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Versions: 00

Network Working Group                                        V. Vasiliev
Internet-Draft                                                    Google
Intended status: Standards Track                             May 3, 2019
Expires: November 4, 2019

                  The WebTransport Protocol Framework


   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

   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 November 4, 2019.

Copyright Notice

   Copyright (c) 2019 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

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   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  . . . . . . . . . . . . . . . . . . . . . . .   2
     1.2.  Definitions . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Common Transport Requirements . . . . . . . . . . . . . . . .   5
   3.  Session Establishment . . . . . . . . . . . . . . . . . . . .   5
   4.  Transport Features  . . . . . . . . . . . . . . . . . . . . .   6
     4.1.  Datagrams . . . . . . . . . . . . . . . . . . . . . . . .   6
     4.2.  Streams . . . . . . . . . . . . . . . . . . . . . . . . .   6
     4.3.  Protocol-Specific Features  . . . . . . . . . . . . . . .   7
     4.4.  Bandwidth Prediction  . . . . . . . . . . . . . . . . . .   7
   5.  Buffering and Prioritization  . . . . . . . . . . . . . . . .   8
   6.  Transport Properties  . . . . . . . . . . . . . . . . . . . .   8
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .   8
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   9
   9.  References  . . . . . . . . . . . . . . . . . . . . . . . . .   9
     9.1.  Normative References  . . . . . . . . . . . . . . . . . .   9
     9.2.  Informative References  . . . . . . . . . . . . . . . . .  10
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  11

1.  Introduction

1.1.  Background

   Historically, web applications that needed bidirectional data stream
   between a client and a server could rely on WebSockets [RFC6455], a
   message-based protocol compatible with 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 the 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
   general, it is possible to use it for the use cases addressed by this
   specification; however, in practice, its adoption in a non-browser-
   to-browser by the web developers has been quite low due to dependency
   on ICE (which fits poorly with the Web model) and userspace SCTP
   (which has very few implementations available).

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   Another option potentially available is layering 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 for 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 QUIC DATAGRAM
      frame) 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 the end 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
      impossible 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).

1.2.  Definitions

   The keywords "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "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 the specific transport-layer
   aspects of it to the application developers.  It is structured around
   the following concepts:

   Transport:  A transport is a session established between a client and
      a server.  It may correspond to a specific physical connection on
      the transport layer, or it may be a logical entity within an
      existing multiplexed connection.  Each instance of a transport is

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      logically independent of each other even if some transports can
      share same connection underneath.

   Transport protocol:  A transport protocol (WebTransprot protocol in
      context where this might be ambiguos) is a protocol that can be
      used to back a transport on the wire.  It can provide the
      transport features described in this document, and is expected to
      fulfill all of the requirements described herein.

   Datagram:  A datagram is a unit of transmission that is generally
      treated as a single PDU by underlying network layer protocols,
      and, absent fragmentation, is expected to be treated atomically by
      the queues in the network.

   Stream:  A stream is a sequence of bytes that is delivered to the
      receiving application in the same order as it is transmitted by
      the sender.  Streams are assumed to be sufficiently long that they
      cannot be buffered entirely into memory, thus requiring the
      transport protocol and the API to provide stream data before the
      stream is finished.

   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 feature:  A transport feature refers to the ability of a
      specific transport to provide a specific way of communicating
      data, such as supporting datagrams or streams.

   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 the

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   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, for instance, the web browser.

2.  Common Transport Requirements

   Since the clients are potentially untrusted 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

   Any transport protocol used MUST require the user agent to obtain and
   maintain an 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] may 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

   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 the network endpoints that are not
   WebTransport servers.

   Any transport protocol used MUST provide a way for the server to
   filter the clients that can access it by the origin [RFC6454].

3.  Session Establishment

   WebTransport session establishment is in general 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.

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   The client MUST NOT open streams or send datagrams until the session
   is established.  In some situations, it might be possible for the
   client to be able to start sending data earlier, notably using TLS
   0-RTT.  In those situations, the user agent MAY provide client with
   ability to send a limited amount of data (using either streams or
   datagrams).  The client MUST explicitly request for 0-RTT to be used.

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 SHOULD provide datagrams, unidirectional and
   bidirectional streams in order to make the transport protocols easily

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 reliable.  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

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   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

   As streams are reliable, the data sent on a stream has to be flow
   controlled by the transport protocol.  In addition to the flow
   control for the stream data, the creation of new streams has to be
   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

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 a
   lot of WebTransport use cases; for instance, real time media
   applications adapt the video bitrate to be a fraction of throughput
   they expect to be available.  While not all transport protocols can
   provide low-level transport details, any protocol SHOULD provide a
   way to estimate the bandwidth available to the client.

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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

   o  In general, the transport should not use any aggregation
      algorithms, e.g.  Nagle's algorithm [RFC0896].

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 nature of transport without being aware of a
   specific implementation, thus simplifying introducing new transports
   as a drop-in replacement.

   The following properties are defined in this specification:

   o  Stream independence.  Indicates that there is no head of line
      blocking between different streams.

   o  Partial reliability.  Indicates that if stream is reset, none of
      the data sent on it will be retransmitted.  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 the congestion control and other

   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 a lot of 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

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   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 that is reachable but is not a
   WebTransport server.

   WebTransport does not support any traditional means of browser-based
   authentication.  It is not based on HTTP, and hence does not support
   HTTP cookies or HTTP authentication.  Since it uses TLS, individual
   transport protocols MAY expose TLS-based authentication capabilities
   such as client certificates.  However, since in some of those
   protocols, multiple transports can be pooled within the same TLS
   connection, such features would not be universally available.

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,

   [RFC6454]  Barth, A., "The Web Origin Concept", RFC 6454,
              DOI 10.17487/RFC6454, December 2011,

   [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,

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9.2.  Informative References

              Bishop, M., "Hypertext Transfer Protocol Version 3
              (HTTP/3)", draft-ietf-quic-http-20 (work in progress),
              April 2019.

              Iyengar, J. and I. Swett, "QUIC Loss Detection and
              Congestion Control", draft-ietf-quic-recovery-20 (work in
              progress), April 2019.

              Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed
              and Secure Transport", draft-ietf-quic-transport-20 (work
              in progress), April 2019.

              Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
              Channels", draft-ietf-rtcweb-data-channel-13 (work in
              progress), January 2015.

              Rescorla, E., Tschofenig, H., and N. Modadugu, "The
              Datagram Transport Layer Security (DTLS) Protocol Version
              1.3", draft-ietf-tls-dtls13-31 (work in progress), March

   [RFC0896]  Nagle, J., "Congestion Control in IP/TCP Internetworks",
              RFC 896, DOI 10.17487/RFC0896, January 1984,

   [RFC5681]  Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
              Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,

   [RFC6455]  Fette, I. and A. Melnikov, "The WebSocket Protocol",
              RFC 6455, DOI 10.17487/RFC6455, December 2011,

   [RFC8441]  McManus, P., "Bootstrapping WebSockets with HTTP/2",
              RFC 8441, DOI 10.17487/RFC8441, September 2018,

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   [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,

Author's Address

   Victor Vasiliev

   Email: vasilvv@google.com

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