The WebTransport Protocol Framework
draft-ietf-webtrans-overview-06

WEBTRANS                                                     V. Vasiliev
Internet-Draft                                                    Google
Intended status: Standards Track                        6 September 2023
Expires: 9 March 2024

                  The WebTransport Protocol Framework
                    draft-ietf-webtrans-overview-06

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 an abstract 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 9 March 2024.

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Copyright Notice

   Copyright (c) 2023 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/
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   Please review these documents carefully, as they describe your rights
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   provided without warranty as described in the Revised 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.  Session-Wide Features . . . . . . . . . . . . . . . . . .   6
     4.2.  Datagrams . . . . . . . . . . . . . . . . . . . . . . . .   7
     4.3.  Streams . . . . . . . . . . . . . . . . . . . . . . . . .   7
   5.  Transport Properties  . . . . . . . . . . . . . . . . . . . .   9
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .  10
   7.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  10
   8.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  10
     8.1.  Normative References  . . . . . . . . . . . . . . . . . .  10
     8.2.  Informative References  . . . . . . . . . . . . . . . . .  11
   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 an abstract 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 reliable stream of messages, it suffers from head-of-line
   blocking, meaning that all messages must be sent and received in
   order even if they could be processed independently of each other,
   and some messages may no longer be relevant.  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 [RFC8831], 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, it has not seen wide adoption outside of browser-to-
   browser settings due to its dependency on ICE (which fits poorly with
   the Web model) and userspace SCTP (which has a limited number of
   implementations available due to not being used in other contexts).

   An alternative design would be to open multiple WebSocket connections
   over HTTP/3 [RFC9220].  That would avoid head-of-line blocking and
   provide an ability to cancel a stream by closing the corresponding
   WebSocket session.  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 [RFC9221]) 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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   WebTransport 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:

   WebTransport session:  A WebTransport 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.  WebTransport sessions are logically independent
      from one another even if some sessions can share an underlying
      transport-layer connection.

   WebTransport protocol:  A WebTransport protocol is a specific
      protocol that can be used to establish a WebTransport session.

   Datagram:  A datagram is a unit of transmission that is limited in
      size (typically to the path MTU), does not have an expectation of
      being delivered reliably, and is treated atomically by the
      transport.

   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.
      WebTransport protocols and APIs are expected 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

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      distinction is important to highlight since some of the similar
      protocols and APIs (notably WebSocket [RFC6455]) use messages as a
      core abstraction.

   Server:  A WebTransport server is an application that accepts
      incoming WebTransport sessions.  In cases when WebTransport is
      served over a multiplexed protocol (such as HTTP/2 or HTTP/3),
      "WebTransport server" refers to a handler for a specific
      multiplexed endpoint (e.g. an application handling specific HTTP
      resource), rather than the application listening on a given TCP or
      UDP socket.

   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 protocol used.

   All WebTransport protocols MUST use TLS [RFC8446] or a semantically
   equivalent security protocol (for instance, DTLS [RFC9147]).  The
   protocols SHOULD use TLS version 1.3 or later, unless they aim for
   backwards compatibility with legacy systems.

   All WebTransport protocols 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 a mechanism satisfying this
   requirement.

   All WebTransport protocols 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 [RFC9002]).

   All WebTransport protocols MUST support simultaneously establishing
   multiple sessions between the same client and server.

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   All WebTransport protocols MUST prevent clients from establishing
   transport sessions to network endpoints that are not WebTransport
   servers.

   All WebTransport protocols MUST provide a way for the user agent to
   indicate the origin [RFC6454] of the client to the server.

   All WebTransport protocols 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 an asynchronous process.  A
   session is considered _ready_ from the client's perspective when the
   server has confirmed that it is willing to accept the session with
   the provided origin and URI.  WebTransport protocols MAY allow
   clients to send data before the session is ready; however, they MUST
   NOT use mechanisms that are unsafe against replay attacks without an
   explicit indication from the client.

4.  Transport Features

   All transport protocols MUST provide datagrams, unidirectional and
   bidirectional streams in order to make the transport protocols
   interchangeable.

4.1.  Session-Wide Features

   Any WebTransport protocol SHALL provide the following operations on
   the session:

   establish a session  Create a new WebTransport session given a URI
      [RFC3986] of the requester.  An origin [RFC6454] MUST be given if
      the WebTransport session is coming from a browser client;
      otherwise, it is OPTIONAL.

   terminate a session  Terminate the session while communicating to the
      peer an unsigned 32-bit error code and an error reason string of
      at most 1024 bytes.  As soon as the session is terminated, no
      further application data will be exchanged on it.  The error code
      and string are optional; the default values are 0 and "".  The
      delivery of the error code and string MAY be best-effort.

   Any WebTransport protocol SHALL provide the following events:

   session terminated event  Indicates that the WebTransport session has

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      been terminated, either by the peer or by the local networking
      stack, and no user data can be exchanged on it any further.  If
      the session has been terminated as a result of the peer performing
      the "terminate a session" operation above, a corresponding error
      code and an error string can be provided.

4.2.  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.

   A WebTransport sender is not expected to retransmit datagrams, though
   it may end up doing so if it is using TCP or some other underlying
   protocol that only provides 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 maximum datagram size that
   it can send.  The size SHOULD be derived from the result of
   performing path MTU discovery.

   In the WebTransport model, all of the outgoing and incoming datagrams
   are placed into a size-bound queue (similar to a network interface
   card queue).

