WebTransport using HTTP/2
draft-ietf-webtrans-http2-01
The information below is for an old version of the document.
| Document | Type |
This is an older version of an Internet-Draft whose latest revision state is "Active".
|
|
|---|---|---|---|
| Authors | Alan Frindell , Eric Kinnear , Tommy Pauly , Victor Vasiliev , Guowu Xie | ||
| Last updated | 2021-07-30 | ||
| Replaces | draft-kinnear-webtransport-http2 | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Formats | |||
| Additional resources | Mailing list discussion | ||
| Stream | WG state | WG Document | |
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draft-ietf-webtrans-http2-01
webtrans A. Frindell
Internet-Draft Facebook Inc.
Intended status: Standards Track E. Kinnear
Expires: 31 January 2022 T. Pauly
Apple Inc.
V. Vasiliev
Google
G. Xie
Facebook Inc.
30 July 2021
WebTransport using HTTP/2
draft-ietf-webtrans-http2-01
Abstract
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/2 [RFC7540] and
provides support for unidirectional streams, bidirectional streams
and datagrams, all multiplexed within the same HTTP/2 connection.
Note to Readers
Discussion of this draft takes place on the WebTransport mailing list
(webtransport@ietf.org (mailto: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/.
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/.
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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 31 January 2022.
Copyright Notice
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.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 3
2. Protocol Overview . . . . . . . . . . . . . . . . . . . . . . 3
3. Session Establishment . . . . . . . . . . . . . . . . . . . . 4
3.1. Establishing a Transport-Capable HTTP/2 Connection . . . 4
3.2. Extended CONNECT in HTTP/2 . . . . . . . . . . . . . . . 4
3.3. Creating a New Session . . . . . . . . . . . . . . . . . 4
3.4. Limiting the Number of Simultaneous Sessions . . . . . . 5
4. WebTransport Features . . . . . . . . . . . . . . . . . . . . 5
4.1. WT_STREAM Frame . . . . . . . . . . . . . . . . . . . . . 6
4.2. WT_RST_STREAM Frame . . . . . . . . . . . . . . . . . . . 7
4.3. WT_STOP_SENDING Frame . . . . . . . . . . . . . . . . . . 8
4.4. WT_DATAGRAM Frame . . . . . . . . . . . . . . . . . . . . 9
5. Session Termination . . . . . . . . . . . . . . . . . . . . . 10
6. Transport Properties . . . . . . . . . . . . . . . . . . . . 11
7. Security Considerations . . . . . . . . . . . . . . . . . . . 11
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 12
8.1. HTTP/2 SETTINGS Parameter Registration . . . . . . . . . 12
8.2. Frame Type Registration . . . . . . . . . . . . . . . . . 12
8.3. HTTP/2 Error Code Registry . . . . . . . . . . . . . . . 13
8.4. Examples . . . . . . . . . . . . . . . . . . . . . . . . 13
9. References . . . . . . . . . . . . . . . . . . . . . . . . . 15
9.1. Normative References . . . . . . . . . . . . . . . . . . 15
9.2. Informative References . . . . . . . . . . . . . . . . . 16
Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . . 17
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Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 17
1. Introduction
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 multiplexing non-HTTP data with
HTTP/2 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/2
connection.
1.1. Terminology
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.
2. Protocol Overview
WebTransport servers are identified by an HTTPS URI as defined in
Section 4.2.2 of [HTTP].
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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 exchange data using
the following mechanisms:
* Both client and server can create a bidirectional or
unidirectional stream using a new HTTP/2 extension frame
(WT_STREAM)
* A datagram can be sent using a new HTTP/2 extension frame
WT_DATAGRAM.
A WebTransport session is terminated when the CONNECT stream that
created it is closed.
3. Session Establishment
3.1. Establishing a Transport-Capable HTTP/2 Connection
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.
3.2. Extended CONNECT in HTTP/2
[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.
3.3. Creating a New Session
As WebTransport sessions are established over HTTP/2, they are
identified using the "https" URI scheme [RFC7230].
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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/2 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 open any streams or send any datagrams on a
given session before that session is established.
3.4. Limiting the Number of Simultaneous Sessions
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_STREAM_REFUSED" error code defined in [RFC7540] indicates to
the receiving HTTP/2 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.
4. WebTransport Features
WebTransport over HTTP/2 provides the following features described in
[OVERVIEW]: unidirectional streams, bidirectional streams and
datagrams, initiated by either endpoint.
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Session IDs are used to demultiplex streams and datagrams belonging
to different WebTransport sessions. On the wire, session IDs are
encoded using a 31-bit integer field.
