|Vasiliev||Expires 21 August 2021||[Page]|
- Network Working Group
- Intended Status:
- Standards Track
WebTransport over HTTP/3
WebTransport [OVERVIEW] is a protocol framework that enables clients constrained by the Web security model to communicate with a remote server using a secure multiplexed transport. This document describes a WebTransport protocol that is based on HTTP/3 [HTTP3] and provides support for unidirectional streams, bidirectional streams and datagrams, all multiplexed within the same HTTP/3 connection.¶
Discussion of this draft takes place on the WebTransport mailing list (email@example.com), 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-http3/issues>. The web API draft corresponding to this document can be found at <https://w3c.github.io/webtransport/>.¶
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 21 August 2021.¶
Copyright (c) 2021 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.¶
HTTP/3 [HTTP3] is a protocol defined on top of QUIC [QUIC-TRANSPORT] that can multiplex HTTP requests over a QUIC connection. This document defines a mechanism for multiplexing non-HTTP data with HTTP/3 in a manner that conforms with the WebTransport protocol requirements and semantics [OVERVIEW]. Using the mechanism described here, multiple WebTransport instances can be multiplexed simultaneously with regular HTTP traffic on the same HTTP/3 connection.¶
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.¶
This document follows terminology defined in Section 1.2 of [OVERVIEW]. Note that this document distinguishes between a WebTransport server and an HTTP/3 server. An HTTP/3 server is the server that terminates HTTP/3 connections; a WebTransport server is an application that accepts WebTransport sessions, which can be accessed via an HTTP/3 server.¶
When an HTTP/3 connection is established, both the client and server have to send a SETTINGS_ENABLE_WEBTRANSPORT setting in order to indicate that they both support WebTransport over HTTP/3.¶
WebTransport sessions are initiated inside a given HTTP/3 connection by the client, who sends an extended CONNECT request [RFC8441]. If the server accepts the request, an WebTransport session is established. The resulting stream will be further referred to as a CONNECT stream, and its stream ID is used to uniquely identify a given WebTransport session within the connection. The ID of the CONNECT stream that established a given WebTransport session will be further referred to as a Session ID.¶
After the session is established, the peers can exchange data using the following mechanisms:¶
- A client can create a bidirectional stream using a special indefinite-length HTTP/3 frame that transfers ownership of the stream to WebTransport.¶
- A server can create a bidirectional stream, which is possible since HTTP/3 does not define any semantics for server-initiated bidirectional streams.¶
- Both client and server can create a unidirectional stream using a special stream type.¶
- A datagram can be sent using a QUIC DATAGRAM frame [QUIC-DATAGRAM].¶
An WebTransport session is terminated when the CONNECT stream that created it is closed.¶
In order to indicate support for WebTransport, both the client and the server MUST send a SETTINGS_ENABLE_WEBTRANSPORT value set to "1" in their SETTINGS frame. Endpoints MUST NOT use any WebTransport-related functionality unless the parameter has been negotiated.¶
If SETTINGS_ENABLE_WEBTRANSPORT is negotiated, support for the QUIC DATAGRAMs within HTTP/3 MUST be negotiated as described in [HTTP3-DATAGRAM]; negotiating WebTransport support without negotiating QUIC DATAGRAM extension SHALL result in a H3_SETTINGS_ERROR error.¶
[HTTP3] requires client's
initial_max_bidi_streams transport parameter to be
set to zero. Existing implementation might enforce this requirement before
negotiating settings; thus, the client MUST send a non-zero MAX_STREAMS for
client-initiated bidirectional streams after receiving an appropriate SETTINGS
frame from the server.¶
[RFC8441] defines an extended CONNECT method in Section 4, enabled by the SETTINGS_ENABLE_CONNECT_PROTOCOL parameter. That parameter is only defined for HTTP/2. This document does not create a new multi-purpose parameter to indicate support for extended CONNECT in HTTP/3; instead, the SETTINGS_ENABLE_WEBTRANSPORT setting implies that an endpoint supports extended CONNECT.¶
In order to create a new WebTransport session, a client can send an HTTP
CONNECT request. The
:protocol pseudo-header field ([RFC8441]) MUST be
:scheme field MUST be
https. Both the
:authority and the
:path value MUST be set; those fields indicate the
desired WebTransport server. An
Origin header [RFC6454] MUST be provided
within the request.¶
Upon receiving an extended CONNECT request with a
:protocol field set to
webtransport, the HTTP/3 server can check if it has a WebTransport
server associated with the specified
:path values. If it
does not, it SHOULD reply with status code 404 (Section 6.5.4, [RFC7231]).
