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Versions: (draft-hamilton-quic-transport-protocol)
00 01 02 03 04 05 06 07 08 09
QUIC J. Iyengar, Ed.
Internet-Draft Google
Intended status: Standards Track M. Thomson, Ed.
Expires: June 8, 2018 Mozilla
December 5, 2017
QUIC: A UDP-Based Multiplexed and Secure Transport
draft-ietf-quic-transport-08
Abstract
This document defines the core of the QUIC transport protocol. This
document describes connection establishment, packet format,
multiplexing and reliability. Accompanying documents describe the
cryptographic handshake and loss detection.
Note to Readers
Discussion of this draft takes place on the QUIC working group
mailing list (quic@ietf.org), which is archived at
https://mailarchive.ietf.org/arch/search/?email_list=quic [1].
Working Group information can be found at https://github.com/quicwg
[2]; source code and issues list for this draft can be found at
https://github.com/quicwg/base-drafts/labels/-transport [3].
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 June 8, 2018.
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Copyright Notice
Copyright (c) 2017 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 . . . . . . . . . . . . . . . . . . . . . . . . 5
2. Conventions and Definitions . . . . . . . . . . . . . . . . . 5
2.1. Notational Conventions . . . . . . . . . . . . . . . . . 6
3. A QUIC Overview . . . . . . . . . . . . . . . . . . . . . . . 6
3.1. Low-Latency Connection Establishment . . . . . . . . . . 6
3.2. Stream Multiplexing . . . . . . . . . . . . . . . . . . . 7
3.3. Rich Signaling for Congestion Control and Loss Recovery . 7
3.4. Stream and Connection Flow Control . . . . . . . . . . . 7
3.5. Authenticated and Encrypted Header and Payload . . . . . 8
3.6. Connection Migration and Resilience to NAT Rebinding . . 8
3.7. Version Negotiation . . . . . . . . . . . . . . . . . . . 8
4. Versions . . . . . . . . . . . . . . . . . . . . . . . . . . 8
5. Packet Types and Formats . . . . . . . . . . . . . . . . . . 9
5.1. Long Header . . . . . . . . . . . . . . . . . . . . . . . 10
5.2. Short Header . . . . . . . . . . . . . . . . . . . . . . 11
5.3. Version Negotiation Packet . . . . . . . . . . . . . . . 13
5.4. Cryptographic Handshake Packets . . . . . . . . . . . . . 14
5.4.1. Initial Packet . . . . . . . . . . . . . . . . . . . 14
5.4.2. Retry Packet . . . . . . . . . . . . . . . . . . . . 15
5.4.3. Handshake Packet . . . . . . . . . . . . . . . . . . 15
5.5. Protected Packets . . . . . . . . . . . . . . . . . . . . 16
5.6. Connection ID . . . . . . . . . . . . . . . . . . . . . . 16
5.7. Packet Numbers . . . . . . . . . . . . . . . . . . . . . 17
5.7.1. Initial Packet Number . . . . . . . . . . . . . . . . 18
5.8. Handling Packets from Different Versions . . . . . . . . 18
6. Frames and Frame Types . . . . . . . . . . . . . . . . . . . 19
7. Life of a Connection . . . . . . . . . . . . . . . . . . . . 20
7.1. Matching Packets to Connections . . . . . . . . . . . . . 21
7.2. Version Negotiation . . . . . . . . . . . . . . . . . . . 22
7.2.1. Sending Version Negotiation Packets . . . . . . . . . 22
7.2.2. Handling Version Negotiation Packets . . . . . . . . 23
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7.2.3. Using Reserved Versions . . . . . . . . . . . . . . . 23
7.3. Cryptographic and Transport Handshake . . . . . . . . . . 24
7.4. Transport Parameters . . . . . . . . . . . . . . . . . . 25
7.4.1. Transport Parameter Definitions . . . . . . . . . . . 27
7.4.2. Values of Transport Parameters for 0-RTT . . . . . . 29
7.4.3. New Transport Parameters . . . . . . . . . . . . . . 29
7.4.4. Version Negotiation Validation . . . . . . . . . . . 29
7.5. Stateless Retries . . . . . . . . . . . . . . . . . . . . 31
7.6. Proof of Source Address Ownership . . . . . . . . . . . . 31
7.6.1. Client Address Validation Procedure . . . . . . . . . 32
7.6.2. Address Validation on Session Resumption . . . . . . 33
7.6.3. Address Validation Token Integrity . . . . . . . . . 34
7.7. Connection Migration . . . . . . . . . . . . . . . . . . 34
7.7.1. Privacy Implications of Connection Migration . . . . 35
7.7.2. Address Validation for Migrated Connections . . . . . 36
7.8. Spurious Connection Migrations . . . . . . . . . . . . . 37
7.9. Connection Termination . . . . . . . . . . . . . . . . . 38
7.9.1. Closing and Draining Connection States . . . . . . . 38
7.9.2. Idle Timeout . . . . . . . . . . . . . . . . . . . . 40
7.9.3. Immediate Close . . . . . . . . . . . . . . . . . . . 40
7.9.4. Stateless Reset . . . . . . . . . . . . . . . . . . . 41
8. Frame Types and Formats . . . . . . . . . . . . . . . . . . . 43
8.1. Variable-Length Integer Encoding . . . . . . . . . . . . 44
8.2. PADDING Frame . . . . . . . . . . . . . . . . . . . . . . 44
8.3. RST_STREAM Frame . . . . . . . . . . . . . . . . . . . . 45
8.4. CONNECTION_CLOSE frame . . . . . . . . . . . . . . . . . 45
8.5. APPLICATION_CLOSE frame . . . . . . . . . . . . . . . . . 46
8.6. MAX_DATA Frame . . . . . . . . . . . . . . . . . . . . . 46
8.7. MAX_STREAM_DATA Frame . . . . . . . . . . . . . . . . . . 47
8.8. MAX_STREAM_ID Frame . . . . . . . . . . . . . . . . . . . 48
8.9. PING Frame . . . . . . . . . . . . . . . . . . . . . . . 49
8.10. BLOCKED Frame . . . . . . . . . . . . . . . . . . . . . . 50
8.11. STREAM_BLOCKED Frame . . . . . . . . . . . . . . . . . . 50
8.12. STREAM_ID_BLOCKED Frame . . . . . . . . . . . . . . . . . 51
8.13. NEW_CONNECTION_ID Frame . . . . . . . . . . . . . . . . . 51
8.14. STOP_SENDING Frame . . . . . . . . . . . . . . . . . . . 52
8.15. PONG Frame . . . . . . . . . . . . . . . . . . . . . . . 52
8.16. ACK Frame . . . . . . . . . . . . . . . . . . . . . . . . 53
8.16.1. ACK Block Section . . . . . . . . . . . . . . . . . 54
8.16.2. Sending ACK Frames . . . . . . . . . . . . . . . . . 56
8.16.3. ACK Frames and Packet Protection . . . . . . . . . . 57
8.17. STREAM Frames . . . . . . . . . . . . . . . . . . . . . . 58
9. Packetization and Reliability . . . . . . . . . . . . . . . . 59
9.1. Special Considerations for PMTU Discovery . . . . . . . . 62
10. Streams: QUIC's Data Structuring Abstraction . . . . . . . . 62
10.1. Stream Identifiers . . . . . . . . . . . . . . . . . . . 63
10.2. Stream States . . . . . . . . . . . . . . . . . . . . . 64
10.2.1. Send Stream States . . . . . . . . . . . . . . . . . 65
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10.2.2. Receive Stream States . . . . . . . . . . . . . . . 67
10.2.3. Permitted Frame Types . . . . . . . . . . . . . . . 70
10.2.4. Bidirectional Stream States . . . . . . . . . . . . 70
10.3. Solicited State Transitions . . . . . . . . . . . . . . 71
10.4. Stream Concurrency . . . . . . . . . . . . . . . . . . . 72
10.5. Sending and Receiving Data . . . . . . . . . . . . . . . 73
10.6. Stream Prioritization . . . . . . . . . . . . . . . . . 73
11. Flow Control . . . . . . . . . . . . . . . . . . . . . . . . 74
11.1. Edge Cases and Other Considerations . . . . . . . . . . 75
11.1.1. Response to a RST_STREAM . . . . . . . . . . . . . . 76
11.1.2. Data Limit Increments . . . . . . . . . . . . . . . 76
11.2. Stream Limit Increment . . . . . . . . . . . . . . . . . 77
11.2.1. Blocking on Flow Control . . . . . . . . . . . . . . 77
11.3. Stream Final Offset . . . . . . . . . . . . . . . . . . 77
12. Error Handling . . . . . . . . . . . . . . . . . . . . . . . 78
12.1. Connection Errors . . . . . . . . . . . . . . . . . . . 78
12.2. Stream Errors . . . . . . . . . . . . . . . . . . . . . 79
12.3. Transport Error Codes . . . . . . . . . . . . . . . . . 79
12.4. Application Protocol Error Codes . . . . . . . . . . . . 81
13. Security and Privacy Considerations . . . . . . . . . . . . . 81
13.1. Spoofed ACK Attack . . . . . . . . . . . . . . . . . . . 81
13.2. Slowloris Attacks . . . . . . . . . . . . . . . . . . . 82
13.3. Stream Fragmentation and Reassembly Attacks . . . . . . 82
13.4. Stream Commitment Attack . . . . . . . . . . . . . . . . 82
14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 83
14.1. QUIC Transport Parameter Registry . . . . . . . . . . . 83
14.2. QUIC Transport Error Codes Registry . . . . . . . . . . 84
15. References . . . . . . . . . . . . . . . . . . . . . . . . . 87
15.1. Normative References . . . . . . . . . . . . . . . . . . 87
15.2. Informative References . . . . . . . . . . . . . . . . . 88
15.3. URIs . . . . . . . . . . . . . . . . . . . . . . . . . . 89
Appendix A. Contributors . . . . . . . . . . . . . . . . . . . . 89
Appendix B. Acknowledgments . . . . . . . . . . . . . . . . . . 89
Appendix C. Change Log . . . . . . . . . . . . . . . . . . . . . 90
C.1. Since draft-ietf-quic-transport-07 . . . . . . . . . . . 90
C.2. Since draft-ietf-quic-transport-06 . . . . . . . . . . . 90
C.3. Since draft-ietf-quic-transport-05 . . . . . . . . . . . 90
C.4. Since draft-ietf-quic-transport-04 . . . . . . . . . . . 91
C.5. Since draft-ietf-quic-transport-03 . . . . . . . . . . . 91
C.6. Since draft-ietf-quic-transport-02 . . . . . . . . . . . 91
C.7. Since draft-ietf-quic-transport-01 . . . . . . . . . . . 92
C.8. Since draft-ietf-quic-transport-00 . . . . . . . . . . . 94
C.9. Since draft-hamilton-quic-transport-protocol-01 . . . . . 95
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 95
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1. Introduction
QUIC is a multiplexed and secure transport protocol that runs on top
of UDP. QUIC aims to provide a flexible set of features that allow
it to be a general-purpose transport for multiple applications.
QUIC implements techniques learned from experience with TCP, SCTP and
other transport protocols. QUIC uses UDP as substrate so as to not
require changes to legacy client operating systems and middleboxes to
be deployable. QUIC authenticates all of its headers and encrypts
most of the data it exchanges, including its signaling. This allows
the protocol to evolve without incurring a dependency on upgrades to
middleboxes. This document describes the core QUIC protocol,
including the conceptual design, wire format, and mechanisms of the
QUIC protocol for connection establishment, stream multiplexing,
stream and connection-level flow control, and data reliability.
Accompanying documents describe QUIC's loss detection and congestion
control [QUIC-RECOVERY], and the use of TLS 1.3 for key negotiation
[QUIC-TLS].
2. Conventions and Definitions
The key words "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.
Definitions of terms that are used in this document:
Client: The endpoint initiating a QUIC connection.
Server: The endpoint accepting incoming QUIC connections.
Endpoint: The client or server end of a connection.
Stream: A logical, bi-directional channel of ordered bytes within a
QUIC connection.
Connection: A conversation between two QUIC endpoints with a single
encryption context that multiplexes streams within it.
Connection ID: The 64-bit unsigned number used as an identifier for
a QUIC connection.
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QUIC packet: A well-formed UDP payload that can be parsed by a QUIC
receiver. QUIC packet size in this document refers to the UDP
payload size.
2.1. Notational Conventions
Packet and frame diagrams use the format described in Section 3.1 of
[RFC2360], with the following additional conventions:
[x] Indicates that x is optional
{x} Indicates that x is encrypted
x (A) Indicates that x is A bits long
x (A/B/C) ... Indicates that x is one of A, B, or C bits long
x (i) ... Indicates that x uses the variable-length encoding in
Section 8.1
x (*) ... Indicates that x is variable-length
3. A QUIC Overview
This section briefly describes QUIC's key mechanisms and benefits.
Key strengths of QUIC include:
o Low-latency connection establishment
o Multiplexing without head-of-line blocking
o Authenticated and encrypted header and payload
o Rich signaling for congestion control and loss recovery
o Stream and connection flow control
o Connection migration and resilience to NAT rebinding
o Version negotiation
3.1. Low-Latency Connection Establishment
QUIC relies on a combined cryptographic and transport handshake for
setting up a secure transport connection. QUIC connections are
expected to commonly use 0-RTT handshakes, meaning that for most QUIC
connections, data can be sent immediately following the client
handshake packet, without waiting for a reply from the server. QUIC
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provides a dedicated stream (Stream ID 0) to be used for performing
the cryptographic handshake and QUIC options negotiation. The format
of the QUIC options and parameters used during negotiation are
described in this document, but the handshake protocol that runs on
Stream ID 0 is described in the accompanying cryptographic handshake
draft [QUIC-TLS].
3.2. Stream Multiplexing
When application messages are transported over TCP, independent
application messages can suffer from head-of-line blocking. When an
application multiplexes many streams atop TCP's single-bytestream
abstraction, a loss of a TCP segment results in blocking of all
subsequent segments until a retransmission arrives, irrespective of
the application streams that are encapsulated in subsequent segments.
QUIC ensures that lost packets carrying data for an individual stream
only impact that specific stream. Data received on other streams can
continue to be reassembled and delivered to the application.
3.3. Rich Signaling for Congestion Control and Loss Recovery
QUIC's packet framing and acknowledgments carry rich information that
help both congestion control and loss recovery in fundamental ways.
Each QUIC packet carries a new packet number, including those
carrying retransmitted data. This obviates the need for a separate
mechanism to distinguish acknowledgments for retransmissions from
those for original transmissions, avoiding TCP's retransmission
ambiguity problem. QUIC acknowledgments also explicitly encode the
delay between the receipt of a packet and its acknowledgment being
sent, and together with the monotonically-increasing packet numbers,
this allows for precise network roundtrip-time (RTT) calculation.
QUIC's ACK frames support multiple ACK blocks, so QUIC is more
resilient to reordering than TCP with SACK support, as well as able
to keep more bytes on the wire when there is reordering or loss.
3.4. Stream and Connection Flow Control
QUIC implements stream- and connection-level flow control. At a high
level, a QUIC receiver advertises the maximum amount of data that it
is willing to receive on each stream. As data is sent, received, and
delivered on a particular stream, the receiver sends MAX_STREAM_DATA
frames that increase the advertised limit for that stream, allowing
the peer to send more data on that stream.
In addition to this stream-level flow control, QUIC implements
connection-level flow control to limit the aggregate buffer that a
QUIC receiver is willing to allocate to all streams on a connection.
Connection-level flow control works in the same way as stream-level
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flow control, but the bytes delivered and the limits are aggregated
across all streams.
3.5. Authenticated and Encrypted Header and Payload
TCP headers appear in plaintext on the wire and are not
authenticated, causing a plethora of injection and header
manipulation issues for TCP, such as receive-window manipulation and
sequence-number overwriting. While some of these are mechanisms used
by middleboxes to improve TCP performance, others are active attacks.
Even "performance-enhancing" middleboxes that routinely interpose on
the transport state machine end up limiting the evolvability of the
transport protocol, as has been observed in the design of MPTCP
[RFC6824] and in its subsequent deployability issues.
Generally, QUIC packets are always authenticated and the payload is
typically fully encrypted. The parts of the packet header which are
not encrypted are still authenticated by the receiver, so as to
thwart any packet injection or manipulation by third parties. Some
early handshake packets, such as the Version Negotiation packet, are
not encrypted, but information sent in these unencrypted handshake
packets is later verified as part of cryptographic processing.
3.6. Connection Migration and Resilience to NAT Rebinding
QUIC connections are identified by a Connection ID, a 64-bit unsigned
number randomly generated by the server. QUIC's consistent
connection ID allows connections to survive changes to the client's
IP and port, such as those caused by NAT rebindings or by the client
changing network connectivity to a new address. QUIC provides
automatic cryptographic verification of a rebound client, since the
client continues to use the same session key for encrypting and
decrypting packets. The consistent connection ID can be used to
allow migration of the connection to a new server IP address as well,
since the Connection ID remains consistent across changes in the
client's and the server's network addresses.
3.7. Version Negotiation
QUIC version negotiation allows for multiple versions of the protocol
to be deployed and used concurrently. Version negotiation is
described in Section 7.2.
4. Versions
QUIC versions are identified using a 32-bit unsigned number.
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The version 0x00000000 is reserved to represent version negotiation.
This version of the specification is identified by the number
0x00000001.
Version 0x00000001 of QUIC uses TLS as a cryptographic handshake
protocol, as described in [QUIC-TLS].
Versions with the most significant 16 bits of the version number
cleared are reserved for use in future IETF consensus documents.
Versions that follow the pattern 0x?a?a?a?a are reserved for use in
forcing version negotiation to be exercised. That is, any version
number where the low four bits of all octets is 1010 (in binary). A
client or server MAY advertise support for any of these reserved
versions.
Reserved version numbers will probably never represent a real
protocol; a client MAY use one of these version numbers with the
expectation that the server will initiate version negotiation; a
server MAY advertise support for one of these versions and can expect
that clients ignore the value.
[[RFC editor: please remove the remainder of this section before
publication.]]
The version number for the final version of this specification
(0x00000001), is reserved for the version of the protocol that is
published as an RFC.
Version numbers used to identify IETF drafts are created by adding
the draft number to 0xff000000. For example, draft-ietf-quic-
transport-13 would be identified as 0xff00000D.