4.1. WT_STREAM Frame
A new HTTP/2 frame called WT_STREAM is introduced for either endpoint
to establish WebTransport streams. WT_STREAM frames can be sent on a
stream in the "idle", "reserved (local)", "open", or "half-closed
(remote)" state.
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
+---------------+
|Pad Length? (8)|
+-+-------------+-----------------------------------------------+
|R| Session ID (31) |
+-+-------------------------------------------------------------+
| Padding (*) ...
+---------------------------------------------------------------+
Figure 1: WT_STREAM Frame Format
The WT_STREAM frame define the following fields:
Pad Length: An 8-bit field containing the length of the frame padding
in units of octets. This field is conditional (as signified by a "?"
in the diagram) and is only present if the PADDED flag is set.
Session ID: An unsigned 31-bit integer that identifies the stream
Connect Stream for this Web Transport stream. The Session ID MUST be
MUST be an open stream negotiated via the extended CONNECT protocol
with a ":protocol" value of "webtransport".
The WT_STREAM frame defines the following flags:
UNIDIRECTIONAL (0x1): When set, the stream begins in the "half-closed
(remote)" state at the sender, and in the "half-closed (local)" state
at the receiver.
As with all HTTP/2 streams, WebTransport streams initiated by a
client have odd stream IDs and those initiated by a server have even
stream IDs.
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The recipient MUST respond with a stream error of type
WT_STREAM_ERROR if the specified WebTransport Connect Stream does not
exist, is not a stream established via extended CONNECT to use the
"webtransport" protocol, or if it is in the "closed" or "half-closed
(remote)" stream state.
4.2. WT_RST_STREAM Frame
The RST_STREAM frame defined in HTTP/2 moves a stream to the "closed"
state, but WebTransport streams can be reset unidirectionally, as in
QUIC and HTTP/3. A new HTTP/2 frame called WT_RST_STREAM is
introduced for an endpoint to unidirectionally reset a stream for
writing. WT_RST_STREAM frames can be sent on a stream in the "open",
or "half-closed (remote)" state.
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
+---------------------------------------------------------------+
| Error Code (32) |
+---------------------------------------------------------------+
Figure 2: WT_RST_STREAM Frame Format
The WT_RST_STREAM frame contains a single unsigned, 32-bit integer
identifying the error code. The error code indicates why the stream
is being abruptly terminated for writing.
The WT_RST_STREAM frame does not define any flags.
The WT_RST_STREAM frame half-closes the referenced stream and effects
the same state machine transitions as sending or receiving the
END_STREAM flag. If the sender is in the "open" state, it
transitions to "half-closed (local)". If the sender is in the "half-
closed (remote)" state, it transitions to "closed". If the receiver
is in the "open" state, it transitions to "half-closed (remote)". If
the receiver is in the "half-closed (local)" state, it transitions to
"closed".
After sending a WT_RST_STREAM on a stream, the sender MUST NOT send
additional DATA frames for that stream. If another frame is received
on a stream after receiving a WT_RST_STREAM, the recipient MUST treat
this as a connection error of type PROTOCOL_ERROR.
WT_RST_STREAM frames MUST be associated with a WebTransport stream.
If a WT_RST_STREAM frame is received with a stream identifier of 0x0,
or a request or push stream, the recipient MUST treat this as a
connection error of type PROTOCOL_ERROR.
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WT_RST_STREAM frames MUST NOT be sent for a stream in any state other
than "open" or "half-closed (remote)". If a WT_RST_STREAM frame
identifying a stream in the "idle", "reserved (local)", "reserved
(remote)" state is received, the recipient MUST treat this as a
connection error (Section 5.4.1) of type PROTOCOL_ERROR. Because of
race conditions with WT_STOP_SENDING, it is possible to receive a
WT_RST_STREAM in the "half-closed (remote)" or "closed" state. The
recipient should ignore the frame in this case.
A WT_RST_STREAM frame with a length other than 4 octets MUST be
treated as a connection error (Section 5.4.1) of type
FRAME_SIZE_ERROR.
4.3. WT_STOP_SENDING Frame
The RST_STREAM frame defined in HTTP/2 moves a stream to the "closed"
state, but WebTransport streams can be reset unidirectionally, as in
QUIC and HTTP/3. A new HTTP/2 frame called WT_STOP_SENDING is
introduced for an endpoint to unidirectionally reset a stream for
reading. WT_STOP_SENDING frames can be sent on a stream in the
"open", or "half-closed (local)" state.
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
+---------------------------------------------------------------+
| Error Code (32) |
+---------------------------------------------------------------+
Figure 3: WT_STOP_SENDING Frame Format
The WT_STOP_SENDING frame contains a single unsigned, 32-bit integer
identifying the error code. The error code indicates why the reading
the stream is being abandoned.