   Any WebTransport protocol SHALL provide the following operations on
   the session:

   send a datagram  Enqueues a datagram to be sent to the peer.  This
      can potentially result in the datagram being dropped if the queue
      is full.

   receive a datagram  Dequeues an incoming datagram, if one is
      available.

   get maxiumum datagram size  Returns the largest size of the datagram
      that a WebTransport session is expected to be able to send.

4.3.  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.

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   The streams are in general expected to follow the semantics and the
   state machine of QUIC streams ([RFC9000], 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.  From the perspective of the client, this is presented
   as a size-bounded queue of incoming streams.

   Any WebTransport protocol SHALL provide the following operations on
   the session:

   create a unidirectional stream  Creates an outgoing unidirectional
      stream; this operation may block until the flow control of the
      underlying protocol allows for it to be completed.

   create a bidirectional stream  Creates an outgoing bidirectional
      stream; this operation may block until the flow control of the
      underlying protocol allows for it to be completed.

   receive a unidirectional stream  Removes a stream from the queue of
      incoming unidirectional streams, if one is available.

   receive a bidirectional stream  Removes a stream from the queue of
      incoming unidirectional streams, if one is available.

   Any WebTransport protocol SHALL provide the following operations on
   an individual stream:

   send bytes  Add bytes into the stream send buffer.  The sender can
      also indicate a FIN, signalling the fact that no new data will be
      send on the stream.  Not applicable for incoming unidirectional
      streams.

   receive bytes  Removes bytes from the stream receive buffer.  FIN can
      be received together with the stream data.  Not applicable for
      outgoing unidirectional streams.

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   abort send side  Sends a signal to the peer that the write side of
      the stream has been aborted.  Discards the send buffer; if
      possible, no currently outstanding data is transmitted or
      retransmitted.  An unsigned 8-bit error code can be supplied as a
      part of the signal to the peer; if omitted, the error code is
      presumed to be 0.

   abort receive side  Sends a signal to the peer that the read side of
      the stream has been aborted.  Discards the receive buffer; the
      peer is typically expected to abort the corresponding send side in
      response.  An unsigned 8-bit error code can be supplied as a part
      of the signal to the peer.

   Any WebTransport protocol SHALL provide the following events for an
   individual stream:

   send side aborted  Indicates that the peer has aborted the
      corresponding receive side of the stream.  An unsigned 8-bit error
      code from the peer may be available.

   receive side aborted  Indicates that the peer has aborted the
      corresponding send side of the stream.  An unsigned 8-bit error
      code from the peer may be available.

   Data Recvd state reached  Indicates that no further data will be
      transmitted or retransmitted on the local send side, and that the
      FIN has been sent.  Data Recvd implies that aborting send-side is
      a no-op.

5.  Transport Properties

   WebTransport defines common semantics for multiple protocols to allow
   them to be used interchangeably.  Nevertheless, those protocols still
   have substantially different performance properties that an
   application may want to query.

   The most notable property is support for unreliable data delivery.
   The protocol is defined to support unreliable delivery if:

   *  Resetting a stream results in the lost stream data no longer being
      retransmitted, and

   *  The datagrams are never retransmitted.

   Another important property is pooling support.  Pooling means that
   multiple transport sessions may end up sharing the same transport
   layer connection, and thus share a congestion controller and other
   contexts.

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6.  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.

7.  IANA Considerations

   There are no requests to IANA in this document.

8.  References

8.1.  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/rfc/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/rfc/rfc3986>.

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   [RFC6454]  Barth, A., "The Web Origin Concept", RFC 6454,
              DOI 10.17487/RFC6454, December 2011,
              <https://www.rfc-editor.org/rfc/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/rfc/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/rfc/rfc8446>.

8.2.  Informative References

   [CSP]      W3C, "Content Security Policy Level 3", September 2023,
              <https://www.w3.org/TR/CSP/>.

   [RFC5681]  Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
              Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
              <https://www.rfc-editor.org/rfc/rfc5681>.

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

   [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/rfc/rfc7675>.

   [RFC8831]  Jesup, R., Loreto, S., and M. Tüxen, "WebRTC Data
              Channels", RFC 8831, DOI 10.17487/RFC8831, January 2021,
              <https://www.rfc-editor.org/rfc/rfc8831>.

   [RFC9000]  Iyengar, J., Ed. and M. Thomson, Ed., "QUIC: A UDP-Based
              Multiplexed and Secure Transport", RFC 9000,
              DOI 10.17487/RFC9000, May 2021,
              <https://www.rfc-editor.org/rfc/rfc9000>.

   [RFC9002]  Iyengar, J., Ed. and I. Swett, Ed., "QUIC Loss Detection
              and Congestion Control", RFC 9002, DOI 10.17487/RFC9002,
              May 2021, <https://www.rfc-editor.org/rfc/rfc9002>.

   [RFC9147]  Rescorla, E., Tschofenig, H., and N. Modadugu, "The
              Datagram Transport Layer Security (DTLS) Protocol Version
              1.3", RFC 9147, DOI 10.17487/RFC9147, April 2022,
              <https://www.rfc-editor.org/rfc/rfc9147>.

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   [RFC9220]  Hamilton, R., "Bootstrapping WebSockets with HTTP/3",
              RFC 9220, DOI 10.17487/RFC9220, June 2022,
              <https://www.rfc-editor.org/rfc/rfc9220>.

   [RFC9221]  Pauly, T., Kinnear, E., and D. Schinazi, "An Unreliable
              Datagram Extension to QUIC", RFC 9221,
              DOI 10.17487/RFC9221, March 2022,
              <https://www.rfc-editor.org/rfc/rfc9221>.

Author's Address

   Victor Vasiliev
   Google
   Email: vasilvv@google.com

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