Implementors are encouraged to register version numbers of QUIC that
they are using for private experimentation on the github wiki [4].
5. Packet Types and Formats
We first describe QUIC's packet types and their formats, since some
are referenced in subsequent mechanisms.
All numeric values are encoded in network byte order (that is, big-
endian) and all field sizes are in bits. When discussing individual
bits of fields, the least significant bit is referred to as bit 0.
Hexadecimal notation is used for describing the value of fields.
Any QUIC packet has either a long or a short header, as indicated by
the Header Form bit. Long headers are expected to be used early in
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the connection before version negotiation and establishment of 1-RTT
keys. Short headers are minimal version-specific headers, which are
used after version negotiation and 1-RTT keys are established.
5.1. Long Header
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
+-+-+-+-+-+-+-+-+
|1| Type (7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ Connection ID (64) +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Version (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Packet Number (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Payload (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 1: Long Header Format
Long headers are used for packets that are sent prior to the
completion of version negotiation and establishment of 1-RTT keys.
Once both conditions are met, a sender switches to sending packets
using the short header (Section 5.2). The long form allows for
special packets - such as the Version Negotiation packet - to be
represented in this uniform fixed-length packet format. A long
header contains the following fields:
Header Form: The most significant bit (0x80) of octet 0 (the first
octet) is set to 1 for long headers.
Long Packet Type: The remaining seven bits of octet 0 contain the
packet type. This field can indicate one of 128 packet types.
The types specified for this version are listed in Table 1.
Connection ID: Octets 1 through 8 contain the connection ID.
Section 5.6 describes the use of this field in more detail.
Version: Octets 9 to 12 contain the selected protocol version. This
field indicates which version of QUIC is in use and determines how
the rest of the protocol fields are interpreted.
Packet Number: Octets 13 to 16 contain the packet number.
Section 5.7 describes the use of packet numbers.
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Payload: Octets from 17 onwards (the rest of QUIC packet) are the
payload of the packet.
The following packet types are defined:
+------+-----------------+---------------+
| Type | Name | Section |
+------+-----------------+---------------+
| 0x7F | Initial | Section 5.4.1 |
| | | |
| 0x7E | Retry | Section 5.4.2 |
| | | |
| 0x7D | Handshake | Section 5.4.3 |
| | | |
| 0x7C | 0-RTT Protected | Section 5.5 |
+------+-----------------+---------------+
Table 1: Long Header Packet Types
The header form, packet type, connection ID, packet number and
version fields of a long header packet are version-independent. The
types of packets defined in Table 1 are version-specific. See
Section 5.8 for details on how packets from different versions of
QUIC are interpreted.
The interpretation of the fields and the payload are specific to a
version and packet type. Type-specific semantics for this version
are described in the following sections.
5.2. Short Header
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
+-+-+-+-+-+-+-+-+
|0|C|K| Type (5)|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ [Connection ID (64)] +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Packet Number (8/16/32) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Protected Payload (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2: Short Header Format
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The short header can be used after the version and 1-RTT keys are
negotiated. This header form has the following fields:
Header Form: The most significant bit (0x80) of octet 0 is set to 0
for the short header.
Omit Connection ID Flag: The second bit (0x40) of octet 0 indicates
whether the Connection ID field is omitted. If set to 0, then the
Connection ID field is present; if set to 1, the Connection ID
field is omitted. The Connection ID field can only be omitted if
the omit_connection_id transport parameter (Section 7.4.1) is
specified by the intended recipient of the packet.
Key Phase Bit: The third bit (0x20) of octet 0 indicates the key
phase, which allows a recipient of a packet to identify the packet
protection keys that are used to protect the packet. See
[QUIC-TLS] for details.
Short Packet Type: The remaining 5 bits of octet 0 include one of 32
packet types. Table 2 lists the types that are defined for short
packets.
Connection ID: If the Omit Connection ID Flag is not set, a
connection ID occupies octets 1 through 8 of the packet. See
Section 5.6 for more details.
Packet Number: The length of the packet number field depends on the
packet type. This field can be 1, 2 or 4 octets long depending on
the short packet type.
Protected Payload: Packets with a short header always include a
1-RTT protected payload.
The packet type in a short header currently determines only the size
of the packet number field. Additional types can be used to signal
the presence of other fields.
+------+--------------------+
| Type | Packet Number Size |
+------+--------------------+
| 0x1F | 1 octet |
| | |
| 0x1E | 2 octets |
| | |
| 0x1D | 4 octets |
+------+--------------------+
Table 2: Short Header Packet Types
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The header form, omit connection ID flag, and connection ID of a
short header packet are version-independent. The remaining fields
are specific to the selected QUIC version. See Section 5.8 for
details on how packets from different versions of QUIC are
interpreted.
5.3. Version Negotiation Packet
A Version Negotiation packet is inherently not version-specific, and
does not use the packet headers defined above. Upon receipt by a
client, it will appear to be a packet using the long header, but will
be identified as a Version Negotiation packet based on the Version
field.
The Version Negotiation packet is a response to a client packet that
contains a version that is not supported by the server, and is only
sent by servers.
The layout of a Version Negotiation packet is:
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
+-+-+-+-+-+-+-+-+
|1| Unused (7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ Connection ID (64) +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Version (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Supported Version 1 (32) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Supported Version 2 (32)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Supported Version N (32)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 3: Version Negotiation Packet
The value in the Unused field is selected randomly by the server.
The Connection ID field echoes the corresponding value from the
triggering client packet. This allows clients some assurance that
the server received the packet and that the Version Negotiation
packet is in fact from the server. The Version field MUST be set to
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0x00000000. The remainder of the Version Negotiation packet is a
list of 32-bit versions which the server supports.
A Version Negotiation packet cannot be explicitly acknowledged in an
ACK frame by a client. Receiving another Initial packet implicitly
acknowledges a Version Negotiation packet.
See Section 7.2 for a description of the version negotiation process.
5.4. Cryptographic Handshake Packets
Once version negotiation is complete, the cryptographic handshake is
used to agree on cryptographic keys. The cryptographic handshake is
carried in Initial (Section 5.4.1), Retry (Section 5.4.2) and
Handshake (Section 5.4.3) packets.
All these packets use the long header and contain the current QUIC
version in the version field.
In order to prevent tampering by version-unaware middleboxes,
handshake packets are protected with a connection- and version-
specific key, as described in [QUIC-TLS]. This protection does not
provide confidentiality or integrity against on-path attackers, but
provides some level of protection against off-path attackers.
5.4.1. Initial Packet
The Initial packet uses long headers with a type value of 0x7E. It
carries the first cryptographic handshake message sent by the client.
The client populates the connection ID field with randomly selected
values, unless it has received a packet from the server. If the
client has received a packet from the server, the connection ID field
uses the value provided by the server.
The first Initial packet that is sent by a client contains a
randomized packet number. All subsequent packets contain a packet
number that is incremented by one, see (Section 5.7).
The payload of a Initial packet consists of a STREAM frame (or
frames) for stream 0 containing a cryptographic handshake message,
with enough PADDING frames that the packet is at least 1200 octets
(see Section 9). The stream in this packet always starts at an
offset of 0 (see Section 7.5) and the complete cryptographic
handshake message MUST fit in a single packet (see Section 7.3).
The client uses the Initial packet type for any packet that contains
an initial cryptographic handshake message. This includes all cases
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where a new packet containing the initial cryptographic message needs
to be created, this includes the packets sent after receiving a
Version Negotiation (Section 5.3) or Retry packet (Section 5.4.2).
5.4.2. Retry Packet
A Retry packet uses long headers with a type value of 0x7D. It
carries cryptographic handshake messages and acknowledgments. It is
used by a server that wishes to perform a stateless retry (see
Section 7.5).
The packet number and connection ID fields echo the corresponding
fields from the triggering client packet. This allows a client to
verify that the server received its packet.
A Retry packet is never explicitly acknowledged in an ACK frame by a
client. Receiving another Initial packet implicitly acknowledges a
Retry packet.
After receiving a Retry packet, the client uses a new Initial packet
containing the next cryptographic handshake message. The client
retains the state of its cryptographic handshake, but discards all
transport state. The Initial packet that is generated in response to
a Retry packet includes STREAM frames on stream 0 that start again at
an offset of 0.
Continuing the cryptographic handshake is necessary to ensure that an
attacker cannot force a downgrade of any cryptographic parameters.
In addition to continuing the cryptographic handshake, the client
MUST remember the results of any version negotiation that occurred
(see Section 7.2). The client MAY also retain any observed RTT or
congestion state that it has accumulated for the flow, but other
transport state MUST be discarded.
The payload of the Retry packet contains a single STREAM frame on
stream 0 with offset 0 containing the server's cryptographic
stateless retry material. It MUST NOT contain any other frames. The
next STREAM frame sent by the server will also start at stream offset
0.
5.4.3. Handshake Packet
A Handshake packet uses long headers with a type value of 0x7C. It
is used to carry acknowledgments and cryptographic handshake messages
from the server and client.
The connection ID field in a Handshake packet contains a connection
ID that is chosen by the server (see Section 5.6).
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The first Handshake packet sent by a server contains a randomized
packet number. This value is increased for each subsequent packet
sent by the server as described in Section 5.7. The client
increments the packet number from its previous packet by one for each
Handshake packet that it sends (which might be an Initial, 0-RTT
Protected, or Handshake packet).
The payload of this packet contains STREAM frames and could contain
PADDING and ACK frames.
5.5. Protected Packets
Packets that are protected with 0-RTT keys are sent with long
headers; all packets protected with 1-RTT keys are sent with short
headers. The different packet types explicitly indicate the
encryption level and therefore the keys that are used to remove
packet protection.
Packets protected with 0-RTT keys use a type value of 0x7B. The
connection ID field for a 0-RTT packet is selected by the client.
The client can send 0-RTT packets after receiving a Handshake packet
(Section 5.4.3), if that packet does not complete the handshake.
Even if the client receives a different connection ID in the
Handshake packet, it MUST continue to use the connection ID selected
by the client for 0-RTT packets, see Section 5.6.
The version field for protected packets is the current QUIC version.
The packet number field contains a packet number, which increases
with each packet sent, see Section 5.7 for details.
The payload is protected using authenticated encryption. [QUIC-TLS]
describes packet protection in detail. After decryption, the
plaintext consists of a sequence of frames, as described in
Section 6.
5.6. Connection ID
QUIC connections are identified by their 64-bit Connection ID. All
long headers contain a Connection ID. Short headers indicate the
presence of a Connection ID using the Omit Connection ID flag. When
present, the Connection ID is in the same location in all packet
headers, making it straightforward for middleboxes, such as load
balancers, to locate and use it.
The client MUST choose a random connection ID and use it in Initial
packets (Section 5.4.1) and 0-RTT packets (Section 5.5).
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When the server receives a Initial packet and decides to proceed with
the handshake, it chooses a new value for the connection ID and sends
that in a Handshake packet (Section 5.4.3). The server MAY choose to
use the value that the client initially selects.
Once the client receives the connection ID that the server has
chosen, it MUST use it for all subsequent Handshake (Section 5.4.3)
and 1-RTT (Section 5.5) packets but not for 0-RTT packets
(Section 5.5).
Server's Version Negotiation (Section 5.3) and Retry (Section 5.4.2)
packets MUST use connection ID selected by the client.
5.7. Packet Numbers
The packet number is an integer in the range 0 to 2^62-1. The value
is used in determining the cryptographic nonce for packet encryption.
Each endpoint maintains a separate packet number for sending and
receiving. The packet number for sending MUST increase by at least
one after sending any packet, unless otherwise specified (see
Section 5.7.1).
A QUIC endpoint MUST NOT reuse a packet number within the same
connection (that is, under the same cryptographic keys). If the
packet number for sending reaches 2^62 - 1, the sender MUST close the
connection without sending a CONNECTION_CLOSE frame or any further
packets; a server MAY send a Stateless Reset (Section 7.9.4) in
response to further packets that it receives.
For the packet header, the number of bits required to represent the
packet number are reduced by including only the least significant
bits of the packet number. The actual packet number for each packet
is reconstructed at the receiver based on the largest packet number
received on a successfully authenticated packet.
A packet number is decoded by finding the packet number value that is
closest to the next expected packet. The next expected packet is the
highest received packet number plus one. For example, if the highest
successfully authenticated packet had a packet number of 0xaa82f30e,
then a packet containing a 16-bit value of 0x1f94 will be decoded as
0xaa831f94.
The sender MUST use a packet number size able to represent more than
twice as large a range than the difference between the largest
acknowledged packet and packet number being sent. A peer receiving
the packet will then correctly decode the packet number, unless the
packet is delayed in transit such that it arrives after many higher-
numbered packets have been received. An endpoint SHOULD use a large
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enough packet number encoding to allow the packet number to be
recovered even if the packet arrives after packets that are sent
afterwards.
As a result, the size of the packet number encoding is at least one
more than the base 2 logarithm of the number of contiguous
unacknowledged packet numbers, including the new packet.
For example, if an endpoint has received an acknowledgment for packet
0x6afa2f, sending a packet with a number of 0x6b4264 requires a
16-bit or larger packet number encoding; whereas a 32-bit packet
number is needed to send a packet with a number of 0x6bc107.
Version Negotiation (Section 5.3) and Retry (Section 5.4.2) packets
have special rules for populating the packet number field.
5.7.1. Initial Packet Number
The initial value for packet number MUST be selected randomly from a
range between 0 and 2^32 - 1025 (inclusive). This value is selected
so that Initial and Handshake packets exercise as many possible
values for the Packet Number field as possible.
Limiting the range allows both for loss of packets and for any
stateless exchanges. Packet numbers are incremented for subsequent
packets, but packet loss and stateless handling can both mean that
the first packet sent by an endpoint isn't necessarily the first
packet received by its peer. The first packet received by a peer
cannot be 2^32 or greater or the recipient will incorrectly assume a
packet number that is 2^32 values lower and discard the packet.
Use of a secure random number generator [RFC4086] is not necessary
for generating the initial packet number, nor is it necessary that
the value be uniformly distributed.
5.8. Handling Packets from Different Versions
Between different versions the following things are guaranteed to
remain constant:
o the location of the header form flag,
o the location of the Omit Connection ID flag in short headers,
o the location and size of the Connection ID field in both header
forms,
o the location and size of the Version field in long headers,
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o the format and semantics of the Version Negotiation packet.
Implementations MUST assume that an unsupported version uses an
unknown packet format. All other fields MUST be ignored when
processing a packet that contains an unsupported version.
6. Frames and Frame Types
The payload of all packets, after removing packet protection,
consists of a sequence of frames, as shown in Figure 4. Version
Negotiation and Stateless Reset do not contain frames.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Frame 1 (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Frame 2 (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Frame N (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 4: Contents of Protected Payload
Protected payloads MUST contain at least one frame, and MAY contain
multiple frames and multiple frame types.
Frames MUST fit within a single QUIC packet and MUST NOT span a QUIC
packet boundary. Each frame begins with a Frame Type byte,
indicating its type, followed by additional type-dependent fields:
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Type (8) | Type-Dependent Fields (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 5: Generic Frame Layout
Frame types are listed in Table 3. Note that the Frame Type byte in
STREAM and ACK frames is used to carry other frame-specific flags.
For all other frames, the Frame Type byte simply identifies the
frame. These frames are explained in more detail as they are
referenced later in the document.
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+-------------+-------------------+--------------+
| Type Value | Frame Type Name | Definition |
+-------------+-------------------+--------------+
| 0x00 | PADDING | Section 8.2 |
| | | |
| 0x01 | RST_STREAM | Section 8.3 |
| | | |
| 0x02 | CONNECTION_CLOSE | Section 8.4 |
| | | |
| 0x03 | APPLICATION_CLOSE | Section 8.5 |
| | | |
| 0x04 | MAX_DATA | Section 8.6 |
| | | |
| 0x05 | MAX_STREAM_DATA | Section 8.7 |
| | | |
| 0x06 | MAX_STREAM_ID | Section 8.8 |
| | | |
| 0x07 | PING | Section 8.9 |
| | | |
| 0x08 | BLOCKED | Section 8.10 |
| | | |
| 0x09 | STREAM_BLOCKED | Section 8.11 |
| | | |
| 0x0a | STREAM_ID_BLOCKED | Section 8.12 |
| | | |
| 0x0b | NEW_CONNECTION_ID | Section 8.13 |
| | | |
| 0x0c | STOP_SENDING | Section 8.14 |
| | | |
| 0x0d | PONG | Section 8.15 |
| | | |
| 0x0e | ACK | Section 8.16 |
| | | |
| 0x10 - 0x17 | STREAM | Section 8.17 |
+-------------+-------------------+--------------+
Table 3: Frame Types
7. Life of a Connection
A QUIC connection is a single conversation between two QUIC
endpoints. QUIC's connection establishment intertwines version
negotiation with the cryptographic and transport handshakes to reduce
connection establishment latency, as described in Section 7.3. Once
established, a connection may migrate to a different IP or port at
either endpoint, due to NAT rebinding or mobility, as described in
Section 7.7. Finally a connection may be terminated by either
endpoint, as described in Section 7.9.
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7.1. Matching Packets to Connections
Incoming packets are classified on receipt. Packets can either be
associated with an existing connection, be discarded, or - for
servers - potentially create a new connection.
Packets that can be associated with an existing connection are
handled according to the current state of that connection. Packets
are associated with existing connections using connection ID if it is
present; this might include connection IDs that were advertised using
NEW_CONNECTION_ID (Section 8.13). Packets without connection IDs and
long-form packets for connections that have incomplete cryptographic
handshakes are associated with an existing connection using the tuple
of source and destination IP addresses and ports.
A packet that uses the short header could be associated with an
existing connection with an incomplete cryptographic handshake. Such
a packet could be a valid packet that has been reordered with respect
to the long-form packets that will complete the cryptographic
handshake. This might happen after the final set of cryptographic
handshake messages from either peer. These packets are expected to
be correlated with a connection using the tuple of IP addresses and
ports. Packets that might be reordered in this fashion SHOULD be
buffered in anticipation of the handshake completing.
0-RTT packets might be received prior to a Client Initial packet at a
server. If the version of these packets is acceptable to the server,
it MAY buffer these packets in anticipation of receiving a reordered
Client Initial packet.