The WT_STOP_SENDING frame does not define any flags.
The WT_STOP_SENDING frame half-closes the referenced stream and
effects the same state machine transitions as sending or receiving
the END_STREAM flag. If the sender is in the open state, it
transitions to "half-closed (remote)". If the sender is in the
"half-closed (local)" state, it transitions to "closed". If the
receiver is in the "open" state, it transitions to "half-closed
(local)". If the receiver is in the "half-closed (remote)" state, it
transitions to "closed".
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After receiving a WT_STOP_SENDING on a stream, the sender MUST NOT
send additional frames for that stream. After sending the
WT_STOP_SENDING, the sending endpoint MUST be prepared to receive and
handle additional frames sent on the stream that might have been sent
by the peer prior to the arrival of the WT_STOP_SENDING.
WT_STOP_SENDING frames MUST be associated with a WebTransport stream.
If a WT_STOP_SENDING frame is received with a stream identifier of
0x0, or a request or push stream, the recipient MUST treat this as a
connection error of type PROTOCOL_ERROR.
WT_STOP_SENDING frames MUST NOT be sent for a stream in any state
other than "open" or "half-closed (local)". It is possible to
receive a WT_STOP_SENDING in another state however, because the
sender might have closed or reset the stream while the
WT_STOP_SENDING was in flight. If a WT_STOP_SENDING frame
identifying a stream that is already "closed" or "half-closed
(local)", the recipient SHOULD ignore the frame.
It is also possible to receive DATA frames on a WebTransport stream
in the "half-closed (remote)" or "closed" states if the stream
transitioned there via WT_STOP_SENDING. If a DATA frame is received
on a WebTransport stream in one of these states, the recipient MUST
account for its contribution against the connection flow-control
window and MUST NOT treat it as an error.
A WT_STOP_SENDING frame with a length other than 4 octets MUST be
treated as a connection error (Section 5.4.1) of type
FRAME_SIZE_ERROR.
4.4. WT_DATAGRAM Frame
A new HTTP/2 frame called WT_DATAGRAM is introduced for either
endpoint to transmit a datagram. WT_DATAGRAM frames are sent with
Stream Identifier 0.
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
+---------------+
|Pad Length? (8)|
+-+-------------+-----------------------------------------------+
|R| Session ID (31) |
+-+-------------------------------------------------------------+
| Data (*) ...
+---------------------------------------------------------------+
| Padding (*) ...
+---------------------------------------------------------------+
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Figure 4: WT_DATAGRAM Frame Format
The WT_DATAGRAM frame define the following fields:
Pad Length: An 8-bit field containing the length of the frame padding
in units of octets. This field is conditional (as signified by a "?"
in the diagram) and is only present if the PADDED flag is set.
Session ID: An unsigned 31-bit integer that identifies the stream
Connect Stream for this Web Transport stream. The Session ID MUST be
MUST be an open stream negotiated via the extended CONNECT protocol
with a ":protocol" value of "webtransport".
Data: Application data. The amount of data is the remainder of the
frame payload after subtracting the length of the other fields that
are present.
The WT_DATAGRAM frame does not define any flags.
The recipient MAY respond with a stream error of type WT_STREAM_ERROR
if the specified WebTransport Connect Stream does not exist, is not a
stream established via extended CONNECT to use the "webtransport"
protocol, or if it is in the "closed" or "half-closed (remote)"
stream state.
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.
5. Session Termination
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.
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6. Transport Properties
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.
Stream Independence: Http2Transport does not support stream
independence, as HTTP/2 inherently has head of line blocking.
Partial Reliability: 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.
Pooling Support: 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.
Connection Mobility: 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.
7. Security Considerations
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
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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.
8. IANA Considerations
8.1. HTTP/2 SETTINGS Parameter Registration
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.
Setting Name: ENABLE_WEBTRANSPORT
Value: 0x2b603742
Default: 0
Specification: This document
8.2. Frame Type Registration
The following entries are added to the "HTTP/2 Frame Type" registry
established by [RFC7540]:
The "WT_STREAM" frame allows HTTP/2 client- and server-initiated
unidirectional and bidirectional streams to be used by WebTransport:
Code: 0xTBD
Frame Type: WEBTRANSPORT_STREAM
Specification: This document
The "WT_RST_STREAM" frame allows HTTP/2 WebTransport streams to be
unidirectionally reset for writing:
Code: 0xTBD
Frame Type: WEBTRANSPORT_RST_STREAM
Specification: This document
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The "WT_STOP_SENDING" frame allows HTTP/2 WebTransport streams to be
unidirectionally reset for reading:
Code: 0xTBD
Frame Type: WEBTRANSPORT_STOP_SENDING
Specification: This document
The "WT_DATAGRAM" frame allows HTTP/2 client and server to exchange
datagrams used by WebTransport:
Code: 0xTBD
Frame Type: WEBTRANSPORT_DATAGRAM
Specification: This document
8.3. HTTP/2 Error Code Registry
The following entries are added to the "HTTP/2 Error Code" registry
that was established by Section 11.2 of [RFC7540].
