| Internet-Draft | WebTransport-H2 | October 2021 |
| Frindell, et al. | Expires 28 April 2022 | [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 uses HTTP semantics to provide support for unidirectional streams, bidirectional streams, and datagrams. This protocol can be used with any HTTP version, however this document focuses on TCP-based HTTP versions, specifically providing examples of these capabilities when multiplexed within the same HTTP/2 [RFC7540] 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-webtransport-http2). 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 28 April 2022.¶
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.¶
This protocol satisfies the WebTransport protocol framework's requirements using HTTP semantics to provide support for unidirectional streams, bidirectional streams, and datagrams.¶
By using HTTP semantics, this protocol allows deployment using any HTTP version, although this document in particular focuses on HTTP/2 as the current most common TCP-based fallback to HTTP/3 running over QUIC [RFC9000].¶
Currently, the only mechanism in HTTP/2 for server to client communication is server push. That is, servers can initiate unidirectional push promised streams to clients, but clients cannot respond to them; they can only accept them or discard them. Additionally, intermediaries along the path may have different server push policies and may not forward push promised streams to the downstream client. This best effort mechanism is not sufficient to reliably deliver messages from servers to clients, limiting server to client use-cases such as chat messages or notifications.¶
Several techniques have been developed to workaround these limitations: long polling [RFC6202], WebSocket [RFC8441], and tunneling using the CONNECT method. All of these approaches have limitations.¶
This document defines a mechanism for establishing bidirectional communication using HTTP semantics in a manner that conforms with the WebTransport protocol requirements and semantics [OVERVIEW]. When running with HTTP/2, multiple WebTransport instances can be multiplexed simultaneously with regular HTTP traffic on the same HTTP/2 connection.¶
It is important to note that, while it is possible to have the WebTransport session remain entirely "self contained" within a given HTTP flow, certain HTTP versions may provide native features, such as datagram support, which can allow for a better performing implementation of the WebTransport client's requirements.¶
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/2 server. An HTTP/2 server is the server that terminates HTTP/2 connections; a WebTransport server is an application that accepts WebTransport sessions, which can be accessed via an HTTP/2 server.¶
WebTransport servers are identified by an HTTPS URI as defined in Section 4.2.2 of [HTTP].¶
When an HTTP/2 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/2.¶
WebTransport sessions are initiated inside a given HTTP/2 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 utilitze the bidirectional bytestream established by the extended CONNECT request to exchange WebTransport frames that multiplex flows of WebTransport data as WebTransport streams. These frames closely mirror a subset of QUIC frames and provide the essential WebTransport features, such as:¶
Data flow on these streams is controlled via a flow control mechanism very similar to the one provided by QUIC using WT_MAX_DATA, WT_MAX_STREAM_DATA, WT_MAX_STREAMS, WT_DATA_BLOCKED, WT_STREAM_DATA_BLOCKED, and WT_STREAMS_BLOCKED frames.¶
WebTransport streams can be closed via a WT_RESET_STREAM frame and a receiver can request that a sender stop sending with a WT_STOP_SENDING frame.¶
A WebTransport session is terminated when the CONNECT stream that created it is closed. This implicitly closes all WebTransport streams that were multiplexed over that CONNECT stream.¶
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.¶
[RFC8441] defines an extended CONNECT method in Section 4, enabled by the SETTINGS_ENABLE_CONNECT_PROTOCOL parameter. An endpoint does not need to send both SETTINGS_ENABLE_CONNECT_PROTOCOL and SETTINGS_ENABLE_WEBTRANSPORT; the SETTINGS_ENABLE_WEBTRANSPORT setting implies that an endpoint supports extended CONNECT.¶
As WebTransport sessions are established over HTTP, 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 (Section 7.1 of [WEBTRANSPORT-H3]). 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 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. Both endpoints MUST NOT send any WebTransport frames on a given session before that session is established.¶