Buffering ensures that data is not lost, which improves performance;
conversely, discarding these packets could create false loss signals
for the congestion controllers. However, limiting the number and
size of buffered packets might be needed to prevent exposure to
denial of service.
For clients, any packet that cannot be associated with an existing
connection SHOULD be discarded if it is not buffered. Discarded
packets MAY be logged for diagnostic or security purposes.
For servers, packets that aren't associated with a connection
potentially create a new connection. However, only packets that use
the long packet header and that are at least the minimum size defined
for the protocol version can be initial packets. A server MAY
discard packets with a short header or packets that are smaller than
the smallest minimum size for any version that the server supports.
A server that discards a packet that cannot be associated with a
connection MAY also generate a stateless reset (Section 7.9.4).
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This version of QUIC defines a minimum size for initial packets of
1200 octets (see Section 9). Versions of QUIC that define smaller
minimum initial packet sizes need to be aware that initial packets
will be discarded without action by servers that only support
versions with larger minimums. Clients that support multiple QUIC
versions can avoid this problem by ensuring that they increase the
size of their initial packets to the largest minimum size across all
of the QUIC versions they support. Servers need to recognize initial
packets that are the minimum size of all QUIC versions they support.
7.2. Version Negotiation
QUIC's connection establishment begins with version negotiation,
since all communication between the endpoints, including packet and
frame formats, relies on the two endpoints agreeing on a version.
A QUIC connection begins with a client sending an Initial packet
(Section 5.4.1). The details of the handshake mechanisms are
described in Section 7.3, but any Initial packet sent from the client
to the server MUST use the long header format - which includes the
version of the protocol being used - and they MUST be padded to at
least 1200 octets.
The server receives this packet and determines whether it potentially
creates a new connection (see Section 7.1). If the packet might
generate a new connection, the server then checks whether it
understands the version that the client has selected.
If the packet contains a version that is acceptable to the server,
the server proceeds with the handshake (Section 7.3). This commits
the server to the version that the client selected.
7.2.1. Sending Version Negotiation Packets
If the version selected by the client is not acceptable to the
server, the server responds with a Version Negotiation packet
(Section 5.3). This includes a list of versions that the server will
accept.
A server sends a Version Negotiation packet for any packet with an
unacceptable version if that packet could create a new connection.
This allows a server to process packets with unsupported versions
without retaining state. Though either the Client Initial packet or
the version negotiation packet that is sent in response could be
lost, the client will send new packets until it successfully receives
a response or it abandons the connection attempt.
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7.2.2. Handling Version Negotiation Packets
When the client receives a Version Negotiation packet, it first
checks that the connection ID matches the connection ID the client
sent. If this check fails, the packet MUST be discarded.
Once the Version Negotiation packet is determined to be valid, the
client then selects an acceptable protocol version from the list
provided by the server. The client then attempts to create a
connection using that version. Though the contents of the Client
Initial packet the client sends might not change in response to
version negotiation, a client MUST increase the packet number it uses
on every packet it sends. Packets MUST continue to use long headers
and MUST include the new negotiated protocol version.
The client MUST use the long header format and include its selected
version on all packets until it has 1-RTT keys and it has received a
packet from the server which is not a Version Negotiation packet.
A client MUST NOT change the version it uses unless it is in response
to a Version Negotiation packet from the server. Once a client
receives a packet from the server which is not a Version Negotiation
packet, it MUST discard other Version Negotiation packets on the same
connection. Similarly, a client MUST ignore a Version Negotiation
packet if it has already received and acted on a Version Negotiation
packet.
A client MUST ignore a Version Negotiation packet that lists the
client's chosen version.
Version negotiation packets have no cryptographic protection. The
result of the negotiation MUST be revalidated as part of the
cryptographic handshake (see Section 7.4.4).
7.2.3. Using Reserved Versions
For a server to use a new version in the future, clients must
correctly handle unsupported versions. To help ensure this, a server
SHOULD include a reserved version (see Section 4) while generating a
Version Negotiation packet.
The design of version negotiation permits a server to avoid
maintaining state for packets that it rejects in this fashion. The
validation of version negotiation (see Section 7.4.4) only validates
the result of version negotiation, which is the same no matter which
reserved version was sent. A server MAY therefore send different
reserved version numbers in the Version Negotiation Packet and in its
transport parameters.
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A client MAY send a packet using a reserved version number. This can
be used to solicit a list of supported versions from a server.
7.3. Cryptographic and Transport Handshake
QUIC relies on a combined cryptographic and transport handshake to
minimize connection establishment latency. QUIC allocates stream 0
for the cryptographic handshake. Version 0x00000001 of QUIC uses TLS
1.3 as described in [QUIC-TLS]; a different QUIC version number could
indicate that a different cryptographic handshake protocol is in use.
QUIC provides this stream with reliable, ordered delivery of data.
In return, the cryptographic handshake provides QUIC with:
o authenticated key exchange, where
* a server is always authenticated,
* a client is optionally authenticated,
* every connection produces distinct and unrelated keys,
* keying material is usable for packet protection for both 0-RTT
and 1-RTT packets, and
* 1-RTT keys have forward secrecy
o authenticated values for the transport parameters of the peer (see
Section 7.4)
o authenticated confirmation of version negotiation (see
Section 7.4.4)
o authenticated negotiation of an application protocol (TLS uses
ALPN [RFC7301] for this purpose)
o for the server, the ability to carry data that provides assurance
that the client can receive packets that are addressed with the
transport address that is claimed by the client (see Section 7.6)
The initial cryptographic handshake message MUST be sent in a single
packet. Any second attempt that is triggered by address validation
MUST also be sent within a single packet. This avoids having to
reassemble a message from multiple packets. Reassembling messages
requires that a server maintain state prior to establishing a
connection, exposing the server to a denial of service risk.
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The first client packet of the cryptographic handshake protocol MUST
fit within a 1232 octet QUIC packet payload. This includes overheads
that reduce the space available to the cryptographic handshake
protocol.
Details of how TLS is integrated with QUIC is provided in more detail
in [QUIC-TLS].
7.4. Transport Parameters
During connection establishment, both endpoints make authenticated
declarations of their transport parameters. These declarations are
made unilaterally by each endpoint. Endpoints are required to comply
with the restrictions implied by these parameters; the description of
each parameter includes rules for its handling.
The format of the transport parameters is the TransportParameters
struct from Figure 6. This is described using the presentation
language from Section 3 of [I-D.ietf-tls-tls13].
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uint32 QuicVersion;
enum {
initial_max_stream_data(0),
initial_max_data(1),
initial_max_stream_id_bidi(2),
idle_timeout(3),
omit_connection_id(4),
max_packet_size(5),
stateless_reset_token(6),
ack_delay_exponent(7),
initial_max_stream_id_uni(8),
(65535)
} TransportParameterId;
struct {
TransportParameterId parameter;
opaque value<0..2^16-1>;
} TransportParameter;
struct {
select (Handshake.msg_type) {
case client_hello:
QuicVersion initial_version;
case encrypted_extensions:
QuicVersion negotiated_version;
QuicVersion supported_versions<4..2^8-4>;
case new_session_ticket:
struct {};
};
TransportParameter parameters<30..2^16-1>;
} TransportParameters;
Figure 6: Definition of TransportParameters
The "extension_data" field of the quic_transport_parameters extension
defined in [QUIC-TLS] contains a TransportParameters value. TLS
encoding rules are therefore used to encode the transport parameters.
QUIC encodes transport parameters into a sequence of octets, which
are then included in the cryptographic handshake. Once the handshake
completes, the transport parameters declared by the peer are
available. Each endpoint validates the value provided by its peer.
In particular, version negotiation MUST be validated (see
Section 7.4.4) before the connection establishment is considered
properly complete.
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Definitions for each of the defined transport parameters are included
in Section 7.4.1. Any given parameter MUST appear at most once in a
given transport parameters extension. An endpoint MUST treat receipt
of duplicate transport parameters as a connection error of type
TRANSPORT_PARAMETER_ERROR.
7.4.1. Transport Parameter Definitions
An endpoint MUST include the following parameters in its encoded
TransportParameters:
initial_max_stream_data (0x0000): The initial stream maximum data
parameter contains the initial value for the maximum data that can
be sent on any newly created stream. This parameter is encoded as
an unsigned 32-bit integer in units of octets. This is equivalent
to an implicit MAX_STREAM_DATA frame (Section 8.7) being sent on
all streams immediately after opening.
initial_max_data (0x0001): The initial maximum data parameter
contains the initial value for the maximum amount of data that can
be sent on the connection. This parameter is encoded as an
unsigned 32-bit integer in units of octets. This is equivalent to
sending a MAX_DATA (Section 8.6) for the connection immediately
after completing the handshake.
idle_timeout (0x0003): The idle timeout is a value in seconds that
is encoded as an unsigned 16-bit integer. The maximum value is
600 seconds (10 minutes).
A server MUST include the following transport parameters:
stateless_reset_token (0x0006): The Stateless Reset Token is used in
verifying a stateless reset, see Section 7.9.4. This parameter is
a sequence of 16 octets.
A client MUST NOT include a stateless reset token. A server MUST
treat receipt of a stateless_reset_token transport parameter as a
connection error of type TRANSPORT_PARAMETER_ERROR.
An endpoint MAY use the following transport parameters:
initial_max_stream_id_bidi (0x0002): The initial maximum stream ID
parameter contains the initial maximum stream number the peer may
initiate for bidirectional streams, encoded as an unsigned 32-bit
integer. This value MUST be a valid bidirectional stream ID for a
peer-initiated stream (that is, the two least significant bits are
set to 0 by a server and to 1 by a client). If an invalid value
is provided, the recipient MUST generate a connection error of
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type TRANSPORT_PARAMETER_ERROR. Setting this parameter is
equivalent to sending a MAX_STREAM_ID (Section 8.8) immediately
after completing the handshake. The maximum bidirectional stream
ID is set to 0 if this parameter is absent, preventing the
creation of new bidirectional streams until a MAX_STREAM_ID frame
is sent. Note that a default value of 0 does not prevent the
cryptographic handshake stream (that is, stream 0) from being
used.
initial_max_stream_id_uni (0x0008): The initial maximum stream ID
parameter contains the initial maximum stream number the peer may
initiate for unidirectional streams, encoded as an unsigned 32-bit
integer. The value MUST be a valid unidirectional ID for the
recipient (that is, the two least significant bits are set to 2 by
a server and to 3 by a client). If an invalid value is provided,
the recipient MUST generate a connection error of type
TRANSPORT_PARAMETER_ERROR. Setting this parameter is equivalent
to sending a MAX_STREAM_ID (Section 8.8) immediately after
completing the handshake. The maximum unidirectional stream ID is
set to 0 if this parameter is absent, preventing the creation of
new unidirectional streams until a MAX_STREAM_ID frame is sent.
omit_connection_id (0x0004): The omit connection identifier
parameter indicates that packets sent to the endpoint that
advertises this parameter MAY omit the connection ID in packets
using short header format. This can be used by an endpoint where
it knows that source and destination IP address and port are
sufficient for it to identify a connection. This parameter is
zero length. Absence of this parameter means that the connection
ID MUST be present in every packet sent to this endpoint.
max_packet_size (0x0005): The maximum packet size parameter places a
limit on the size of packets that the endpoint is willing to
receive, encoded as an unsigned 16-bit integer. This indicates
that packets larger than this limit will be dropped. The default
for this parameter is the maximum permitted UDP payload of 65527.
Values below 1200 are invalid. This limit only applies to
protected packets (Section 5.5).
ack_delay_exponent (0x0007): An 8-bit unsigned integer value
indicating an exponent used to decode the ACK Delay field in the
ACK frame, see Section 8.16. If this value is absent, a default
value of 3 is assumed (indicating a multiplier of 8). Values
above 20 are invalid.
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7.4.2. Values of Transport Parameters for 0-RTT
Transport parameters from the server MUST be remembered by the client
for use with 0-RTT data. If the TLS NewSessionTicket message
includes the quic_transport_parameters extension, then those values
are used for the server values when establishing a new connection
using that ticket. Otherwise, the transport parameters that the
server advertises during connection establishment are used.
A server can remember the transport parameters that it advertised, or
store an integrity-protected copy of the values in the ticket and
recover the information when accepting 0-RTT data. A server uses the
transport parameters in determining whether to accept 0-RTT data.
A server MAY accept 0-RTT and subsequently provide different values
for transport parameters for use in the new connection. If 0-RTT
data is accepted by the server, the server MUST NOT reduce any limits
or alter any values that might be violated by the client with its
0-RTT data. In particular, a server that accepts 0-RTT data MUST NOT
set values for initial_max_data or initial_max_stream_data that are
smaller than the remembered value of those parameters. Similarly, a
server MUST NOT reduce the value of initial_max_stream_id_bidi or
initial_max_stream_id_uni.
Omitting or setting a zero value for certain transport parameters can
result in 0-RTT data being enabled, but not usable. The following
transport parameters SHOULD be set to non-zero values for 0-RTT:
initial_max_stream_id_bidi, initial_max_stream_id_uni,
initial_max_data, initial_max_stream_data.
A server MUST reject 0-RTT data or even abort a handshake if the
implied values for transport parameters cannot be supported.
7.4.3. New Transport Parameters
New transport parameters can be used to negotiate new protocol
behavior. An endpoint MUST ignore transport parameters that it does
not support. Absence of a transport parameter therefore disables any
optional protocol feature that is negotiated using the parameter.
New transport parameters can be registered according to the rules in
Section 14.1.
7.4.4. Version Negotiation Validation
Though the cryptographic handshake has integrity protection, two
forms of QUIC version downgrade are possible. In the first, an
attacker replaces the QUIC version in the Initial packet. In the
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second, a fake Version Negotiation packet is sent by an attacker. To
protect against these attacks, the transport parameters include three
fields that encode version information. These parameters are used to
retroactively authenticate the choice of version (see Section 7.2).
The cryptographic handshake provides integrity protection for the
negotiated version as part of the transport parameters (see
Section 7.4). As a result, attacks on version negotiation by an
attacker can be detected.
The client includes the initial_version field in its transport
parameters. The initial_version is the version that the client
initially attempted to use. If the server did not send a version
negotiation packet Section 5.3, this will be identical to the
negotiated_version field in the server transport parameters.
A server that processes all packets in a stateful fashion can
remember how version negotiation was performed and validate the
initial_version value.
A server that does not maintain state for every packet it receives
(i.e., a stateless server) uses a different process. If the
initial_version matches the version of QUIC that is in use, a
stateless server can accept the value.
If the initial_version is different from the version of QUIC that is
in use, a stateless server MUST check that it would have sent a
version negotiation packet if it had received a packet with the
indicated initial_version. If a server would have accepted the
version included in the initial_version and the value differs from
the QUIC version that is in use, the server MUST terminate the
connection with a VERSION_NEGOTIATION_ERROR error.
The server includes both the version of QUIC that is in use and a
list of the QUIC versions that the server supports.
The negotiated_version field is the version that is in use. This
MUST be set by the server to the value that is on the Initial packet
that it accepts (not an Initial packet that triggers a Retry or
Version Negotiation packet). A client that receives a
negotiated_version that does not match the version of QUIC that is in
use MUST terminate the connection with a VERSION_NEGOTIATION_ERROR
error code.
The server includes a list of versions that it would send in any
version negotiation packet (Section 5.3) in the supported_versions
field. The server populates this field even if it did not send a
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version negotiation packet. This field is absent if the parameters
are included in a NewSessionTicket message.
The client validates that the negotiated_version is included in the
supported_versions list and - if version negotiation was performed -
that it would have selected the negotiated version. A client MUST
terminate the connection with a VERSION_NEGOTIATION_ERROR error code
if the current QUIC version is not listed in the supported_versions
list. A client MUST terminate with a VERSION_NEGOTIATION_ERROR error
code if version negotiation occurred but it would have selected a
different version based on the value of the supported_versions list.
When an endpoint accepts multiple QUIC versions, it can potentially
interpret transport parameters as they are defined by any of the QUIC
versions it supports. The version field in the QUIC packet header is
authenticated using transport parameters. The position and the
format of the version fields in transport parameters MUST either be
identical across different QUIC versions, or be unambiguously
different to ensure no confusion about their interpretation. One way
that a new format could be introduced is to define a TLS extension
with a different codepoint.
7.5. Stateless Retries
A server can process an initial cryptographic handshake messages from
a client without committing any state. This allows a server to
perform address validation (Section 7.6, or to defer connection
establishment costs.
A server that generates a response to an initial packet without
retaining connection state MUST use the Retry packet (Section 5.4.2).
This packet causes a client to reset its transport state and to
continue the connection attempt with new connection state while
maintaining the state of the cryptographic handshake.
A server MUST NOT send multiple Retry packets in response to a client
handshake packet. Thus, any cryptographic handshake message that is
sent MUST fit within a single packet.
In TLS, the Retry packet type is used to carry the HelloRetryRequest
message.
7.6. Proof of Source Address Ownership
Transport protocols commonly spend a round trip checking that a
client owns the transport address (IP and port) that it claims.
Verifying that a client can receive packets sent to its claimed
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transport address protects against spoofing of this information by
malicious clients.
This technique is used primarily to avoid QUIC from being used for
traffic amplification attack. In such an attack, a packet is sent to
a server with spoofed source address information that identifies a
victim. If a server generates more or larger packets in response to
that packet, the attacker can use the server to send more data toward
the victim than it would be able to send on its own.
Several methods are used in QUIC to mitigate this attack. Firstly,
the initial handshake packet is padded to at least 1200 octets. This
allows a server to send a similar amount of data without risking
causing an amplification attack toward an unproven remote address.
A server eventually confirms that a client has received its messages
when the cryptographic handshake successfully completes. This might
be insufficient, either because the server wishes to avoid the
computational cost of completing the handshake, or it might be that
the size of the packets that are sent during the handshake is too
large. This is especially important for 0-RTT, where the server
might wish to provide application data traffic - such as a response
to a request - in response to the data carried in the early data from
the client.
To send additional data prior to completing the cryptographic
handshake, the server then needs to validate that the client owns the
address that it claims.
Source address validation is therefore performed during the
establishment of a connection. TLS provides the tools that support
the feature, but basic validation is performed by the core transport
protocol.
A different type of source address validation is performed after a
connection migration, see Section 7.7.2.