Name: WT_STREAM_ERROR
Code: 0xTBD
Description: Invalid use of WT_STREAM frame
Specification: _RFC Editor: Please fill in this value with the RFC
number for this document_
8.4. Examples
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.
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[[ 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
Session ID = 3
DATA
Stream ID = 5
WebTransport Data
DATA + END_STREAM
Stream ID = 5
WebTransport Data
DATA + END_STREAM
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.
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[[ 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
Session ID = 3
DATA
Stream ID = 2
WebTransport Data
DATA + END_STREAM
Stream ID = 2
WebTransport Data
DATA + END_STREAM
Stream ID = 2
WebTransport Data
9. References
9.1. Normative References
[HTTP] Fielding, R. T., Nottingham, M., and J. Reschke, "HTTP
Semantics", Work in Progress, Internet-Draft, draft-ietf-
httpbis-semantics-17, 25 July 2021,
<https://datatracker.ietf.org/doc/html/draft-ietf-httpbis-
semantics-17>.
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[OVERVIEW] Vasiliev, V., "The WebTransport Protocol Framework", Work
in Progress, Internet-Draft, draft-ietf-webtrans-overview-
02, 28 July 2021, <https://datatracker.ietf.org/doc/html/
draft-ietf-webtrans-overview-02>.
[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>.
[RFC6454] Barth, A., "The Web Origin Concept", RFC 6454,
DOI 10.17487/RFC6454, December 2011,
<https://www.rfc-editor.org/rfc/rfc6454>.
[RFC6585] Nottingham, M. and R. Fielding, "Additional HTTP Status
Codes", RFC 6585, DOI 10.17487/RFC6585, April 2012,
<https://www.rfc-editor.org/rfc/rfc6585>.
[RFC7230] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
Protocol (HTTP/1.1): Message Syntax and Routing",
RFC 7230, DOI 10.17487/RFC7230, June 2014,
<https://www.rfc-editor.org/rfc/rfc7230>.
[RFC7231] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
Protocol (HTTP/1.1): Semantics and Content", RFC 7231,
DOI 10.17487/RFC7231, June 2014,
<https://www.rfc-editor.org/rfc/rfc7231>.
[RFC7540] Belshe, M., Peon, R., and M. Thomson, Ed., "Hypertext
Transfer Protocol Version 2 (HTTP/2)", RFC 7540,
DOI 10.17487/RFC7540, May 2015,
<https://www.rfc-editor.org/rfc/rfc7540>.
[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>.
[RFC8441] McManus, P., "Bootstrapping WebSockets with HTTP/2",
RFC 8441, DOI 10.17487/RFC8441, September 2018,
<https://www.rfc-editor.org/rfc/rfc8441>.
[WEBTRANSPORT-H3]
Vasiliev, V., "WebTransport over HTTP/3", Work in
Progress, Internet-Draft, draft-ietf-webtrans-http3-01, 19
May 2021, <https://datatracker.ietf.org/doc/html/draft-
ietf-webtrans-http3-01>.
9.2. Informative References
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[RFC6202] Loreto, S., Saint-Andre, P., Salsano, S., and G. Wilkins,
"Known Issues and Best Practices for the Use of Long
Polling and Streaming in Bidirectional HTTP", RFC 6202,
DOI 10.17487/RFC6202, April 2011,
<https://www.rfc-editor.org/rfc/rfc6202>.
Acknowledgments
Thanks to Anthony Chivetta, Joshua Otto, and Valentin Pistol for
their contributions in the design and implementation of this work.
Authors' Addresses
Alan Frindell
Facebook Inc.
Email: afrind@fb.com
Eric Kinnear
Apple Inc.
One Apple Park Way
Cupertino, California 95014,
United States of America
Email: ekinnear@apple.com
Tommy Pauly
Apple Inc.
One Apple Park Way
Cupertino, California 95014,
United States of America
Email: tpauly@apple.com
Victor Vasiliev
Google
Email: vasilvv@google.com
Guowu Xie
Facebook Inc.
Email: woo@fb.com
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