From a flow control perspective, WebTransport sessions count against HTTP/2 session flow control limits 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 WebTransport sessions. If the server needs to limit the rate of incoming requests, it has alternative mechanisms at its disposal:¶
HTTP_STREAM_REFUSED error code defined in [RFC7540] indicates to the
receiving HTTP/2 stack that the request was not processed in any way.¶
WebTransport over TCP-based HTTP semantics provides the following features described in [OVERVIEW]: unidirectional streams, bidirectional streams, and datagrams, initiated by either endpoint.¶
WebTransport streams and datagrams that belong to different WebTransport sessions are identified by the CONNECT stream on which they are transmitted, with one WebTransport session consuming one CONNECT stream.¶
Because WebTransport over TCP-based HTTP semantics relies on the underlying protocols to provide in order and reliable delivery, there are some notable differences from the way in which QUIC handles application data.¶
Endpoints MUST send stream data in order. As there is no ordering mechanism available for the receiver to reassemble incoming data, receivers assume that all data arriving in STREAM frames is contiguous and in order.¶
DATAGRAM frames are delivered to the remote WebTransport endpoint reliably, however this does not require that the receiving implementation deliver that data to the application in a reliable manner.¶
WebTransport streams have states that mirror the states of QUIC streams (Section 3 of [RFC9000]) as closely as possible to aid in ease of implementation.¶
Because WebTransport does not provide an acknowledgement mechanism for WebTransport frames, it relies on the underlying transport's in order delivery to inform stream state transitions. Wherever QUIC relies on receiving an ack for a packet to transition between stream states, WebTransport performs that transition immediately.¶
WebTransport frames mirror their QUIC counterparts as closely as possible to enable maximal reuse of any applicable QUIC infrastructure by implementors.¶
A WebTransport frame begins with a Frame Type and Frame Length which are followed by zero or more fields that are type-dependent.¶
Frame {
Frame Type (i),
Frame Length (i),
Type-Dependent Fields (..),
}
The Frame Type field indicates the type of the frame, defining what type-dependent fields will be present.¶
The Frame Length field indicates the length of the WebTransport frame, including all type-dependent fields and other information. It does not include the size of the Frame Type or Frame Length fields themselves.¶
Both of these fields use a variable-length integer encoding (see Section 16 of [RFC9000]), with one exception. To ensure simple and efficient implementations of frame parsing, the frame type and length MUST use the shortest possible encoding. For example, for the frame types defined in this document, this means a single-byte encoding, even though it is possible to encode these values as a two-, four-, or eight-byte variable-length integer.¶
A WT_PADDING frame (type=0x00) has no semantic value. PADDING frames can be used to introduce additional data between other WebTransport frames and can also be used to provide protection against traffic analysis or for other reasons.¶
WT_PADDING Frame {
Type (i) = 0x00,
Length (i),
}
The padding extends to the end of the WT_PADDING frame and that many bytes can be discarded by the recipient.¶
A WebTransport frame called WT_RESET_STREAM is introduced for either endpoint to abruptly terminate the sending part of a WebTransport stream.¶
An endpoint uses a WT_RESET_STREAM frame (type=0x04) to abruptly terminate the sending part of a stream.¶
After sending a WT_RESET_STREAM, an endpoint ceases transmission and retransmission of WT_STREAM frames on the identified stream. A receiver of WT_RESET_STREAM can discard any data that it already received on that stream.¶
WT_RESET_STREAM Frame {
Type (i) = 0x04,
Length (i) = 0x00,
Stream ID (i),
Application Protocol Error Code (i),
Final Size (i),
}
The WT_RESET_STREAM frame defines the following fields:¶
Stream ID: A variable-length integer encoding of the WebTransport stream ID of the stream being terminated.¶
Application Protocol Error Code: A variable-length integer containing the application protocol error code that indicates why the stream is being closed.¶
Final Size: A variable-length integer indicating the final size of the stream by the WT_RESET_STREAM sender, in units of bytes. This is the amount of flow control credit that is consumed by a stream, see Section 4.5 of [RFC9000].¶
A WebTransport frame called WT_STOP_SENDING is introduced to communicate that incoming data is being discarded on receipt per application request. WT_STOP_SENDING requests that a peer cease transmission on a stream.¶
WT_STOP_SENDING Frame {
Type (i) = 0x05,