7.6.1. Client Address Validation Procedure
QUIC uses token-based address validation. Any time the server wishes
to validate a client address, it provides the client with a token.
As long as the token cannot be easily guessed (see Section 7.6.3), if
the client is able to return that token, it proves to the server that
it received the token.
During the processing of the cryptographic handshake messages from a
client, TLS will request that QUIC make a decision about whether to
proceed based on the information it has. TLS will provide QUIC with
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any token that was provided by the client. For an initial packet,
QUIC can decide to abort the connection, allow it to proceed, or
request address validation.
If QUIC decides to request address validation, it provides the
cryptographic handshake with a token. The contents of this token are
consumed by the server that generates the token, so there is no need
for a single well-defined format. A token could include information
about the claimed client address (IP and port), a timestamp, and any
other supplementary information the server will need to validate the
token in the future.
The cryptographic handshake is responsible for enacting validation by
sending the address validation token to the client. A legitimate
client will include a copy of the token when it attempts to continue
the handshake. The cryptographic handshake extracts the token then
asks QUIC a second time whether the token is acceptable. In
response, QUIC can either abort the connection or permit it to
proceed.
A connection MAY be accepted without address validation - or with
only limited validation - but a server SHOULD limit the data it sends
toward an unvalidated address. Successful completion of the
cryptographic handshake implicitly provides proof that the client has
received packets from the server.
7.6.2. Address Validation on Session Resumption
A server MAY provide clients with an address validation token during
one connection that can be used on a subsequent connection. Address
validation is especially important with 0-RTT because a server
potentially sends a significant amount of data to a client in
response to 0-RTT data.
A different type of token is needed when resuming. Unlike the token
that is created during a handshake, there might be some time between
when the token is created and when the token is subsequently used.
Thus, a resumption token SHOULD include an expiration time. It is
also unlikely that the client port number is the same on two
different connections; validating the port is therefore unlikely to
be successful.
This token can be provided to the cryptographic handshake immediately
after establishing a connection. QUIC might also generate an updated
token if significant time passes or the client address changes for
any reason (see Section 7.7). The cryptographic handshake is
responsible for providing the client with the token. In TLS the
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token is included in the ticket that is used for resumption and
0-RTT, which is carried in a NewSessionTicket message.
7.6.3. Address Validation Token Integrity
An address validation token MUST be difficult to guess. Including a
large enough random value in the token would be sufficient, but this
depends on the server remembering the value it sends to clients.
A token-based scheme allows the server to offload any state
associated with validation to the client. For this design to work,
the token MUST be covered by integrity protection against
modification or falsification by clients. Without integrity
protection, malicious clients could generate or guess values for
tokens that would be accepted by the server. Only the server
requires access to the integrity protection key for tokens.
In TLS the address validation token is often bundled with the
information that TLS requires, such as the resumption secret. In
this case, adding integrity protection can be delegated to the
cryptographic handshake protocol, avoiding redundant protection. If
integrity protection is delegated to the cryptographic handshake, an
integrity failure will result in immediate cryptographic handshake
failure. If integrity protection is performed by QUIC, QUIC MUST
abort the connection if the integrity check fails with a
PROTOCOL_VIOLATION error code.
7.7. Connection Migration
QUIC connections are identified by their 64-bit Connection ID.
QUIC's consistent connection ID allows connections to survive changes
to the client's IP and/or port, such as those caused by client or
server migrating to a new network. Connection migration allows a
client to retain any shared state with a connection when they move
networks. This includes state that can be hard to recover such as
outstanding requests, which might otherwise be lost with no easy way
to retry them.
An endpoint that receives packets that contain a source IP address
and port that has not yet been used can start sending new packets
with those as a destination IP address and port. Packets exchanged
between endpoints can then follow the new path.
Due to variations in path latency or packet reordering, packets from
different source addresses might be reordered. The packet with the
highest packet number MUST be used to determine which path to use.
Endpoints also need to be prepared to receive packets from an older
source address.
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An endpoint MUST validate that its peer can receive packets at the
new address before sending any significant quantity of data to that
address, or it risks being used for denial of service. See
Section 7.7.2 for details.
7.7.1. Privacy Implications of Connection Migration
Using a stable connection ID on multiple network paths allows a
passive observer to correlate activity between those paths. A client
that moves between networks might not wish to have their activity
correlated by any entity other than a server. The NEW_CONNECTION_ID
message can be sent by a server to provide an unlinkable connection
ID for use in case the client wishes to explicitly break linkability
between two points of network attachment.
A client might need to send packets on multiple networks without
receiving any response from the server. To ensure that the client is
not linkable across each of these changes, a new connection ID and
packet number gap are needed for each network. To support this, a
server sends multiple NEW_CONNECTION_ID messages. Each
NEW_CONNECTION_ID is marked with a sequence number. Connection IDs
MUST be used in the order in which they are numbered.
A client which wishes to break linkability upon changing networks
MUST use the connection ID provided by the server as well as
incrementing the packet sequence number by an externally
unpredictable value computed as described in Section 7.7.1.1. Packet
number gaps are cumulative. A client might skip connection IDs, but
it MUST ensure that it applies the associated packet number gaps for
connection IDs that it skips in addition to the packet number gap
associated with the connection ID that it does use.
A server that receives a packet that is marked with a new connection
ID recovers the packet number by adding the cumulative packet number
gap to its expected packet number. A server SHOULD discard packets
that contain a smaller gap than it advertised.
For instance, a server might provide a packet number gap of 7
associated with a new connection ID. If the server received packet
10 using the previous connection ID, it should expect packets on the
new connection ID to start at 18. A packet with the new connection
ID and a packet number of 17 is discarded as being in error.
7.7.1.1. Packet Number Gap
In order to avoid linkage, the packet number gap MUST be externally
indistinguishable from random. The packet number gap for a
connection ID with sequence number is computed by encoding the
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sequence number as a 32-bit integer in big-endian format, and then
computing:
Gap = HKDF-Expand-Label(packet_number_secret,
"QUIC packet sequence gap", sequence, 4)
The output of HKDF-Expand-Label is interpreted as a big-endian
number. "packet_number_secret" is derived from the TLS key exchange,
as described in Section 5.6 of [QUIC-TLS].
7.7.2. Address Validation for Migrated Connections
An endpoint that receives a packet from a new remote IP address and
port (or just a new remote port) on packets from its peer is likely
seeing a connection migration at the peer.
However, it is also possible that the peer is spoofing its source
address in order to cause the endpoint to send excessive amounts of
data to an unwilling host. If the endpoint sends significantly more
data than the peer, connection migration might be used to amplify the
volume of data that an attacker can generate toward a victim.
Thus, when seeing a new remote transport address, an endpoint MUST
verify that its peer can receive and respond to packets at that new
address. By providing copies of the data that it receives, the peer
proves that it is receiving packets at the new address and consents
to receive data.
Prior to validating the new remote address, and endpoint MUST limit
the amount of data and packets that it sends to its peer. At a
minimum, this needs to consider the possibility that packets are sent
without congestion feedback.
Once a connection is established, address validation is relatively
simple (see Section 7.6 for the process that is used during the
handshake). An endpoint validates a remote address by sending a PING
frame containing a payload that is hard to guess. This frame MUST be
sent in a packet that is sent to the new address. Once a PONG frame
containing the same payload is received, the address is considered to
be valid. The PONG frame can use any path on its return. A PING
frame containing 12 randomly generated [RFC4086] octets is sufficient
to ensure that it is easier to receive the packet than it is to guess
the value correctly.
If the PING frame is determined to be lost, a new PING frame SHOULD
be generated. This PING frame MUST include a new Data field that is
similarly difficult to guess.
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If validation of the new remote address fails, after allowing enough
time for possible loss and recovery of packets carrying PING and PONG
frames, the endpoint MUST terminate the connection. When setting
this timer, implementations are cautioned that the new path could
have a longer round trip time than the original. The endpoint MUST
NOT send a CONNECTION_CLOSE frame in this case; it has to assume that
the remote peer does not want to receive any more packets.
If the remote address is validated successfully, the endpoint MAY
increase the rate that it sends on the new path using the state from
the previous path. The capacity available on the new path might not
be the same as the old path. An endpoint MUST NOT restore its send
rate unless it is reasonably sure that the path is the same as the
previous path. For instance, a change in only port number is likely
indicative of a rebinding in a middlebox and not a complete change in
path. This determination likely depends on heuristics, which could
be imperfect; if the new path capacity is significantly reduced,
ultimately this relies on the congestion controller responding to
congestion signals and reduce send rates appropriately.
After verifying an address, the endpoint SHOULD update any address
validation tokens (Section 7.6) that it has issued to its peer if
those are no longer valid based on the changed address.
Address validation using the PING frame MAY be used at any time by
either peer. For instance, an endpoint might check that a peer is
still in possession of its address after a period of quiescence.
Upon seeing a connection migration, an endpoint that sees a new
address MUST abandon any address validation it is performing with
other addresses on the expectation that the validation is likely to
fail. Abandoning address validation primarily means not closing the
connection when a PONG frame is not received, but it could also mean
ceasing retransmissions of the PING frame. An endpoint that doesn't
retransmit a PING frame might receive a PONG frame, which it MUST
ignore.
7.8. Spurious Connection Migrations
A connection migration could be triggered by an attacker that is able
to capture and forward a packet such that it arrives before the
legitimate copy of that packet. Such a packet will appear to be a
legitimate connection migration and the legitimate copy will be
dropped as a duplicate.
After a spurious migration, validation of the source address will
fail because the entity at the source address does not have the
necessary cryptographic keys to read or respond to the PING frame
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that is sent to it, even if it wanted to. Such a spurious connection
migration could result in the connection being dropped when the
source address validation fails. This grants an attacker the ability
to terminate the connection.
Receipt of packets with higher packet numbers from the legitimate
address will trigger another connection migration. This will cause
the validation of the address of the spurious migration to be
abandoned.
To ensure that a peer sends packets from the legitimate address
before the validation of the new address can fail, an endpoint SHOULD
attempt to validate the old remote address before attempting to
validate the new address. If the connection migration is spurious,
then the legitimate address will be used to respond and the
connection will migrate back to the old address.
As with any address validation, packets containing retransmissions of
the PING frame validating an address MUST be sent to the address
being validated. Consequently, during a migration of a peer, an
endpoint could be sending to multiple remote addresses.
An endpoint MAY abandon address validation for an address that it
considers to be already valid. That is, if successive connection
migrations occur in quick succession with the final remote address
being identical to the initial remote address, the endpoint MAY
abandon address validation for that address.
7.9. Connection Termination
Connections should remain open until they become idle for a pre-
negotiated period of time. A QUIC connection, once established, can
be terminated in one of three ways:
o idle timeout (Section 7.9.2)
o immediate close (Section 7.9.3)
o stateless reset (Section 7.9.4)
7.9.1. Closing and Draining Connection States
The closing and draining connection states exist to ensure that
connections close cleanly and that delayed or reordered packets are
properly discarded. These states SHOULD persist for three times the
current Retransmission Timeout (RTO) interval as defined in
[QUIC-RECOVERY].
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An endpoint enters a closing period after initiating an immediate
close (Section 7.9.3) and optionally after an idle timeout
(Section 7.9.2). While closing, an endpoint MUST NOT send packets
unless they contain a CONNECTION_CLOSE or APPLICATION_CLOSE frame
(see Section 7.9.3 for details).
In the closing state, only a packet containing a closing frame can be
sent. An endpoint retains only enough information to generate a
packet containing a closing frame and to identify packets as
belonging to the connection. The connection ID and QUIC version is
sufficient information to identify packets for a closing connection;
an endpoint can discard all other connection state. An endpoint MAY
retain packet protection keys for incoming packets to allow it to
read and process a closing frame.
The draining state is entered once an endpoint receives a signal that
its peer is closing or draining. While otherwise identical to the
closing state, an endpoint in the draining state MUST NOT send any
packets. Retaining packet protection keys is unnecessary once a
connection is in the draining state.
An endpoint MAY transition from the closing period to the draining
period if it can confirm that its peer is also closing or draining.
Receiving a closing frame is sufficient confirmation, as is receiving
a stateless reset. The draining period SHOULD end when the closing
period would have ended. In other words, the endpoint can use the
same end time, but cease retransmission of the closing packet.
Disposing of connection state prior to the end of the closing or
draining period could cause delayed or reordered packets to be
handled poorly. Endpoints that have some alternative means to ensure
that late-arriving packets on the connection do not create QUIC
state, such as those that are able to close the UDP socket, MAY use
an abbreviated draining period which can allow for faster resource
recovery. Servers that retain an open socket for accepting new
connections SHOULD NOT exit the closing or draining period early.
Once the closing or draining period has ended, an endpoint SHOULD
discard all connection state. This results in new packets on the
connection being handled generically. For instance, an endpoint MAY
send a stateless reset in response to any further incoming packets.
The draining and closing periods do not apply when a stateless reset
(Section 7.9.4) is sent.
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7.9.2. Idle Timeout
A connection that remains idle for longer than the idle timeout (see
Section 7.4.1) is closed. A connection enters the draining state
when the idle timeout expires.
The time at which an idle timeout takes effect won't be perfectly
synchronized on both endpoints. An endpoint that sends packets near
the end of an idle period could have those packets discarded if its
peer enters the draining state before the packet is received.
7.9.3. Immediate Close
An endpoint sends a closing frame, either CONNECTION_CLOSE or
APPLICATION_CLOSE, to terminate the connection immediately. Either
closing frame causes all streams to immediately become closed; open
streams can be assumed to be implicitly reset.
After sending a closing frame, endpoints immediately enter the
closing state. During the closing period, an endpoint that sends a
closing frame SHOULD respond to any packet that it receives with
another packet containing a closing frame. To minimize the state
that an endpoint maintains for a closing connection, endpoints MAY
send the exact same packet. However, endpoints SHOULD limit the
number of packets they generate containing a closing frame. For
instance, an endpoint could progressively increase the number of
packets that it receives before sending additional packets or
increase the time between packets.
Note: Allowing retransmission of a packet contradicts other advice
in this document that recommends the creation of new packet
numbers for every packet. Sending new packet numbers is primarily
of advantage to loss recovery and congestion control, which are
not expected to be relevant for a closed connection.
Retransmitting the final packet requires less state.
After receiving a closing frame, endpoints enter the draining state.
An endpoint that receives a closing frame MAY send a single packet
containing a closing frame before entering the draining state, using
a CONNECTION_CLOSE frame and a NO_ERROR code if appropriate. An
endpoint MUST NOT send further packets, which could result in a
constant exchange of closing frames until the closing period on
either peer ended.
An immediate close can be used after an application protocol has
arranged to close a connection. This might be after the application
protocols negotiates a graceful shutdown. The application protocol
exchanges whatever messages that are needed to cause both endpoints
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to agree to close the connection, after which the application
requests that the connection be closed. The application protocol can
use an APPLICATION_CLOSE message with an appropriate error code to
signal closure.
7.9.4. Stateless Reset
A stateless reset is provided as an option of last resort for a
server that does not have access to the state of a connection. A
server crash or outage might result in clients continuing to send
data to a server that is unable to properly continue the connection.
A server that wishes to communicate a fatal connection error MUST use
a closing frame if it has sufficient state to do so.
To support this process, the server sends a stateless_reset_token
value during the handshake in the transport parameters. This value
is protected by encryption, so only client and server know this
value.
A server that receives packets that it cannot process sends a packet
in the following layout:
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
+-+-+-+-+-+-+-+-+
|0|C|K|Type (5) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ [Connection ID (64)] +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Packet Number (8/16/32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Random Octets (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ +
| |
+ Stateless Reset Token (128) +
| |
+ +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
A server copies the connection ID field from the packet that triggers
the stateless reset. A server omits the connection ID if explicitly
configured to do so, or if the client packet did not include a
connection ID.
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The Packet Number field is set to a randomized value. The server
SHOULD send a packet with a short header and a type of 0x1F. This
produces the shortest possible packet number encoding, which
minimizes the perceived gap between the last packet that the server
sent and this packet. A server MAY use a different short header
type, indicating a different packet number length, but a longer
packet number encoding might allow this message to be identified as a
stateless reset more easily using heuristics.
After the first short header octet and optional connection ID, the
server includes the value of the Stateless Reset Token that it
included in its transport parameters.
After the Packet Number, the server pads the message with an
arbitrary number of octets containing random values.
Finally, the last 16 octets of the packet are set to the value of the
Stateless Reset Token.
This design ensures that a stateless reset packet is - to the extent
possible - indistinguishable from a regular packet.
A stateless reset is not appropriate for signaling error conditions.
An endpoint that wishes to communicate a fatal connection error MUST
use a CONNECTION_CLOSE or APPLICATION_CLOSE frame if it has
sufficient state to do so.
This stateless reset design is specific to QUIC version 1. A server
that supports multiple versions of QUIC needs to generate a stateless
reset that will be accepted by clients that support any version that
the server might support (or might have supported prior to losing
state). Designers of new versions of QUIC need to be aware of this
and either reuse this design, or use a portion of the packet other
than the last 16 octets for carrying data.
7.9.4.1. Detecting a Stateless Reset
A client detects a potential stateless reset when a packet with a
short header either cannot be decrypted or is marked as a duplicate
packet. The client then compares the last 16 octets of the packet
with the Stateless Reset Token provided by the server in its
transport parameters. If these values are identical, the client MUST
enter the draining period and not send any further packets on this
connection. If the comparison fails, the packet can be discarded.
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7.9.4.2. Calculating a Stateless Reset Token
The stateless reset token MUST be difficult to guess. In order to
create a Stateless Reset Token, a server could randomly generate
[RFC4086] a secret for every connection that it creates. However,
this presents a coordination problem when there are multiple servers
in a cluster or a storage problem for a server that might lose state.
Stateless reset specifically exists to handle the case where state is
lost, so this approach is suboptimal.
A single static key can be used across all connections to the same
endpoint by generating the proof using a second iteration of a
preimage-resistant function that takes three inputs: the static key,
a the connection ID for the connection (see Section 5.6), and an
identifier for the server instance. A server could use HMAC
[RFC2104] (for example, HMAC(static_key, server_id || connection_id))
or HKDF [RFC5869] (for example, using the static key as input keying
material, with server and connection identifiers as salt). The
output of this function is truncated to 16 octets to produce the
Stateless Reset Token for that connection.