If it does, it MAY accept the session by replying with status code 200.
The WebTransport server MUST verify the
Origin header to ensure that the
specified origin is allowed to access the server in question.¶
From the client's perspective, a WebTransport session is established when the client receives a 200 response. From the server's perspective, a session is established once it sends a 200 response. Both endpoints MUST NOT open any streams or send any datagrams on a given session before that session is established. WebTransport over HTTP/3 does not support 0-RTT.¶
From the flow control perspective, WebTransport sessions count against the stream flow control just like regular HTTP requests, since they are established via an HTTP CONNECT request. This document does not make any effort to introduce a separate flow control mechanism for sessions, nor to separate HTTP requests from WebTransport data streams. If the server needs to limit the rate of incoming requests, it has alternative mechanisms at its disposal:¶
HTTP_REQUEST_REJECTEDerror code defined in [HTTP3] indicates to the receiving HTTP/3 stack that the request was not processed in any way.¶
- HTTP status code 429 indicates that the request was rejected due to rate limiting [RFC6585]. Unlike the previous method, this signal is directly propagated to the application.¶
Session IDs are used to demultiplex streams and datagrams belonging to different WebTransport sessions. On the wire, session IDs are encoded using the QUIC variable length integer scheme described in [QUIC-TRANSPORT].¶
Once established, both endpoints can open unidirectional streams. The HTTP/3 unidirectional stream type SHALL be 0x54. The body of the stream SHALL be the stream type, followed by the session ID, encoded as a variable-length integer, followed by the user-specified stream data (Figure 1).¶
WebTransport clients can initiate bidirectional streams by opening an HTTP/3
bidirectional stream and sending an HTTP/3 frame with type
WEBTRANSPORT_STREAM (type=0x41). The format of the frame SHALL be the frame
type, followed by the session ID, encoded as a variable-length integer,
followed by the user-specified stream data (Figure 2). The frame
SHALL last until the end of the stream.¶
WebTransport servers can initiate bidirectional streams by opening a bidirectional stream within the HTTP/3 connection. Note that since HTTP/3 does not define any semantics for server-initiated bidirectional streams, this document is a normative reference for the semantics of such streams for all HTTP/3 connections in which the SETTINGS_ENABLE_WEBTRANSPORT option is negotiated. The format of those streams SHALL be the session ID, encoded as a variable-length integer, followed by the user-specified stream data (Figure 3).¶
Datagrams can be sent using the DATAGRAM frame as defined in [QUIC-DATAGRAM] and [HTTP3-DATAGRAM]. For all HTTP/3 connections in which the SETTINGS_ENABLE_WEBTRANSPORT option is negotiated, the Flow Identifier is set to the session ID. In other words, the format of datagrams SHALL be the session ID, followed by the user-specified payload (Figure 4).¶
In QUIC, a datagram frame can span at most one packet. Because of that, the applications have to know the maximum size of the datagram they can send. However, when proxying the datagrams, the hop-by-hop MTUs can vary. TODO: Describe how the path MTU can be computed, specifically propagation across HTTP proxies.¶
An WebTransport session over HTTP/3 is terminated when either endpoint closes the stream associated with the CONNECT request that initiated the session. Upon learning about the session being terminated, the endpoint MUST stop sending new datagrams and reset all of the streams associated with the session.¶
WebTransport over HTTP/3 satisfies all of the security requirements imposed by [OVERVIEW] on WebTransport protocols, thus providing a secure framework for client-server communication in cases when the client is potentially untrusted.¶