Length (i) = 0x00,
Stream ID (i),
Application Protocol Error Code (i),
}
The WT_STOP_SENDING frame defines the following fields:¶
Stream ID: A variable-length integer carrying the WebTransport stream ID of the stream being ignored.¶
Application Protocol Error Code: A variable-length integer containing the application-specified reason the sender is ignoring the stream.¶
WT_STREAM frames implicitly create a stream and carry stream data. The Type field in the WT_STREAM frame takes the form 0b00001XXX (or the set of values from 0x08 to 0x0f) to maximize compatibility with QUIC. However, unlike QUIC, there are only one bit used to determine the fields that are present in the frame:¶
The FIN bit (0x01) indicates that the frame marks the end of the stream. The final size of the stream is the sum of the length, in bytes, of all data previously sent on this stream and the length of this frame without the length of the Stream ID.¶
WT_STREAM Frame {
Type (i) = 0x08..0x0f,
Stream ID (i),
Stream Data (..),
}
A WebTransport frame called WT_MAX_DATA is introduced to inform the peer of the maximum amount of data that can be sent on the WebTransport session as a whole.¶
WT_MAX_DATA Frame {
Type (i) = 0x10,
Length (i) = 0x00,
Maximum Data (i),
}
WT_MAX_DATA frames contain the following field:¶
Maximum Data: A variable-length integer indicating the maximum amount of data that can be sent on the entire connection, in units of bytes.¶
All data sent in WT_STREAM frames counts toward this limit. The sum of the final sizes on all streams, including streams in terminal states, MUST NOT exceed the value advertised by a receiver.¶
A WebTransport frame called WT_MAX_STREAM_DATA is introduced to inform a peer of the maximum amount of data that can be sent on a stream.¶
WT_MAX_STREAM_DATA Frame {
Type (i) = 0x11,
Length (i) = 0x00,
Stream ID (i),
Maximum Stream Data (i),
}
WT_MAX_STREAM_DATA frames contain the following fields:¶
Stream ID: The stream ID of the affected WebTransport stream, encoded as a variable-length integer.¶
Maximum Stream Data: A variable-length integer indicating the maximum amount of data that can be sent on the identified stream, in units of bytes.¶
When counting data toward this limit, an endpoint accounts for the largest amount of data that is sent or received on the stream. The data sent on a stream MUST NOT exceed the largest maximum stream data value advertised by the receiver.¶
A WebTransport frame called WT_MAX_STREAMS is introduced to inform the peer of the cumulative number of streams of a given type it is permitted to open. A WT_MAX_STREAMS frame with a type of 0x12 applies to bidirectional streams, and a WT_MAX_STREAMS frame with a type of 0x13 applies to unidirectional streams.¶
WT_MAX_STREAMS Frame {
Type (i) = 0x12..0x13,
Length (i) = 0x00,
Maximum Streams (i),
}
WT_MAX_STREAMS frames contain the following field:¶
Maximum Streams: A count of the cumulative number of streams of the corresponding type that can be opened over the lifetime of the connection. This value cannot exceed 2^60, as it is not possible to encode stream IDs larger than 2^(62-1).¶
An endpoint MUST NOT open more streams than permitted by the current stream limit set by its peer. For instance, a server that receives a unidirectional stream limit of 3 is permitted to open streams 3, 7, and 11, but not stream 15.¶
Note that this limit includes streams that have been closed as well as those that are open.¶
A sender SHOULD send a WT_DATA_BLOCKED frame (type=0x14) when it wishes to send data but is unable to do so due to WebTransport session-level flow control. WT_DATA_BLOCKED frames can be used as input to tuning of flow control algorithms.¶
WT_DATA_BLOCKED Frame {
Type (i) = 0x14,
Length (i) = 0x00,
Maximum Data (i),
}
WT_DATA_BLOCKED frames contain the following field:¶
Maximum Data: A variable-length integer indicating the session-level limit at which blocking occurred.¶
A sender SHOULD send a WT_STREAM_DATA_BLOCKED frame (type=0x15) when it wishes to send data but is unable to do so due to stream-level flow control. This frame is analogous to WT_DATA_BLOCKED.¶
WT_STREAM_DATA_BLOCKED Frame {
Type (i) = 0x15,
Length (i) = 0x00,
Stream ID (i),
Maximum Stream Data (i),
}
WT_STREAM_DATA_BLOCKED frames contain the following fields:¶
Stream ID: A variable-length integer indicating the WebTransport stream that is blocked due to flow control.¶
Maximum Stream Data: A variable-length integer indicating the offset of the stream at which the blocking occurred.¶
A sender SHOULD send a WT_STREAMS_BLOCKED frame (type=0x16 or 0x17) when it wishes to open a stream but is unable to do so due to the maximum stream limit set by its peer. A WT_STREAMS_BLOCKED frame of type 0x16 is used to indicate reaching the bidirectional stream limit, and a STREAMS_BLOCKED frame of type 0x17 is used to indicate reaching the unidirectional stream limit.¶