A server that loses state can use the same method to generate a valid
Stateless Reset Secret. The connection ID comes from the packet that
the server receives.
This design relies on the client always sending a connection ID in
its packets so that the server can use the connection ID from a
packet to reset the connection. A server that uses this design
cannot allow clients to omit a connection ID (that is, it cannot use
the truncate_connection_id transport parameter Section 7.4.1).
Revealing the Stateless Reset Token allows any entity to terminate
the connection, so a value can only be used once. This method for
choosing the Stateless Reset Token means that the combination of
server instance, connection ID, and static key cannot occur for
another connection. A connection ID from a connection that is reset
by revealing the Stateless Reset Token cannot be reused for new
connections at the same server without first changing to use a
different static key or server identifier.
Note that Stateless Reset messages do not have any cryptographic
protection.
8. Frame Types and Formats
As described in Section 6, Regular packets contain one or more
frames. We now describe the various QUIC frame types that can be
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present in a Regular packet. The use of these frames and various
frame header bits are described in subsequent sections.
8.1. Variable-Length Integer Encoding
QUIC frames use a common variable-length encoding for all non-
negative integer values. This encoding ensures that smaller integer
values need fewer octets to encode.
The QUIC variable-length integer encoding reserves the two most
significant bits of the first octet to encode the base 2 logarithm of
the integer encoding length in octets. The integer value is encoded
on the remaining bits, in network byte order.
This means that integers are encoded on 1, 2, 4, or 8 octets and can
encode 6, 14, 30, or 62 bit values respectively. Table 4 summarizes
the encoding properties.
+------+--------+-------------+-----------------------+
| 2Bit | Length | Usable Bits | Range |
+------+--------+-------------+-----------------------+
| 00 | 1 | 6 | 0-63 |
| | | | |
| 01 | 2 | 14 | 0-16383 |
| | | | |
| 10 | 4 | 30 | 0-1073741823 |
| | | | |
| 11 | 8 | 62 | 0-4611686018427387903 |
+------+--------+-------------+-----------------------+
Table 4: Summary of Integer Encodings
For example, the eight octet sequence c2 19 7c 5e ff 14 e8 8c (in
hexadecimal) decodes to the decimal value 151288809941952652; the
four octet sequence 9d 7f 3e 7d decodes to 494878333; the two octet
sequence 7b bd decodes to 15293; and the single octet 25 decodes to
37 (as does the two octet sequence 40 25).
Error codes (Section 12.3) are described using integers, but do not
use this encoding.
8.2. PADDING Frame
The PADDING frame (type=0x00) has no semantic value. PADDING frames
can be used to increase the size of a packet. Padding can be used to
increase an initial client packet to the minimum required size, or to
provide protection against traffic analysis for protected packets.
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A PADDING frame has no content. That is, a PADDING frame consists of
the single octet that identifies the frame as a PADDING frame.
8.3. RST_STREAM Frame
An endpoint may use a RST_STREAM frame (type=0x01) to abruptly
terminate a stream.
After sending a RST_STREAM, an endpoint ceases transmission and
retransmission of STREAM frames on the identified stream. A receiver
of RST_STREAM can discard any data that it already received on that
stream.
The RST_STREAM frame is as follows:
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Application Error Code (16) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Final Offset (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The fields are:
Stream ID: A variable-length integer encoding of the Stream ID of
the stream being terminated.
Application Protocol Error Code: A 16-bit application protocol error
code (see Section 12.4) which indicates why the stream is being
closed.
Final Offset: A variable-length integer indicating the absolute byte
offset of the end of data written on this stream by the RST_STREAM
sender.
8.4. CONNECTION_CLOSE frame
An endpoint sends a CONNECTION_CLOSE frame (type=0x02) to notify its
peer that the connection is being closed. CONNECTION_CLOSE is used
to signal errors at the QUIC layer, or the absence of errors (with
the NO_ERROR code).
If there are open streams that haven't been explicitly closed, they
are implicitly closed when the connection is closed.
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The CONNECTION_CLOSE frame is as follows:
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 (16) | Reason Phrase Length (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Reason Phrase (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The fields of a CONNECTION_CLOSE frame are as follows:
Error Code: A 16-bit error code which indicates the reason for
closing this connection. CONNECTION_CLOSE uses codes from the
space defined in Section 12.3 (APPLICATION_CLOSE uses codes from
the application protocol error code space, see Section 12.4).
Reason Phrase Length: A variable-length integer specifying the
length of the reason phrase in bytes. Note that a
CONNECTION_CLOSE frame cannot be split between packets, so in
practice any limits on packet size will also limit the space
available for a reason phrase.
Reason Phrase: A human-readable explanation for why the connection
was closed. This can be zero length if the sender chooses to not
give details beyond the Error Code. This SHOULD be a UTF-8
encoded string [RFC3629].
8.5. APPLICATION_CLOSE frame
An APPLICATION_CLOSE frame (type=0x03) uses the same format as the
CONNECTION_CLOSE frame (Section 8.4), except that it uses error codes
from the application protocol error code space (Section 12.4) instead
of the transport error code space.
Other than the error code space, the format and semantics of the
APPLICATION_CLOSE frame are identical to the CONNECTION_CLOSE frame.
8.6. MAX_DATA Frame
The MAX_DATA frame (type=0x04) is used in flow control to inform the
peer of the maximum amount of data that can be sent on the connection
as a whole.
The frame is as follows:
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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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Maximum Data (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The fields in the MAX_DATA frame are as follows:
Maximum Data: A variable-length integer indicating the maximum
amount of data that can be sent on the entire connection, in units
of octets.
All data sent in STREAM frames counts toward this limit, with the
exception of data on stream 0. The sum of the largest received
offsets on all streams - including streams in terminal states, but
excluding stream 0 - MUST NOT exceed the value advertised by a
receiver. An endpoint MUST terminate a connection with a
QUIC_FLOW_CONTROL_RECEIVED_TOO_MUCH_DATA error if it receives more
data than the maximum data value that it has sent, unless this is a
result of a change in the initial limits (see Section 7.4.2).
8.7. MAX_STREAM_DATA Frame
The MAX_STREAM_DATA frame (type=0x05) is used in flow control to
inform a peer of the maximum amount of data that can be sent on a
stream.
The frame is as follows:
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Maximum Stream Data (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The fields in the MAX_STREAM_DATA frame are as follows:
Stream ID: The stream ID of the stream that is affected 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 octets.
When counting data toward this limit, an endpoint accounts for the
largest received offset of data that is sent or received on the
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stream. Loss or reordering can mean that the largest received offset
on a stream can be greater than the total size of data received on
that stream. Receiving STREAM frames might not increase the largest
received offset.
The data sent on a stream MUST NOT exceed the largest maximum stream
data value advertised by the receiver. An endpoint MUST terminate a
connection with a FLOW_CONTROL_ERROR error if it receives more data
than the largest maximum stream data that it has sent for the
affected stream, unless this is a result of a change in the initial
limits (see Section 7.4.2).
8.8. MAX_STREAM_ID Frame
The MAX_STREAM_ID frame (type=0x06) informs the peer of the maximum
stream ID that they are permitted to open.
The frame is as follows:
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Maximum Stream ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The fields in the MAX_STREAM_ID frame are as follows:
Maximum Stream ID: ID of the maximum unidirectional or bidirectional
peer-initiated stream ID for the connection encoded as a variable-
length integer. The limit applies to unidirectional steams if the
second least signification bit of the stream ID is 1, and applies
to bidirectional streams if it is 0.
Loss or reordering can mean that a MAX_STREAM_ID frame can be
received which states a lower stream limit than the client has
previously received. MAX_STREAM_ID frames which do not increase the
maximum stream ID MUST be ignored.
A peer MUST NOT initiate a stream with a higher stream ID than the
greatest maximum stream ID it has received. An endpoint MUST
terminate a connection with a STREAM_ID_ERROR error if a peer
initiates a stream with a higher stream ID than it has sent, unless
this is a result of a change in the initial limits (see
Section 7.4.2).
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8.9. PING Frame
Endpoints can use PING frames (type=0x07) to verify that their peers
are still alive or to check reachability to the peer.
The PING frame contains a variable-length payload.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Length(8) | Data (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Length: This 8-bit value describes the length of the Data field.
Data: This variable-length field contains arbitrary data.
A PING frame with an empty Data field causes the packet containing it
to be acknowledged as normal. No other action is required of the
recipient.
An empty PING frame can be used to keep a connection alive when an
application or application protocol wishes to prevent the connection
from timing out. An application protocol SHOULD provide guidance
about the conditions under which generating a PING is recommended.
This guidance SHOULD indicate whether it is the client or the server
that is expected to send the PING. Having both endpoints send PING
frames without coordination can produce an excessive number of
packets and poor performance.
If the Data field is not empty, the recipient of this frame MUST
generate a PONG frame (Section 8.15) containing the same Data. A
PING frame with data is not appropriate for use in keeping a
connection alive, because the PONG frame elicits an acknowledgement,
causing the sender of the original PING to send two packets.
A connection will time out if no packets are sent or received for a
period longer than the time specified in the idle_timeout transport
parameter (see Section 7.9). However, state in middleboxes might
time out earlier than that. Though REQ-5 in [RFC4787] recommends a 2
minute timeout interval, experience shows that sending packets every
15 to 30 seconds is necessary to prevent the majority of middleboxes
from losing state for UDP flows.
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8.10. BLOCKED Frame
A sender SHOULD send a BLOCKED frame (type=0x08) when it wishes to
send data, but is unable to due to connection-level flow control (see
Section 11.2.1). BLOCKED frames can be used as input to tuning of
flow control algorithms (see Section 11.1.2).
The BLOCKED frame is as follows:
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Offset (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The BLOCKED frame contains a single field.
Offset: A variable-length integer indicating the connection-level
offset at which the blocking occurred.
8.11. STREAM_BLOCKED Frame
A sender SHOULD send a STREAM_BLOCKED frame (type=0x09) when it
wishes to send data, but is unable to due to stream-level flow
control. This frame is analogous to BLOCKED (Section 8.10).
The STREAM_BLOCKED frame is as follows:
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Offset (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The STREAM_BLOCKED frame contains two fields:
Stream ID: A variable-length integer indicating the stream which is
flow control blocked.
Offset: A variable-length integer indicating the offset of the
stream at which the blocking occurred.
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8.12. STREAM_ID_BLOCKED Frame
A sender MAY send a STREAM_ID_BLOCKED frame (type=0x0a) when it
wishes to open a stream, but is unable to due to the maximum stream
ID limit set by its peer (see Section 8.8). This does not open the
stream, but informs the peer that a new stream was needed, but the
stream limit prevented the creation of the stream.
The STREAM_ID_BLOCKED frame is as follows:
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The STREAM_ID_BLOCKED frame contains a single field.
Stream ID: A variable-length integer indicating the highest stream
ID that the sender was permitted to open.
8.13. NEW_CONNECTION_ID Frame
A server sends a NEW_CONNECTION_ID frame (type=0x0b) to provide the
client with alternative connection IDs that can be used to break
linkability when migrating connections (see Section 7.7.1).
The NEW_CONNECTION_ID is as follows:
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Sequence (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ Connection ID (64) +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ +
| |
+ Stateless Reset Token (128) +
| |
+ +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The fields are:
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Sequence: A variable-length integer. This value starts at 0 and
increases by 1 for each connection ID that is provided by the
server. The connection ID that is assigned during the handshake
is assumed to have a sequence of -1. That is, the value selected
during the handshake comes immediately before the first value that
a server can send.
Connection ID: A 64-bit connection ID.
Stateless Reset Token: A 128-bit value that will be used to for a
stateless reset when the associated connection ID is used (see
Section 7.9.4).
8.14. STOP_SENDING Frame
An endpoint may use a STOP_SENDING frame (type=0x0c) to communicate
that incoming data is being discarded on receipt at application
request. This signals a peer to abruptly terminate transmission on a
stream.
The STOP_SENDING frame is as follows:
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Application Error Code (16) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The fields are:
Stream ID: A variable-length integer carrying the Stream ID of the
stream being ignored.
Application Error Code: A 16-bit, application-specified reason the
sender is ignoring the stream (see Section 12.4).
8.15. PONG Frame
The PONG frame (type=0x0d) is sent in response to a PING frame that
contains data. Its format is identical to the PING frame
(Section 8.9).
An endpoint that receives an unsolicited PONG frame - that is, a PONG
frame containing a payload that is empty MUST generate a connection
error of type FRAME_ERROR, indicating the PONG frame (that is,
0x10d). If the content of a PONG frame does not match the content of
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a PING frame previously sent by the endpoint, the endpoint MAY
generate a connection error of type UNSOLICITED_PONG.
8.16. ACK Frame
Receivers send ACK frames (type=0xe) to inform senders which packets
they have received and processed. A sent packet that has never been
acknowledged is missing. The ACK frame contains any number of ACK
blocks. ACK blocks are ranges of acknowledged packets.
Unlike TCP SACKs, QUIC acknowledgements are irrevocable. Once a
packet has been acknowledged, even if it does not appear in a future
ACK frame, it remains acknowledged.
A client MUST NOT acknowledge Version Negotiation or Retry packets.
These packet types contain packet numbers selected by the client, not
the server.
A sender MAY intentionally skip packet numbers to introduce entropy
into the connection, to avoid opportunistic acknowledgement attacks.
The sender SHOULD close the connection if an unsent packet number is
acknowledged. The format of the ACK frame is efficient at expressing
even long blocks of missing packets, allowing for large,
unpredictable gaps.
An ACK frame is shown below.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Largest Acknowledged (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ACK Delay (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ACK Block Count (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ACK Blocks (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 7: ACK Frame Format
The fields in the ACK frame are as follows:
Largest Acknowledged: A variable-length integer representing the
largest packet number the peer is acknowledging; this is usually
the largest packet number that the peer has received prior to
generating the ACK frame.
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ACK Delay: A variable-length integer including the time in
microseconds that the largest acknowledged packet, as indicated in
the Largest Acknowledged field, was received by this peer to when
this ACK was sent. The value of the ACK Delay field is scaled by
multiplying the encoded value by the 2 to the power of the value
of the "ack_delay_exponent" transport parameter set by the sender
of the ACK frame. The "ack_delay_exponent" defaults to 3, or a
multiplier of 8 (see Section 7.4.1). Scaling in this fashion
allows for a larger range of values with a shorter encoding at the
cost of lower resolution.
ACK Block Count: The number of Additional ACK Block (and Gap) fields
after the First ACK Block.
ACK Blocks: Contains one or more blocks of packet numbers which have
been successfully received, see Section 8.16.1.
8.16.1. ACK Block Section
The ACK Block Section consists of alternating Gap and ACK Block
fields in descending packet number order. A First Ack Block field is
followed by a variable number of alternating Gap and Additional ACK
Blocks. The number of Gap and Additional ACK Block fields is
determined by the ACK Block Count field.
Gap and ACK Block fields use a relative integer encoding for
efficiency. Though each encoded value is positive, the values are
subtracted, so that each ACK Block describes progressively lower-
numbered packets. As long as contiguous ranges of packets are small,
the variable-length integer encoding ensures that each range can be
expressed in a small number of octets.
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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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| First ACK Block (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Gap (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Additional ACK Block (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Gap (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Additional ACK Block (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Gap (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Additional ACK Block (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 8: ACK Block Section
Each ACK Block acknowledges a contiguous range of packets by
indicating the number of acknowledged packets that precede the
largest packet number in that block. A value of zero indicates that
only the largest packet number is acknowledged. Larger ACK Block
values indicate a larger range, with corresponding lower values for
the smallest packet number in the range. Thus, given a largest
packet number for the ACK, the smallest value is determined by the
formula:
smallest = largest - ack_block
The range of packets that are acknowledged by the ACK block include
the range from the smallest packet number to the largest, inclusive.
The largest value for the First ACK Block is determined by the
Largest Acknowledged field; the largest for Additional ACK Blocks is
determined by cumulatively subtracting the size of all preceding ACK
Blocks and Gaps.
Each Gap indicates a range of packets that are not being
acknowledged. The number of packets in the gap is one higher than
the encoded value of the Gap Field.
The value of the Gap field establishes the largest packet number
value for the ACK block that follows the gap using the following
formula:
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largest = previous_smallest - gap - 2
If the calculated value for largest or smallest packet number for any
ACK Block is negative, an endpoint MUST generate a connection error
of type FRAME_ERROR indicating an error in an ACK frame (that is,
0x10d).
The fields in the ACK Block Section are:
First ACK Block: A variable-length integer indicating the number of
contiguous packets preceding the Largest Acknowledged that are
being acknowledged.
Gap (repeated): A variable-length integer indicating the number of
contiguous unacknowledged packets preceding the packet number one
lower than the smallest in the preceding ACK Block.
ACK Block (repeated): A variable-length integer indicating the
number of contiguous acknowledged packets preceding the largest
packet number, as determined by the preceding Gap.
8.16.2. Sending ACK Frames
Implementations MUST NOT generate packets that only contain ACK
frames in response to packets which only contain ACK frames.
However, they MUST acknowledge packets containing only ACK frames
when sending ACK frames in response to other packets.
Implementations MUST NOT send more than one ACK frame per received
packet that contains frames other than ACK frames. Packets
containing non-ACK frames MUST be acknowledged immediately or when a
delayed ack timer expires.
To limit ACK blocks to those that have not yet been received by the
sender, the receiver SHOULD track which ACK frames have been
acknowledged by its peer. Once an ACK frame has been acknowledged,
the packets it acknowledges SHOULD NOT be acknowledged again.
A receiver that is only sending ACK frames will not receive
acknowledgments for its packets. Sending an occasional MAX_DATA or
MAX_STREAM_DATA frame as data is received will ensure that
acknowledgements are generated by a peer. Otherwise, an endpoint MAY
send a PING frame once per RTT to solicit an acknowledgment.
To limit receiver state or the size of ACK frames, a receiver MAY
limit the number of ACK blocks it sends. A receiver can do this even
without receiving acknowledgment of its ACK frames, with the
knowledge this could cause the sender to unnecessarily retransmit
some data. Standard QUIC [QUIC-RECOVERY] algorithms declare packets
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lost after sufficiently newer packets are acknowledged. Therefore,
the receiver SHOULD repeatedly acknowledge newly received packets in
preference to packets received in the past.
