| Internet-Draft | WebTransport-H3 | May 2021 |
| Vasiliev | Expires 20 November 2021 | [Page] |
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 (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-http3/issues>. The web API draft corresponding to this document can be found at <https://w3c.github.io/webtransport/>.¶
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This Internet-Draft will expire on 20 November 2021.¶
Copyright (c) 2021 IETF Trust and the persons identified as the document authors. All rights reserved.¶
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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.¶
WebTransport servers in general are identified by a pair of authority value and path value (defined in [RFC3986] Sections 3.2 and 3.3 correspondingly).¶
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:¶
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.¶
As WebTransport sessions are established over HTTP/3, they are identified
using the https URI scheme [RFC7230].¶
In order to create a new WebTransport session, a client can send an HTTP
CONNECT request. The :protocol pseudo-header field ([RFC8441]) MUST be
set to webtransport. The :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 :authority and :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. 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_REJECTED error code defined in [HTTP3] indicates to the
receiving HTTP/3 stack that the request was not processed in any way.¶
WebTransport over HTTP/3 provides the following features described in [OVERVIEW]: unidirectional streams, bidirectional streams and datagrams, initiated by either endpoint.¶
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).¶
0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | 0x54 (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Session ID (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Stream Body ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
WebTransport endpoints 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.¶
0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | 0x41 (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Session ID (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Stream Body ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
HTTP/3 does not by itself define any semantics for server-initiated bidirectional streams. If WebTransport setting is negotiated by both endpoints, the syntax of the server-initiated bidirectional streams SHALL be the same as the syntax of client-initated bidirectional streams, that is, a sequence of HTTP/3 frames. The only frame defined by this document for use within server-initiated bidirectional streams is WEBTRANSPORT_STREAM.¶
TODO: move the paragraph above into a separate draft; define what happens with already existing HTTP/3 frames on server-initiated bidirectional streams.¶
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 3).¶
0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Session ID (i) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Datagram Body ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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.¶
In WebTransport over HTTP/3, the client MAY send its SETTINGS frame, as well as multiple WebTransport CONNECT requests, WebTransport data streams and WebTransport datagrams, all within a single flight. As those can arrive out of order, a WebTransport server could be put into a situation where it receives a stream or a datagram without a corresponding session. Similarly, a client may receive a server-initiated stream or a datagram before receiving the CONNECT response headers from the server.¶
To handle this case, WebTransport endpoints SHOULD buffer streams and datagrams
until those can be associated with an established session. To avoid resource
exhaustion, the endpoints MUST limit the number of buffered streams and
datagrams. When the number of buffered streams is exceeded, a stream SHALL be
closed by sending a RESET_STREAM and/or STOP_SENDING with the
H3_WEBTRANSPORT_BUFFERED_STREAM_REJECTED error code. When the number of
buffered datagrams is exceeded, a datagram SHALL be dropped. It is up to
an implementation to choose what stream or datagram to discard.¶
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 following entry is added to the "Hypertext Transfer Protocol (HTTP) Upgrade Token Registry" registry established by [RFC7230]:¶
The "webtransport" label identifies HTTP/3 used as a protocol for WebTransport:¶
webtransport¶
WebTransport over HTTP/3¶
This document and [I-D.kinnear-webtransport-http2]¶
The following entry is added to the "HTTP/3 Settings" registry established by [HTTP3]:¶
The SETTINGS_ENABLE_WEBTRANSPORT parameter indicates that the specified
HTTP/3 connection is WebTransport-capable.¶
The following entry is added to the "HTTP/3 Frame Type" registry established by [HTTP3]:¶
The WEBTRANSPORT_STREAM frame allows HTTP/3 client-initiated bidirectional
streams to be used by WebTransport:¶
The following entry is added to the "HTTP/3 Stream Type" registry established by [HTTP3]:¶
The "WebTransport stream" type allows unidirectional streams to be used by WebTransport:¶
The following entry is added to the "HTTP/3 Error Code" registry established by [HTTP3]:¶