WebTransport over HTTP/3 requires explicit opt-in through the use of a QUIC transport parameter; this avoids potential protocol confusion attacks by ensuring the HTTP/3 server explicitly supports it. It also requires the use of the Origin header, providing the server with the ability to deny access to Web-based clients that do not originate from a trusted origin.¶
Just like HTTP traffic going over HTTP/3, WebTransport pools traffic to different origins within a single connection. Different origins imply different trust domains, meaning that the implementations have to treat each transport as potentially hostile towards others on the same connection. One potential attack is a resource exhaustion attack: since all of the transports share both congestion control and flow control context, a single client aggressively using up those resources can cause other transports to stall. The user agent thus SHOULD implement a fairness scheme that ensures that each transport within connection gets a reasonable share of controlled resources; this applies both to sending data and to opening new streams.¶
The "webtransport" label identifies HTTP/3 used as a protocol for WebTransport:¶
SETTINGS_ENABLE_WEBTRANSPORT parameter indicates that the specified
HTTP/3 connection is WebTransport-capable.¶
WEBTRANSPORT_STREAM frame allows HTTP/3 client-initiated bidirectional
streams to be used by WebTransport:¶
The "WebTransport stream" type allows unidirectional streams to be used by WebTransport:¶
- Bishop, M., Ed., "Hypertext Transfer Protocol Version 3 (HTTP/3)", Work in Progress, Internet-Draft, draft-ietf-quic-http, <https://tools.ietf.org/html/draft-ietf-quic-http>.
- Schinazi, D. and L. Pardue, "Using QUIC Datagrams with HTTP/3", Work in Progress, Internet-Draft, draft-schinazi-masque-h3-datagram-04, , <http://www.ietf.org/internet-drafts/draft-schinazi-masque-h3-datagram-04.txt>.
- Vasiliev, V., "The WebTransport Protocol Framework", Work in Progress, Internet-Draft, draft-ietf-webtrans-overview-latest, <https://tools.ietf.org/html/draft-ietf-webtrans-overview-latest>.
- Pauly, T., Kinnear, E., and D. Schinazi, "An Unreliable Datagram Extension to QUIC", Work in Progress, Internet-Draft, draft-pauly-quic-datagram, <https://tools.ietf.org/html/draft-pauly-quic-datagram>.
- Iyengar, J., Ed. and M. Thomson, Ed., "QUIC: A UDP-Based Multiplexed and Secure Transport", Work in Progress, Internet-Draft, draft-ietf-quic-transport, <https://tools.ietf.org/html/draft-ietf-quic-transport>.
- Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, , <https://www.rfc-editor.org/info/rfc2119>.
- Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform Resource Identifier (URI): Generic Syntax", STD 66, RFC 3986, DOI 10.17487/RFC3986, , <https://www.rfc-editor.org/info/rfc3986>.
- Barth, A., "The Web Origin Concept", RFC 6454, DOI 10.17487/RFC6454, , <https://www.rfc-editor.org/info/rfc6454>.
- Nottingham, M. and R. Fielding, "Additional HTTP Status Codes", RFC 6585, DOI 10.17487/RFC6585, , <https://www.rfc-editor.org/info/rfc6585>.
- Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer Protocol (HTTP/1.1): Message Syntax and Routing", RFC 7230, DOI 10.17487/RFC7230, , <https://www.rfc-editor.org/info/rfc7230>.
- Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer Protocol (HTTP/1.1): Semantics and Content", RFC 7231, DOI 10.17487/RFC7231, , <https://www.rfc-editor.org/info/rfc7231>.
- Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, , <https://www.rfc-editor.org/info/rfc8174>.
- McManus, P., "Bootstrapping WebSockets with HTTP/2", RFC 8441, DOI 10.17487/RFC8441, , <https://www.rfc-editor.org/info/rfc8441>.
- Frindell, A., Kinnear, E., Pauly, T., Vasiliev, V., and G. Xie, "WebTransport using HTTP/2", Work in Progress, Internet-Draft, draft-kinnear-webtransport-http2-01, , <http://www.ietf.org/internet-drafts/draft-kinnear-webtransport-http2-01.txt>.