8.16.3. ACK Frames and Packet Protection
ACK frames that acknowledge protected packets MUST be carried in a
packet that has an equivalent or greater level of packet protection.
Packets that are protected with 1-RTT keys MUST be acknowledged in
packets that are also protected with 1-RTT keys.
A packet that is not protected and claims to acknowledge a packet
number that was sent with packet protection is not valid. An
unprotected packet that carries acknowledgments for protected packets
MUST be discarded in its entirety.
Packets that a client sends with 0-RTT packet protection MUST be
acknowledged by the server in packets protected by 1-RTT keys. This
can mean that the client is unable to use these acknowledgments if
the server cryptographic handshake messages are delayed or lost.
Note that the same limitation applies to other data sent by the
server protected by the 1-RTT keys.
Unprotected packets, such as those that carry the initial
cryptographic handshake messages, MAY be acknowledged in unprotected
packets. Unprotected packets are vulnerable to falsification or
modification. Unprotected packets can be acknowledged along with
protected packets in a protected packet.
An endpoint SHOULD acknowledge packets containing cryptographic
handshake messages in the next unprotected packet that it sends,
unless it is able to acknowledge those packets in later packets
protected by 1-RTT keys. At the completion of the cryptographic
handshake, both peers send unprotected packets containing
cryptographic handshake messages followed by packets protected by
1-RTT keys. An endpoint SHOULD acknowledge the unprotected packets
that complete the cryptographic handshake in a protected packet,
because its peer is guaranteed to have access to 1-RTT packet
protection keys.
For instance, a server acknowledges a TLS ClientHello in the packet
that carries the TLS ServerHello; similarly, a client can acknowledge
a TLS HelloRetryRequest in the packet containing a second TLS
ClientHello. The complete set of server handshake messages (TLS
ServerHello through to Finished) might be acknowledged by a client in
protected packets, because it is certain that the server is able to
decipher the packet.
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8.17. STREAM Frames
STREAM frames implicitly create a stream and carry stream data. The
STREAM frame takes the form 0b00010XXX (or the set of values from
0x10 to 0x17). The value of the three low-order bits of the frame
type determine the fields that are present in the frame.
o The FIN bit (0x01) of the frame type is set only on frames that
contain the final offset of the stream. Setting this bit
indicates that the frame marks the end of the stream.
o The LEN bit (0x02) in the frame type is set to indicate that there
is a Length field present. If this bit is set to 0, the Length
field is absent and the Stream Data field extends to the end of
the packet. If this bit is set to 1, the Length field is present.
o The OFF bit (0x04) in the frame type is set to indicate that there
is an Offset field present. When set to 1, the Offset field is
present; when set to 0, the Offset field is absent and the Stream
Data starts at an offset of 0 (that is, the frame contains the
first octets of the stream, or the end of a stream that includes
no data).
A STREAM frame is shown below.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Offset (i)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Length (i)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream Data (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 9: STREAM Frame Format
The STREAM frame contains the following fields:
Stream ID: A variable-length integer indicating the stream ID of the
stream (see Section 10.1).
Offset: A variable-length integer specifying the byte offset in the
stream for the data in this STREAM frame. This field is present
when the OFF bit is set to 1. When the Offset field is absent,
the offset is 0.
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Length: A variable-length integer specifying the length of the
Stream Data field in this STREAM frame. This field is present
when the LEN bit is set to 1. When the LEN bit is set to 0, the
Stream Data field consumes all the remaining octets in the packet.
Stream Data: The bytes from the designated stream to be delivered.
A stream frame's Stream Data MUST NOT be empty, unless the FIN bit is
set. When the FIN flag is sent on an empty STREAM frame, the offset
in the STREAM frame is the offset of the next byte that would be
sent.
The first byte in the stream has an offset of 0. The largest offset
delivered on a stream - the sum of the re-constructed offset and data
length - MUST be less than 2^62.
Stream multiplexing is achieved by interleaving STREAM frames from
multiple streams into one or more QUIC packets. A single QUIC packet
can include multiple STREAM frames from one or more streams.
Implementation note: One of the benefits of QUIC is avoidance of
head-of-line blocking across multiple streams. When a packet loss
occurs, only streams with data in that packet are blocked waiting for
a retransmission to be received, while other streams can continue
making progress. Note that when data from multiple streams is
bundled into a single QUIC packet, loss of that packet blocks all
those streams from making progress. An implementation is therefore
advised to bundle as few streams as necessary in outgoing packets
without losing transmission efficiency to underfilled packets.
9. Packetization and Reliability
The Path Maximum Transmission Unit (PMTU) is the maximum size of the
entire IP header, UDP header, and UDP payload. The UDP payload
includes the QUIC packet header, protected payload, and any
authentication fields.
All QUIC packets SHOULD be sized to fit within the estimated PMTU to
avoid IP fragmentation or packet drops. To optimize bandwidth
efficiency, endpoints SHOULD use Packetization Layer PMTU Discovery
([PLPMTUD]) and MAY use PMTU Discovery ([PMTUDv4], [PMTUDv6]) for
detecting the PMTU, setting the PMTU appropriately, and storing the
result of previous PMTU determinations.
In the absence of these mechanisms, QUIC endpoints SHOULD NOT send IP
packets larger than 1280 octets. Assuming the minimum IP header
size, this results in a QUIC packet size of 1232 octets for IPv6 and
1252 octets for IPv4.
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QUIC endpoints that implement any kind of PMTU discovery SHOULD
maintain an estimate for each combination of local and remote IP
addresses (as each pairing could have a different maximum MTU in the
path).
QUIC depends on the network path supporting a MTU of at least 1280
octets. This is the IPv6 minimum and therefore also supported by
most modern IPv4 networks. An endpoint MUST NOT reduce their MTU
below this number, even if it receives signals that indicate a
smaller limit might exist.
Clients MUST ensure that the first packet in a connection, and any
retransmissions of those octets, has a QUIC packet size of least 1200
octets. The packet size for a QUIC packet includes the QUIC header
and integrity check, but not the UDP or IP header.
The initial client packet SHOULD be padded to exactly 1200 octets
unless the client has a reasonable assurance that the PMTU is larger.
Sending a packet of this size ensures that the network path supports
an MTU of this size and helps reduce the amplitude of amplification
attacks caused by server responses toward an unverified client
address.
Servers MUST ignore an initial plaintext packet from a client if its
total size is less than 1200 octets.
If a QUIC endpoint determines that the PMTU between any pair of local
and remote IP addresses has fallen below 1280 octets, it MUST
immediately cease sending QUIC packets on the affected path. This
could result in termination of the connection if an alternative path
cannot be found.
A sender bundles one or more frames in a Regular QUIC packet (see
Section 6).
A sender SHOULD minimize per-packet bandwidth and computational costs
by bundling as many frames as possible within a QUIC packet. A
sender MAY wait for a short period of time to bundle multiple frames
before sending a packet that is not maximally packed, to avoid
sending out large numbers of small packets. An implementation may
use heuristics about expected application sending behavior to
determine whether and for how long to wait. This waiting period is
an implementation decision, and an implementation should be careful
to delay conservatively, since any delay is likely to increase
application-visible latency.
Regular QUIC packets are "containers" of frames; a packet is never
retransmitted whole. How an endpoint handles the loss of the frame
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depends on the type of the frame. Some frames are simply
retransmitted, some have their contents moved to new frames, and
others are never retransmitted.
When a packet is detected as lost, the sender re-sends any frames as
necessary:
o All application data sent in STREAM frames MUST be retransmitted,
unless the endpoint has sent a RST_STREAM for that stream. When
an endpoint sends a RST_STREAM frame, data outstanding on that
stream SHOULD NOT be retransmitted, since subsequent data on this
stream is expected to not be delivered by the receiver.
o ACK and PADDING frames MUST NOT be retransmitted. ACK frames
containing updated information will be sent as described in
Section 8.16.
o STOP_SENDING frames MUST be retransmitted until the receive stream
enters either a "Data Recvd" or "Reset Recvd" state. See
Section 10.3.
o The most recent MAX_STREAM_DATA frame for a stream MUST be
retransmitted until the receive stream enters a "Size Known"
state. Any previous unacknowledged MAX_STREAM_DATA frame for the
same stream SHOULD NOT be retransmitted since a newer
MAX_STREAM_DATA frame for a stream obviates the need for
delivering older ones. Similarly, the most recent MAX_DATA frame
MUST be retransmitted; previous unacknowledged ones SHOULD NOT be
retransmitted.
o BLOCKED, STREAM_BLOCKED, and STREAM_ID_BLOCKED frames SHOULD be
retransmitted if the sender is still blocked on the same limit.
If the limit has been increased since the frame was originally
sent, the frame SHOULD NOT be retransmitted.
o All other frames MUST be retransmitted.
Upon detecting losses, a sender MUST take appropriate congestion
control action. The details of loss detection and congestion control
are described in [QUIC-RECOVERY].
A packet MUST NOT be acknowledged until packet protection has been
successfully removed and all frames contained in the packet have been
processed. For STREAM frames, this means the data has been queued
(but not necessarily delivered to the application). This also means
that any stream state transitions triggered by STREAM or RST_STREAM
frames have occurred. Once the packet has been fully processed, a
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receiver acknowledges receipt by sending one or more ACK frames
containing the packet number of the received packet.
To avoid creating an indefinite feedback loop, an endpoint MUST NOT
send an ACK frame in response to a packet containing only ACK or
PADDING frames, even if there are packet gaps which precede the
received packet. The endpoint MUST acknowledge packets containing
only ACK or PADDING frames in the next ACK frame that it sends.
Strategies and implications of the frequency of generating
acknowledgments are discussed in more detail in [QUIC-RECOVERY].
9.1. Special Considerations for PMTU Discovery
Traditional ICMP-based path MTU discovery in IPv4 [RFC1191] is
potentially vulnerable to off-path attacks that successfully guess
the IP/port 4-tuple and reduce the MTU to a bandwidth-inefficient
value. TCP connections mitigate this risk by using the (at minimum)
8 bytes of transport header echoed in the ICMP message to validate
the TCP sequence number as valid for the current connection.
However, as QUIC operates over UDP, in IPv4 the echoed information
could consist only of the IP and UDP headers, which usually has
insufficient entropy to mitigate off-path attacks.
As a result, endpoints that implement PMTUD in IPv4 SHOULD take steps
to mitigate this risk. For instance, an application could:
o Set the IPv4 Don't Fragment (DF) bit on a small proportion of
packets, so that most invalid ICMP messages arrive when there are
no DF packets outstanding, and can therefore be identified as
spurious.
o Store additional information from the IP or UDP headers from DF
packets (for example, the IP ID or UDP checksum) to further
authenticate incoming Datagram Too Big messages.
o Any reduction in PMTU due to a report contained in an ICMP packet
is provisional until QUIC's loss detection algorithm determines
that the packet is actually lost.
10. Streams: QUIC's Data Structuring Abstraction
Streams in QUIC provide a lightweight, ordered byte-stream
abstraction.
There are two basic types of stream in QUIC. Unidirectional streams
carry data in one direction only; bidirectional streams allow for
data to be sent in both directions. Different stream identifiers are
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used to distinguish between unidirectional and bidirectional streams,
as well as to create a separation between streams that are initiated
by the client and server (see Section 10.1).
Either type of stream can be created by either endpoint, can
concurrently send data interleaved with other streams, and can be
cancelled.
Data that is received on a stream is delivered in order within that
stream, but there is no particular delivery order across streams.
Transmit ordering among streams is left to the implementation.
The creation and destruction of streams are expected to have minimal
bandwidth and computational cost. A single STREAM frame may create,
carry data for, and terminate a stream, or a stream may last the
entire duration of a connection.
Streams are individually flow controlled, allowing an endpoint to
limit memory commitment and to apply back pressure. The creation of
streams is also flow controlled, with each peer declaring the maximum
stream ID it is willing to accept at a given time.
An alternative view of QUIC streams is as an elastic "message"
abstraction, similar to the way ephemeral streams are used in SST
[SST], which may be a more appealing description for some
applications.
10.1. Stream Identifiers
Streams are identified by an unsigned 62-bit integer, referred to as
the Stream ID. The least significant two bits of the Stream ID are
used to identify the type of stream (unidirectional or bidirectional)
and the initiator of the stream.
The least significant bit (0x1) of the Stream ID identifies the
initiator of the stream. Clients initiate even-numbered streams
(those with the least significant bit set to 0); servers initiate
odd-numbered streams (with the bit set to 1). Separation of the
stream identifiers ensures that client and server are able to open
streams without the latency imposed by negotiating for an identifier.
If an endpoint receives a frame for a stream that it expects to
initiate (i.e., odd-numbered for the client or even-numbered for the
server), but which it has not yet opened, it MUST close the
connection with error code STREAM_STATE_ERROR.
The second least significant bit (0x2) of the Stream ID
differentiates between unidirectional streams and bidirectional
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streams. Unidirectional streams always have this bit set to 1 and
bidirectional streams have this bit set to 0.
The two type bits from a Stream ID therefore identify streams as
summarized in Table 5.
+----------+----------------------------------+
| Low Bits | Stream Type |
+----------+----------------------------------+
| 0x0 | Client-Initiated, Bidirectional |
| | |
| 0x1 | Server-Initiated, Bidirectional |
| | |
| 0x2 | Client-Initiated, Unidirectional |
| | |
| 0x3 | Server-Initiated, Unidirectional |
+----------+----------------------------------+
Table 5: Stream ID Types
Stream ID 0 (0x0) is a client-initiated, bidirectional stream that is
used for the cryptographic handshake. Stream 0 MUST NOT be used for
application data.
A QUIC endpoint MUST NOT reuse a Stream ID. Open streams can be used
in any order. Streams that are used out of order result in opening
all lower-numbered streams of the same type in the same direction.
Stream IDs are encoded as a variable-length integer (see
Section 8.1).
10.2. Stream States
This section describes the two types of QUIC stream in terms of the
states of their send or receive components. Two state machines are
described: one for streams on which an endpoint transmits data
(Section 10.2.1); another for streams from which an endpoint receives
data (Section 10.2.2).
Unidirectional streams use the applicable state machine directly.
Bidirectional streams use both state machines. For the most part,
the use of these state machines is the same whether the stream is
unidirectional or bidirectional. The conditions for opening a stream
are slightly more complex for a bidirectional stream because the
opening of either send or receive causes the stream to open in both
directions.
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Opening a stream causes all lower-numbered streams of the same type
to implicitly open. This includes both send and receive streams if
the stream is bidirectional. For bidirectional streams, an endpoint
can send data on an implicitly opened stream. On both unidirectional
and bidirectional streams, an endpoint MAY send MAX_STREAM_DATA or
STOP_SENDING on implicitly opened streams. An endpoint SHOULD NOT
implicitly open streams that it initiates, instead opening streams in
order.
Note: These states are largely informative. This document uses
stream states to describe rules for when and how different types
of frames can be sent and the reactions that are expected when
different types of frames are received. Though these state
machines are intended to be useful in implementing QUIC, these
states aren't intended to constrain implementations. An
implementation can define a different state machine as long as its
behavior is consistent with an implementation that implements
these states.
10.2.1. Send Stream States
Figure 10 shows the states for the part of a stream that sends data
to a peer.
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o
| Application Open
| Open Paired Stream (bidirectional)
v
+-------+
| Open | Send RST_STREAM
| |-----------------------.
+-------+ |
| |
| Send STREAM / |
| STREAM_BLOCKED |
v |
+-------+ |
| Send | Send RST_STREAM |
| |---------------------->|
+-------+ |
| |
| Send STREAM + FIN |
v v
+-------+ +-------+
| Data | Send RST_STREAM | Reset |
| Sent +------------------>| Sent |
+-------+ +-------+
| |
| Recv All ACKs | Recv ACK
v v
+-------+ +-------+
| Data | | Reset |
| Recvd | | Recvd |
+-------+ +-------+
Figure 10: States for Send Streams
The sending part of stream that the endpoint initiates (types 0 and 2
for clients, 1 and 3 for servers) is opened by the application or
application protocol. The "Open" state represents a newly created
stream that is able to accept data from the application. Stream data
might be buffered in this state in preparation for sending.
The sending part of a bidirectional stream initiated by a peer (type
0 for a server, type 1 for a client) enters the "Open" state if the
receiving part enters the "Recv" state.
Sending the first STREAM or STREAM_BLOCKED frame causes a send stream
to enter the "Send" state. An implementation might choose to defer
allocating a Stream ID to a send stream until it sends the first
frame and enters this state, which can allow for better stream
prioritization.
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In the "Send" state, an endpoint transmits - and retransmits as
necessary - data in STREAM frames. The endpoint respects the flow
control limits of its peer, accepting MAX_STREAM_DATA frames. An
endpoint in the "Send" state generates STREAM_BLOCKED frames if it
encounters flow control limits.
After the application indicates that stream data is complete and a
STREAM frame containing the FIN bit is sent, the send stream enters
the "Data Sent" state. From this state, the endpoint only
retransmits stream data as necessary. The endpoint no longer needs
to track flow control limits or send STREAM_BLOCKED frames for a send
stream in this state. The endpoint can ignore any MAX_STREAM_DATA
frames it receives from its peer in this state; MAX_STREAM_DATA
frames might be received until the peer receives the final stream
offset.
Once all stream data has been successfully acknowledged, the send
stream enters the "Data Recvd" state, which is a terminal state.
From any of the "Open", "Send", or "Data Sent" states, an application
can signal that it wishes to abandon transmission of stream data.
Similarly, the endpoint might receive a STOP_SENDING frame from its
peer. In either case, the endpoint sends a RST_STREAM frame, which
causes the stream to enter the "Reset Sent" state.
An endpoint MAY send a RST_STREAM as the first frame on a send
stream; this causes the send stream to open and then immediately
transition to the "Reset Sent" state.
Once a packet containing a RST_STREAM has been acknowledged, the send
stream enters the "Reset Recvd" state, which is a terminal state.
10.2.2. Receive Stream States
Figure 11 shows the states for the part of a stream that receives
data from a peer. The states for a receive stream mirror only some
of the states of the send stream at the peer. A receive stream
doesn't track states on the send stream that cannot be observed, such
as the "Open" state; instead, receive streams track the delivery of
data to the application or application protocol some of which cannot
be observed by the sender.
