.

   [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>.

   [RFC6454]  Barth, A., "The Web Origin Concept", RFC 6454,
              DOI 10.17487/RFC6454, December 2011,
              <https://www.rfc-editor.org/info/rfc6454>.

   [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>.

9.2.  Informative References

   [CSP]      W3C, "Content Security Policy Level 3", April 2020,
              <https://www.w3.org/TR/CSP/>.

   [RFC6455]  Fette, I. and A. Melnikov, "The WebSocket Protocol",
              RFC 6455, DOI 10.17487/RFC6455, December 2011,
              <https://www.rfc-editor.org/info/rfc6455>.

   [I-D.ietf-rtcweb-data-channel]
              Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
              Channels", Work in Progress, Internet-Draft, draft-ietf-



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              rtcweb-data-channel-13, 4 January 2015,
              <http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-
              data-channel-13.txt>.

   [I-D.ietf-quic-http]
              Bishop, M., "Hypertext Transfer Protocol Version 3
              (HTTP/3)", Work in Progress, Internet-Draft, draft-ietf-
              quic-http-23, 12 September 2019, <http://www.ietf.org/
              internet-drafts/draft-ietf-quic-http-23.txt>.

   [RFC8441]  McManus, P., "Bootstrapping WebSockets with HTTP/2",
              RFC 8441, DOI 10.17487/RFC8441, September 2018,
              <https://www.rfc-editor.org/info/rfc8441>.

   [I-D.pauly-quic-datagram]
              Pauly, T., Kinnear, E., and D. Schinazi, "An Unreliable
              Datagram Extension to QUIC", Work in Progress, Internet-
              Draft, draft-pauly-quic-datagram-04, 22 October 2019,
              <http://www.ietf.org/internet-drafts/draft-pauly-quic-
              datagram-04.txt>.

   [I-D.ietf-tls-dtls13]
              Rescorla, E., Tschofenig, H., and N. Modadugu, "The
              Datagram Transport Layer Security (DTLS) Protocol Version
              1.3", Work in Progress, Internet-Draft, draft-ietf-tls-
              dtls13-33, 11 October 2019, <http://www.ietf.org/internet-
              drafts/draft-ietf-tls-dtls13-33.txt>.

   [RFC7675]  Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M.
              Thomson, "Session Traversal Utilities for NAT (STUN) Usage
              for Consent Freshness", RFC 7675, DOI 10.17487/RFC7675,
              October 2015, <https://www.rfc-editor.org/info/rfc7675>.

   [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>.

   [I-D.ietf-quic-recovery]
              Iyengar, J. and I. Swett, "QUIC Loss Detection and
              Congestion Control", Work in Progress, Internet-Draft,
              draft-ietf-quic-recovery-23, 11 September 2019,
              <http://www.ietf.org/internet-drafts/draft-ietf-quic-
              recovery-23.txt>.








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   [I-D.ietf-quic-transport]
              Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed
              and Secure Transport", Work in Progress, Internet-Draft,
              draft-ietf-quic-transport-23, 11 September 2019,
              <http://www.ietf.org/internet-drafts/draft-ietf-quic-
              transport-23.txt>.

   [RFC0896]  Nagle, J., "Congestion Control in IP/TCP Internetworks",
              RFC 896, DOI 10.17487/RFC0896, January 1984,
              <https://www.rfc-editor.org/info/rfc896>.

Author's Address

   Victor Vasiliev
   Google

   Email: vasilvv@google.com


































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