A WT_STREAMS_BLOCKED frame does not open the stream, but informs the peer that a new stream was needed and the stream limit prevented the creation of the stream.¶
WT_STREAMS_BLOCKED Frame {
Type (i) = 0x16..0x17,
Length (i) = 0x00,
Maximum Streams (i),
}
WT_STREAMS_BLOCKED frames contain the following field:¶
Maximum Streams: A variable-length integer indicating the maximum number of streams allowed at the time the frame was sent. This value cannot exceed 2^60, as it is not possible to encode stream IDs larger than 2^(62-1).¶
The WT_DATAGRAM frame type (0x31) is used to carry datagram traffic. Frame type 0x30 is also reserved to maintain parity with QUIC, but unused, as all WebTransport frames MUST contain a length field.¶
WT_DATAGRAM Frame {
Type (i) = 0x31,
Length (i),
Datagram Data (..),
}
WT_DATAGRAM frames contain the following fields:¶
Length: A variable-length integer specifying the length of the Datagram Data in bytes. Note that empty (i.e., zero-length) datagrams are allowed.¶
Datagram Data: The content of the datagram to be delivered.¶
The data in WT_DATAGRAM frames is not subject to flow control. The receiver MAY discard this data if it does not have sufficient space to buffer it.¶
An intermediary could forward the data in a WT_DATAGRAM frame over another protocol, such as WebTransport over HTTP/3. 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.¶
An WebTransport session over HTTP/2 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.¶
The WebTransport framework [OVERVIEW] defines a set of optional transport properties that clients can use to determine the presence of features which might allow additional optimizations beyond the common set of properties available via all WebTransport protocols. Below are details about support in Http2Transport for those properties.¶
Http2Transport does not support stream independence, as HTTP/2 inherently has head of line blocking.¶
Http2Transport does not support partial reliability, as HTTP/2 retransmits any lost data. This means that any datagrams sent via Http2Transport will be retransmitted regardless of the preference of the application. The receiver is permitted to drop them, however, if it is unable to buffer them.¶
Http2Transport supports pooling, as multiple transports using Http2Transport may share the same underlying HTTP/2 connection and therefore share a congestion controller and other transport context.¶
Http2Transport does not support connection mobility, unless an underlying transport protocol that supports multipath or migration, such as MPTCP [RFC7540], is used underneath HTTP/2 and TLS. Without such support, Http2Transport connections cannot survive network transitions.¶
WebTransport over HTTP/2 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/2 requires explicit opt-in through the use of HTTP SETTINGS; this avoids potential protocol confusion attacks by ensuring the HTTP/2 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/2, 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 "HTTP/2 Settings" registry established by [RFC7540]:¶
The SETTINGS_ENABLE_WEBTRANSPORT parameter indicates that the specified
HTTP/2 connection is WebTransport-capable.¶
An example of negotiating a WebTransport Stream on an HTTP/2 connection follows. This example is intended to closely follow the example in Section 5.1 of [RFC8441] to help illustrate the differences defined in this document.¶
[[ From Client ]] [[ From Server ]]
SETTINGS
SETTINGS_ENABLE_WEBTRANSPORT = 1
SETTINGS
SETTINGS_ENABLE_WEBTRANSPORT = 1
HEADERS + END_HEADERS
Stream ID = 3
:method = CONNECT
:protocol = webtransport
:scheme = https
:path = /
:authority = server.example.com
origin: server.example.com
HEADERS + END_HEADERS
Stream ID = 3
:status = 200
WT_STREAM
Stream ID = 5
WebTransport Data
WT_STREAM + FIN
Stream ID = 5
WebTransport Data
WT_STREAM + FIN
Stream ID = 5
WebTransport Data
¶
An example of the server initiating a WebTransport Stream follows. The only difference here is the endpoint that sends the first WT_STREAM frame.¶
[[ From Client ]] [[ From Server ]]
SETTINGS
SETTINGS_ENABLE_WEBTRANSPORT = 1
SETTINGS
SETTINGS_ENABLE_WEBTRANSPORT = 1
HEADERS + END_HEADERS
Stream ID = 3
:method = CONNECT
:protocol = webtransport
:scheme = https
:path = /
:authority = server.example.com
origin: server.example.com
HEADERS + END_HEADERS
Stream ID = 3
:status = 200
WT_STREAM
Stream ID = 2
WebTransport Data
WT_STREAM + FIN
Stream ID = 2
WebTransport Data
WT_STREAM + FIN
Stream ID = 2
WebTransport Data
¶
Thanks to Anthony Chivetta, Joshua Otto, and Valentin Pistol for their contributions in the design and implementation of this work.¶