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o
| Recv STREAM / STREAM_BLOCKED / RST_STREAM
| Open Paired Stream (bidirectional)
| Recv MAX_STREAM_DATA
v
+-------+
| Recv | Recv RST_STREAM
| |-----------------------.
+-------+ |
| |
| Recv STREAM + FIN |
v |
+-------+ |
| Size | Recv RST_STREAM |
| Known +---------------------->|
+-------+ |
| |
| Recv All Data |
v v
+-------+ +-------+
| Data | Recv RST_STREAM | Reset |
| Recvd +<-- (optional) --->| Recvd |
+-------+ +-------+
| |
| App Read All Data | App Read RST
v v
+-------+ +-------+
| Data | | Reset |
| Read | | Read |
+-------+ +-------+
Figure 11: States for Receive Streams
The receiving part of a stream initiated by a peer (types 1 and 3 for
a client, or 0 and 2 for a server) are created when the first STREAM,
STREAM_BLOCKED, RST_STREAM, or MAX_STREAM_DATA (bidirectional only,
see below) is received for that stream. The initial state for a
receive stream is "Recv". Receiving a RST_STREAM frame causes the
receive stream to immediately transition to the "Reset Recvd".
The receive stream enters the "Recv" state when the sending part of a
bidirectional stream initiated by the endpoint (type 0 for a client,
type 1 for a server) enters the "Open" state.
A bidirectional stream also opens when a MAX_STREAM_DATA frame is
received. Receiving a MAX_STREAM_DATA frame implies that the remote
peer has opened the stream and is providing flow control credit. A
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MAX_STREAM_DATA frame might arrive before a STREAM or STREAM_BLOCKED
frame if packets are lost or reordered.
In the "Recv" state, the endpoint receives STREAM and STREAM_BLOCKED
frames. Incoming data is buffered and reassembled into the correct
order for delivery to the application. As data is consumed by the
application and buffer space becomes available, the endpoint sends
MAX_STREAM_DATA frames to allow the peer to send more data.
When a STREAM frame with a FIN bit is received, the final offset (see
Section 11.3) is known. The receive stream enters the "Size Known"
state. In this state, the endpoint no longer needs to send
MAX_STREAM_DATA frames, it only receives any retransmissions of
stream data.
Once all data for the stream has been received, the receive stream
enters the "Data Recvd" state. This might happen as a result of
receiving the same STREAM frame that causes the transition to "Size
Known". In this state, the endpoint has all stream data. Any STREAM
or STREAM_BLOCKED frames it receives for the stream can be discarded.
The "Data Recvd" state persists until stream data has been delivered
to the application or application protocol. Once stream data has
been delivered, the stream enters the "Data Read" state, which is a
terminal state.
Receiving a RST_STREAM frame in the "Recv" or "Size Known" states
causes the stream to enter the "Reset Recvd" state. This might cause
the delivery of stream data to the application to be interrupted.
It is possible that all stream data is received when a RST_STREAM is
received (that is, from the "Data Recvd" state). Similarly, it is
possible for remaining stream data to arrive after receiving a
RST_STREAM frame (the "Reset Recvd" state). An implementation is
able to manage this situation as they choose. Sending RST_STREAM
means that an endpoint cannot guarantee delivery of stream data;
however there is no requirement that stream data not be delivered if
a RST_STREAM is received. An implementation MAY interrupt delivery
of stream data, discard any data that was not consumed, and signal
the existence of the RST_STREAM immediately. Alternatively, the
RST_STREAM signal might be suppressed or withheld if stream data is
completely received. In the latter case, the receive stream
effectively transitions to "Data Recvd" from "Reset Recvd".
Once the application has been delivered the signal indicating that
the receive stream was reset, the receive stream transitions to the
"Reset Read" state, which is a terminal state.
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10.2.3. Permitted Frame Types
The sender of a stream sends just three frame types that affect the
state of a stream at either sender or receiver: STREAM
(Section 8.17), STREAM_BLOCKED (Section 8.11), and RST_STREAM
(Section 8.3).
A sender MUST NOT send any of these frames from a terminal state
("Data Recvd" or "Reset Recvd"). A sender MUST NOT send STREAM or
STREAM_BLOCKED after sending a RST_STREAM; that is, in the "Reset
Sent" state in addition to the terminal states. A receiver could
receive any of these frames in any state, but only due to the
possibility of delayed delivery of packets carrying them.
The receiver of a stream sends MAX_STREAM_DATA (Section 8.7) and
STOP_SENDING frames (Section 8.14).
The receiver only sends MAX_STREAM_DATA in the "Recv" state. A
receiver can send STOP_SENDING in any state where it has not received
a RST_STREAM frame; that is states other than "Reset Recvd" or "Reset
Read". However there is little value in sending a STOP_SENDING frame
after all stream data has been received in the "Data Recvd" state. A
sender could receive these frames in any state as a result of delayed
delivery of packets.
10.2.4. Bidirectional Stream States
A bidirectional stream is composed of a send stream and a receive
stream. Implementations may represent states of the bidirectional
stream as composites of send and receive stream states. The simplest
model presents the stream as "open" when either send or receive
stream is in a non-terminal state and "closed" when both send and
receive streams are in a terminal state.
Table 6 shows a more complex mapping of bidirectional stream states
that loosely correspond to the stream states in HTTP/2 [HTTP2]. This
shows that multiple states on send or receive streams are mapped to
the same composite state. Note that this is just one possibility for
such a mapping; this mapping requires that data is acknowledged
before the transition to a "closed" or "half-closed" state.
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+-----------------------+---------------------+---------------------+
| Send Stream | Receive Stream | Composite State |
+-----------------------+---------------------+---------------------+
| No Stream/Open | No Stream/Recv *1 | idle |
| | | |
| Open/Send/Data Sent | Recv/Size Known | open |
| | | |
| Open/Send/Data Sent | Data Recvd/Data | half-closed |
| | Read | (remote) |
| | | |
| Open/Send/Data Sent | Reset Recvd/Reset | half-closed |
| | Read | (remote) |
| | | |
| Data Recvd | Recv/Size Known | half-closed (local) |
| | | |
| Reset Sent/Reset | Recv/Size Known | half-closed (local) |
| Recvd | | |
| | | |
| Data Recvd | Recv/Size Known | half-closed (local) |
| | | |
| Reset Sent/Reset | Data Recvd/Data | closed |
| Recvd | Read | |
| | | |
| Reset Sent/Reset | Reset Recvd/Reset | closed |
| Recvd | Read | |
| | | |
| Data Recvd | Data Recvd/Data | closed |
| | Read | |
| | | |
| Data Recvd | Reset Recvd/Reset | closed |
| | Read | |
+-----------------------+---------------------+---------------------+
Table 6: Possible Mapping of Stream States to HTTP/2
Note (*1): A stream is considered "idle" if it has not yet been
created, or if the receive stream is in the "Recv" state without
yet having received any frames.
10.3. Solicited State Transitions
If an endpoint is no longer interested in the data it is receiving on
a stream, it MAY send a STOP_SENDING frame identifying that stream to
prompt closure of the stream in the opposite direction. This
typically indicates that the receiving application is no longer
reading data it receives from the stream, but is not a guarantee that
incoming data will be ignored.
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STREAM frames received after sending STOP_SENDING are still counted
toward the connection and stream flow-control windows, even though
these frames will be discarded upon receipt. This avoids potential
ambiguity about which STREAM frames count toward flow control.
A STOP_SENDING frame requests that the receiving endpoint send a
RST_STREAM frame. An endpoint that receives a STOP_SENDING frame
MUST send a RST_STREAM frame for that stream, and can use an error
code of STOPPING. If the STOP_SENDING frame is received on a send
stream that is already in the "Data Sent" state, a RST_STREAM frame
MAY still be sent in order to cancel retransmission of previously-
sent STREAM frames.
STOP_SENDING SHOULD only be sent for a receive stream that has not
been reset. STOP_SENDING is most useful for streams in the "Recv" or
"Size Known" states.
An endpoint is expected to send another STOP_SENDING frame if a
packet containing a previous STOP_SENDING is lost. However, once
either all stream data or a RST_STREAM frame has been received for
the stream - that is, the stream is in any state other than "Recv" or
"Size Known" - sending a STOP_SENDING frame is unnecessary.
10.4. Stream Concurrency
An endpoint limits the number of concurrently active incoming streams
by adjusting the maximum stream ID. An initial value is set in the
transport parameters (see Section 7.4.1) and is subsequently
increased by MAX_STREAM_ID frames (see Section 8.8).
The maximum stream ID is specific to each endpoint and applies only
to the peer that receives the setting. That is, clients specify the
maximum stream ID the server can initiate, and servers specify the
maximum stream ID the client can initiate. Each endpoint may respond
on streams initiated by the other peer, regardless of whether it is
permitted to initiated new streams.
Endpoints MUST NOT exceed the limit set by their peer. An endpoint
that receives a STREAM frame with an ID greater than the limit it has
sent MUST treat this as a stream error of type STREAM_ID_ERROR
(Section 12), unless this is a result of a change in the initial
offsets (see Section 7.4.2).
A receiver MUST NOT renege on an advertisement; that is, once a
receiver advertises a stream ID via a MAX_STREAM_ID frame, it MUST
NOT subsequently advertise a smaller maximum ID. A sender may
receive MAX_STREAM_ID frames out of order; a sender MUST therefore
ignore any MAX_STREAM_ID that does not increase the maximum.
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10.5. Sending and Receiving Data
Once a stream is created, endpoints may use the stream to send and
receive data. Each endpoint may send a series of STREAM frames
encapsulating data on a stream until the stream is terminated in that
direction. Streams are an ordered byte-stream abstraction, and they
have no other structure within them. STREAM frame boundaries are not
expected to be preserved in retransmissions from the sender or during
delivery to the application at the receiver.
When new data is to be sent on a stream, a sender MUST set the
encapsulating STREAM frame's offset field to the stream offset of the
first byte of this new data. The first byte of data that is sent on
a stream has the stream offset 0. The largest offset delivered on a
stream MUST be less than 2^62. A receiver MUST ensure that received
stream data is delivered to the application as an ordered byte-
stream. Data received out of order MUST be buffered for later
delivery, as long as it is not in violation of the receiver's flow
control limits.
An endpoint MUST NOT send data on any stream without ensuring that it
is within the data limits set by its peer. The cryptographic
handshake stream, Stream 0, is exempt from the connection-level data
limits established by MAX_DATA. Data on stream 0 other than the
initial cryptographic handshake message is still subject to stream-
level data limits and MAX_STREAM_DATA. This message is exempt from
flow control because it needs to be sent in a single packet
regardless of the server's flow control state. This rule applies
even for 0-RTT handshakes where the remembered value of
MAX_STREAM_DATA would not permit sending a full initial cryptographic
handshake message.
Flow control is described in detail in Section 11, and congestion
control is described in the companion document [QUIC-RECOVERY].
10.6. Stream Prioritization
Stream multiplexing has a significant effect on application
performance if resources allocated to streams are correctly
prioritized. Experience with other multiplexed protocols, such as
HTTP/2 [HTTP2], shows that effective prioritization strategies have a
significant positive impact on performance.
QUIC does not provide frames for exchanging prioritization
information. Instead it relies on receiving priority information
from the application that uses QUIC. Protocols that use QUIC are
able to define any prioritization scheme that suits their application
semantics. A protocol might define explicit messages for signaling
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priority, such as those defined in HTTP/2; it could define rules that
allow an endpoint to determine priority based on context; or it could
leave the determination to the application.
A QUIC implementation SHOULD provide ways in which an application can
indicate the relative priority of streams. When deciding which
streams to dedicate resources to, QUIC SHOULD use the information
provided by the application. Failure to account for priority of
streams can result in suboptimal performance.
Stream priority is most relevant when deciding which stream data will
be transmitted. Often, there will be limits on what can be
transmitted as a result of connection flow control or the current
congestion controller state.
Giving preference to the transmission of its own management frames
ensures that the protocol functions efficiently. That is,
prioritizing frames other than STREAM frames ensures that loss
recovery, congestion control, and flow control operate effectively.
Stream 0 MUST be prioritized over other streams prior to the
completion of the cryptographic handshake. This includes the
retransmission of the second flight of client handshake messages,
that is, the TLS Finished and any client authentication messages.
STREAM frames that are determined to be lost SHOULD be retransmitted
before sending new data, unless application priorities indicate
otherwise. Retransmitting lost stream data can fill in gaps, which
allows the peer to consume already received data and free up flow
control window.
11. Flow Control
It is necessary to limit the amount of data that a sender may have
outstanding at any time, so as to prevent a fast sender from
overwhelming a slow receiver, or to prevent a malicious sender from
consuming significant resources at a receiver. This section
describes QUIC's flow-control mechanisms.
QUIC employs a credit-based flow-control scheme similar to HTTP/2's
flow control [HTTP2]. A receiver advertises the number of octets it
is prepared to receive on a given stream and for the entire
connection. This leads to two levels of flow control in QUIC: (i)
Connection flow control, which prevents senders from exceeding a
receiver's buffer capacity for the connection, and (ii) Stream flow
control, which prevents a single stream from consuming the entire
receive buffer for a connection.
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A data receiver sends MAX_STREAM_DATA or MAX_DATA frames to the
sender to advertise additional credit. MAX_STREAM_DATA frames send
the the maximum absolute byte offset of a stream, while MAX_DATA
sends the maximum sum of the absolute byte offsets of all streams
other than stream 0.
A receiver MAY advertise a larger offset at any point by sending
MAX_DATA or MAX_STREAM_DATA frames. A receiver MUST NOT renege on an
advertisement; that is, once a receiver advertises an offset, it MUST
NOT subsequently advertise a smaller offset. A sender could receive
MAX_DATA or MAX_STREAM_DATA frames out of order; a sender MUST
therefore ignore any flow control offset that does not move the
window forward.
A receiver MUST close the connection with a FLOW_CONTROL_ERROR error
(Section 12) if the peer violates the advertised connection or stream
data limits.
A sender SHOULD send BLOCKED or STREAM_BLOCKED frames to indicate it
has data to write but is blocked by flow control limits. These
frames are expected to be sent infrequently in common cases, but they
are considered useful for debugging and monitoring purposes.
A receiver advertises credit for a stream by sending a
MAX_STREAM_DATA frame with the Stream ID set appropriately. A
receiver could use the current offset of data consumed to determine
the flow control offset to be advertised. A receiver MAY send
MAX_STREAM_DATA frames in multiple packets in order to make sure that
the sender receives an update before running out of flow control
credit, even if one of the packets is lost.
Connection flow control is a limit to the total bytes of stream data
sent in STREAM frames on all streams. A receiver advertises credit
for a connection by sending a MAX_DATA frame. A receiver maintains a
cumulative sum of bytes received on all streams, which are used to
check for flow control violations. A receiver might use a sum of
bytes consumed on all contributing streams to determine the maximum
data limit to be advertised.
11.1. Edge Cases and Other Considerations
There are some edge cases which must be considered when dealing with
stream and connection level flow control. Given enough time, both
endpoints must agree on flow control state. If one end believes it
can send more than the other end is willing to receive, the
connection will be torn down when too much data arrives.
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Conversely if a sender believes it is blocked, while endpoint B
expects more data can be received, then the connection can be in a
deadlock, with the sender waiting for a MAX_DATA or MAX_STREAM_DATA
frame which will never come.
On receipt of a RST_STREAM frame, an endpoint will tear down state
for the matching stream and ignore further data arriving on that
stream. This could result in the endpoints getting out of sync,
since the RST_STREAM frame may have arrived out of order and there
may be further bytes in flight. The data sender would have counted
the data against its connection level flow control budget, but a
receiver that has not received these bytes would not know to include
them as well. The receiver must learn the number of bytes that were
sent on the stream to make the same adjustment in its connection flow
controller.
To avoid this de-synchronization, a RST_STREAM sender MUST include
the final byte offset sent on the stream in the RST_STREAM frame. On
receiving a RST_STREAM frame, a receiver definitively knows how many
bytes were sent on that stream before the RST_STREAM frame, and the
receiver MUST use the final offset to account for all bytes sent on
the stream in its connection level flow controller.
11.1.1. Response to a RST_STREAM
RST_STREAM terminates one direction of a stream abruptly. Whether
any action or response can or should be taken on the data already
received is an application-specific issue, but it will often be the
case that upon receipt of a RST_STREAM an endpoint will choose to
stop sending data in its own direction. If the sender of a
RST_STREAM wishes to explicitly state that no future data will be
processed, that endpoint MAY send a STOP_SENDING frame at the same
time.
11.1.2. Data Limit Increments
This document leaves when and how many bytes to advertise in a
MAX_DATA or MAX_STREAM_DATA to implementations, but offers a few
considerations. These frames contribute to connection overhead.
Therefore frequently sending frames with small changes is
undesirable. At the same time, infrequent updates require larger
increments to limits if blocking is to be avoided. Thus, larger
updates require a receiver to commit to larger resource commitments.
Thus there is a tradeoff between resource commitment and overhead
when determining how large a limit is advertised.
A receiver MAY use an autotuning mechanism to tune the frequency and
amount that it increases data limits based on a roundtrip time
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estimate and the rate at which the receiving application consumes
data, similar to common TCP implementations.
11.2. Stream Limit Increment
As with flow control, this document leaves when and how many streams
to make available to a peer via MAX_STREAM_ID to implementations, but
offers a few considerations. MAX_STREAM_ID frames constitute minimal
overhead, while withholding MAX_STREAM_ID frames can prevent the peer
from using the available parallelism.
Implementations will likely want to increase the maximum stream ID as
peer-initiated streams close. A receiver MAY also advance the
maximum stream ID based on current activity, system conditions, and
other environmental factors.
11.2.1. Blocking on Flow Control
If a sender does not receive a MAX_DATA or MAX_STREAM_DATA frame when
it has run out of flow control credit, the sender will be blocked and
SHOULD send a BLOCKED or STREAM_BLOCKED frame. These frames are
expected to be useful for debugging at the receiver; they do not
require any other action. A receiver SHOULD NOT wait for a BLOCKED
or STREAM_BLOCKED frame before sending MAX_DATA or MAX_STREAM_DATA,
since doing so will mean that a sender is unable to send for an
entire round trip.
For smooth operation of the congestion controller, it is generally
considered best to not let the sender go into quiescence if
avoidable. To avoid blocking a sender, and to reasonably account for
the possibiity of loss, a receiver should send a MAX_DATA or
MAX_STREAM_DATA frame at least two roundtrips before it expects the
sender to get blocked.
A sender sends a single BLOCKED or STREAM_BLOCKED frame only once
when it reaches a data limit. A sender SHOULD NOT send multiple
BLOCKED or STREAM_BLOCKED frames for the same data limit, unless the
original frame is determined to be lost. Another BLOCKED or
STREAM_BLOCKED frame can be sent after the data limit is increased.
11.3. Stream Final Offset
The final offset is the count of the number of octets that are
transmitted on a stream. For a stream that is reset, the final
offset is carried explicitly in a RST_STREAM frame. Otherwise, the
final offset is the offset of the end of the data carried in a STREAM
frame marked with a FIN flag, or 0 in the case of incoming
unidirectional streams.
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An endpoint will know the final offset for a stream when the receive
stream enters the "Size Known" or "Reset Recvd" state.
An endpoint MUST NOT send data on a stream at or beyond the final
offset.
Once a final offset for a stream is known, it cannot change. If a
RST_STREAM or STREAM frame causes the final offset to change for a
stream, an endpoint SHOULD respond with a FINAL_OFFSET_ERROR error
(see Section 12). A receiver SHOULD treat receipt of data at or
beyond the final offset as a FINAL_OFFSET_ERROR error, even after a
stream is closed. Generating these errors is not mandatory, but only
because requiring that an endpoint generate these errors also means
that the endpoint needs to maintain the final offset state for closed
streams, which could mean a significant state commitment.
12. Error Handling
An endpoint that detects an error SHOULD signal the existence of that
error to its peer. Errors can affect an entire connection (see
Section 12.1), or a single stream (see Section 12.2).
The most appropriate error code (Section 12.3) SHOULD be included in
the frame that signals the error. Where this specification
identifies error conditions, it also identifies the error code that
is used.
A stateless reset (Section 7.9.4) is not suitable for any error that
can be signaled with a CONNECTION_CLOSE, APPLICATION_CLOSE, or
RST_STREAM frame. A stateless reset MUST NOT be used by an endpoint
that has the state necessary to send a frame on the connection.
12.1. Connection Errors
Errors that result in the connection being unusable, such as an
obvious violation of protocol semantics or corruption of state that
affects an entire connection, MUST be signaled using a
CONNECTION_CLOSE or APPLICATION_CLOSE frame (Section 8.4,
Section 8.5). An endpoint MAY close the connection in this manner
even if the error only affects a single stream.
Application protocols can signal application-specific protocol errors
using the APPLICATION_CLOSE frame. Errors that are specific to the
transport, including all those described in this document, are
carried in a CONNECTION_CLOSE frame. Other than the type of error
code they carry, these frames are identical in format and semantics.
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A CONNECTION_CLOSE or APPLICATION_CLOSE frame could be sent in a
packet that is lost. An endpoint SHOULD be prepared to retransmit a
packet containing either frame type if it receives more packets on a
terminated connection. Limiting the number of retransmissions and
the time over which this final packet is sent limits the effort
expended on terminated connections.
An endpoint that chooses not to retransmit packets containing
CONNECTION_CLOSE or APPLICATION_CLOSE risks a peer missing the first
such packet. The only mechanism available to an endpoint that
continues to receive data for a terminated connection is to use the
stateless reset process (Section 7.9.4).
An endpoint that receives an invalid CONNECTION_CLOSE or
APPLICATION_CLOSE frame MUST NOT signal the existence of the error to
its peer.
12.2. Stream Errors
If the error affects a single stream, but otherwise leaves the
connection in a recoverable state, the endpoint can send a RST_STREAM
frame (Section 8.3) with an appropriate error code to terminate just
the affected stream.
Stream 0 is critical to the functioning of the entire connection. If
stream 0 is closed with either a RST_STREAM or STREAM frame bearing
the FIN flag, an endpoint MUST generate a connection error of type
PROTOCOL_VIOLATION.
RST_STREAM MUST be instigated by the application and MUST carry an
application error code. Resetting a stream without knowledge of the
application protocol could cause the protocol to enter an
unrecoverable state. Application protocols might require certain
streams to be reliably delivered in order to guarantee consistent
state between endpoints.
12.3. Transport Error Codes
QUIC error codes are 16-bit unsigned integers.
This section lists the defined QUIC transport error codes that may be
used in a CONNECTION_CLOSE frame. These errors apply to the entire
connection.
NO_ERROR (0x0): An endpoint uses this with CONNECTION_CLOSE to
signal that the connection is being closed abruptly in the absence
of any error.
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INTERNAL_ERROR (0x1): The endpoint encountered an internal error and
cannot continue with the connection.
FLOW_CONTROL_ERROR (0x3): An endpoint received more data than it
permitted in its advertised data limits (see Section 11).
STREAM_ID_ERROR (0x4): An endpoint received a frame for a stream
identifier that exceeded its advertised maximum stream ID.
STREAM_STATE_ERROR (0x5): An endpoint received a frame for a stream
that was not in a state that permitted that frame (see
Section 10.2).
FINAL_OFFSET_ERROR (0x6): An endpoint received a STREAM frame
containing data that exceeded the previously established final
offset. Or an endpoint received a RST_STREAM frame containing a
final offset that was lower than the maximum offset of data that
was already received. Or an endpoint received a RST_STREAM frame
containing a different final offset to the one already
established.
FRAME_FORMAT_ERROR (0x7): An endpoint received a frame that was
badly formatted. For instance, an empty STREAM frame that omitted
the FIN flag, or an ACK frame that has more acknowledgment ranges
than the remainder of the packet could carry. This is a generic
error code; an endpoint SHOULD use the more specific frame format
error codes (0x1XX) if possible.
TRANSPORT_PARAMETER_ERROR (0x8): An endpoint received transport
parameters that were badly formatted, included an invalid value,
was absent even though it is mandatory, was present though it is
forbidden, or is otherwise in error.
VERSION_NEGOTIATION_ERROR (0x9): An endpoint received transport
parameters that contained version negotiation parameters that
disagreed with the version negotiation that it performed. This
error code indicates a potential version downgrade attack.
PROTOCOL_VIOLATION (0xA): An endpoint detected an error with
protocol compliance that was not covered by more specific error
codes.
UNSOLICITED_PONG (0xB): An endpoint received a PONG frame that did
not correspond to any PING frame that it previously sent.
FRAME_ERROR (0x1XX): An endpoint detected an error in a specific
frame type. The frame type is included as the last octet of the
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error code. For example, an error in a MAX_STREAM_ID frame would
be indicated with the code (0x106).
See Section 14.2 for details of registering new error codes.
12.4. Application Protocol Error Codes
Application protocol error codes are 16-bit unsigned integers, but
the management of application error codes are left to application
protocols. Application protocol error codes are used for the
RST_STREAM (Section 8.3) and APPLICATION_CLOSE (Section 8.5) frames.
There is no restriction on the use of the 16-bit error code space for
application protocols. However, QUIC reserves the error code with a
value of 0 to mean STOPPING. The application error code of STOPPING
(0) is used by the transport to cancel a stream in response to
receipt of a STOP_SENDING frame.
13. Security and Privacy Considerations
13.1. Spoofed ACK Attack
An attacker receives an STK from the server and then releases the IP
address on which it received the STK. The attacker may, in the
future, spoof this same address (which now presumably addresses a
different endpoint), and initiate a 0-RTT connection with a server on
the victim's behalf. The attacker then spoofs ACK frames to the
server which cause the server to potentially drown the victim in
data.
There are two possible mitigations to this attack. The simplest one
is that a server can unilaterally create a gap in packet-number
space. In the non-attack scenario, the client will send an ACK frame
with the larger value for largest acknowledged. In the attack
scenario, the attacker could acknowledge a packet in the gap. If the
server sees an acknowledgment for a packet that was never sent, the
connection can be aborted.
The second mitigation is that the server can require that
acknowledgments for sent packets match the encryption level of the
sent packet. This mitigation is useful if the connection has an
ephemeral forward-secure key that is generated and used for every new
connection. If a packet sent is protected with a forward-secure key,
then any acknowledgments that are received for them MUST also be
forward-secure protected. Since the attacker will not have the
forward secure key, the attacker will not be able to generate
forward-secure protected packets with ACK frames.
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13.2. Slowloris Attacks
The attacks commonly known as Slowloris [SLOWLORIS] try to keep many
connections to the target endpoint open and hold them open as long as
possible. These attacks can be executed against a QUIC endpoint by
generating the minimum amount of activity necessary to avoid being
closed for inactivity. This might involve sending small amounts of
data, gradually opening flow control windows in order to control the
sender rate, or manufacturing ACK frames that simulate a high loss
rate.
QUIC deployments SHOULD provide mitigations for the Slowloris
attacks, such as increasing the maximum number of clients the server
will allow, limiting the number of connections a single IP address is
allowed to make, imposing restrictions on the minimum transfer speed
a connection is allowed to have, and restricting the length of time
an endpoint is allowed to stay connected.
13.3. Stream Fragmentation and Reassembly Attacks
An adversarial endpoint might intentionally fragment the data on
stream buffers in order to cause disproportionate memory commitment.
An adversarial endpoint could open a stream and send some STREAM
frames containing arbitrary fragments of the stream content.
The attack is mitigated if flow control windows correspond to
available memory. However, some receivers will over-commit memory
and advertise flow control offsets in the aggregate that exceed
actual available memory. The over-commitment strategy can lead to
better performance when endpoints are well behaved, but renders
endpoints vulnerable to the stream fragmentation attack.
QUIC deployments SHOULD provide mitigations against the stream
fragmentation attack. Mitigations could consist of avoiding over-
committing memory, delaying reassembly of STREAM frames, implementing
heuristics based on the age and duration of reassembly holes, or some
combination.
13.4. Stream Commitment Attack
An adversarial endpoint can open lots of streams, exhausting state on
an endpoint. The adversarial endpoint could repeat the process on a
large number of connections, in a manner similar to SYN flooding
attacks in TCP.
Normally, clients will open streams sequentially, as explained in
Section 10.1. However, when several streams are initiated at short
intervals, transmission error may cause STREAM DATA frames opening
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streams to be received out of sequence. A receiver is obligated to
open intervening streams if a higher-numbered stream ID is received.
Thus, on a new connection, opening stream 2000001 opens 1 million
streams, as required by the specification.
The number of active streams is limited by the concurrent stream
limit transport parameter, as explained in Section 10.4. If chosen
judisciously, this limit mitigates the effect of the stream
commitment attack. However, setting the limit too low could affect
performance when applications expect to open large number of streams.
14. IANA Considerations
14.1. QUIC Transport Parameter Registry
IANA [SHALL add/has added] a registry for "QUIC Transport Parameters"
under a "QUIC Protocol" heading.
The "QUIC Transport Parameters" registry governs a 16-bit space.
This space is split into two spaces that are governed by different
policies. Values with the first byte in the range 0x00 to 0xfe (in
hexadecimal) are assigned via the Specification Required policy
[RFC8126]. Values with the first byte 0xff are reserved for Private
Use [RFC8126].
Registrations MUST include the following fields:
Value: The numeric value of the assignment (registrations will be
between 0x0000 and 0xfeff).
Parameter Name: A short mnemonic for the parameter.
Specification: A reference to a publicly available specification for
the value.
The nominated expert(s) verify that a specification exists and is
readily accessible. The expert(s) are encouraged to be biased
towards approving registrations unless they are abusive, frivolous,
or actively harmful (not merely aesthetically displeasing, or
architecturally dubious).
The initial contents of this registry are shown in Table 7.
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+--------+----------------------------+---------------+
| Value | Parameter Name | Specification |
+--------+----------------------------+---------------+
| 0x0000 | initial_max_stream_data | Section 7.4.1 |
| | | |
| 0x0001 | initial_max_data | Section 7.4.1 |
| | | |
| 0x0002 | initial_max_stream_id_bidi | Section 7.4.1 |
| | | |
| 0x0003 | idle_timeout | Section 7.4.1 |
| | | |
| 0x0004 | omit_connection_id | Section 7.4.1 |
| | | |
| 0x0005 | max_packet_size | Section 7.4.1 |
| | | |
| 0x0006 | stateless_reset_token | Section 7.4.1 |
| | | |
| 0x0007 | ack_delay_exponent | Section 7.4.1 |
| | | |
| 0x0008 | initial_max_stream_id_uni | Section 7.4.1 |
+--------+----------------------------+---------------+
Table 7: Initial QUIC Transport Parameters Entries
14.2. QUIC Transport Error Codes Registry
IANA [SHALL add/has added] a registry for "QUIC Transport Error
Codes" under a "QUIC Protocol" heading.
The "QUIC Transport Error Codes" registry governs a 16-bit space.
This space is split into two spaces that are governed by different
policies. Values with the first byte in the range 0x00 to 0xfe (in
hexadecimal) are assigned via the Specification Required policy
[RFC8126]. Values with the first byte 0xff are reserved for Private
Use [RFC8126].
Registrations MUST include the following fields:
Value: The numeric value of the assignment (registrations will be
between 0x0000 and 0xfeff).
Code: A short mnemonic for the parameter.
Description: A brief description of the error code semantics, which
MAY be a summary if a specification reference is provided.
Specification: A reference to a publicly available specification for
the value.
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The initial contents of this registry are shown in Table 8. Note
that FRAME_ERROR takes the range from 0x100 to 0x1FF and private use
occupies the range from 0xFE00 to 0xFFFF.
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+-----------+------------------------+---------------+--------------+
| Value | Error | Description | Specificatio |
| | | | n |
+-----------+------------------------+---------------+--------------+
| 0x0 | NO_ERROR | No error | Section 12.3 |
| | | | |
| 0x1 | INTERNAL_ERROR | Implementatio | Section 12.3 |
| | | n error | |
| | | | |
| 0x3 | FLOW_CONTROL_ERROR | Flow control | Section 12.3 |
| | | error | |
| | | | |
| 0x4 | STREAM_ID_ERROR | Invalid | Section 12.3 |
| | | stream ID | |
| | | | |
| 0x5 | STREAM_STATE_ERROR | Frame | Section 12.3 |
| | | received in | |
| | | invalid | |
| | | stream state | |
| | | | |
| 0x6 | FINAL_OFFSET_ERROR | Change to | Section 12.3 |
| | | final stream | |
| | | offset | |
| | | | |
| 0x7 | FRAME_FORMAT_ERROR | Generic frame | Section 12.3 |
| | | format error | |
| | | | |
| 0x8 | TRANSPORT_PARAMETER_ER | Error in | Section 12.3 |
| | ROR | transport | |
| | | parameters | |
| | | | |
| 0x9 | VERSION_NEGOTIATION_ER | Version | Section 12.3 |
| | ROR | negotiation | |
| | | failure | |
| | | | |
| 0xA | PROTOCOL_VIOLATION | Generic | Section 12.3 |
| | | protocol | |
| | | violation | |
| | | | |
| 0xB | UNSOLICITED_PONG | Unsolicited | Section 12.3 |
| | | PONG frame | |
| | | | |
| 0x100-0x1 | FRAME_ERROR | Specific | Section 12.3 |
| FF | | frame format | |
| | | error | |
+-----------+------------------------+---------------+--------------+
Table 8: Initial QUIC Transport Error Codes Entries
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15. References
15.1. Normative References
[I-D.ietf-tls-tls13]
Rescorla, E., "The Transport Layer Security (TLS) Protocol
Version 1.3", draft-ietf-tls-tls13-22 (work in progress),
November 2017.
[PLPMTUD] Mathis, M. and J. Heffner, "Packetization Layer Path MTU
Discovery", RFC 4821, DOI 10.17487/RFC4821, March 2007,
<https://www.rfc-editor.org/info/rfc4821>.
[PMTUDv4] Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191,
DOI 10.17487/RFC1191, November 1990,
<https://www.rfc-editor.org/info/rfc1191>.
[PMTUDv6] McCann, J., Deering, S., Mogul, J., and R. Hinden, Ed.,
"Path MTU Discovery for IP version 6", STD 87, RFC 8201,
DOI 10.17487/RFC8201, July 2017,
<https://www.rfc-editor.org/info/rfc8201>.
[QUIC-RECOVERY]
Iyengar, J., Ed. and I. Swett, Ed., "QUIC Loss Detection
and Congestion Control", draft-ietf-quic-recovery-00 (work
in progress), December 2017.
[QUIC-TLS]
Thomson, M., Ed. and S. Turner, Ed., "Using Transport
Layer Security (TLS) to Secure QUIC", draft-ietf-quic-
tls-00 (work in progress), December 2017.
[RFC1191] Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191,
DOI 10.17487/RFC1191, November 1990,
<https://www.rfc-editor.org/info/rfc1191>.
[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/info/rfc2119>.
[RFC3629] Yergeau, F., "UTF-8, a transformation format of ISO
10646", STD 63, RFC 3629, DOI 10.17487/RFC3629, November
2003, <https://www.rfc-editor.org/info/rfc3629>.
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[RFC4086] Eastlake 3rd, D., Schiller, J., and S. Crocker,
"Randomness Requirements for Security", BCP 106, RFC 4086,
DOI 10.17487/RFC4086, June 2005,
<https://www.rfc-editor.org/info/rfc4086>.
[RFC8126] Cotton, M., Leiba, B., and T. Narten, "Guidelines for
Writing an IANA Considerations Section in RFCs", BCP 26,
RFC 8126, DOI 10.17487/RFC8126, June 2017,
<https://www.rfc-editor.org/info/rfc8126>.
[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>.
15.2. Informative References
[EARLY-DESIGN]
Roskind, J., "QUIC: Multiplexed Transport Over UDP",
December 2013, <https://goo.gl/dMVtFi>.
[HTTP2] 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/info/rfc7540>.
[RFC2104] Krawczyk, H., Bellare, M., and R. Canetti, "HMAC: Keyed-
Hashing for Message Authentication", RFC 2104,
DOI 10.17487/RFC2104, February 1997,
<https://www.rfc-editor.org/info/rfc2104>.
[RFC2360] Scott, G., "Guide for Internet Standards Writers", BCP 22,
RFC 2360, DOI 10.17487/RFC2360, June 1998,
<https://www.rfc-editor.org/info/rfc2360>.
[RFC4787] Audet, F., Ed. and C. Jennings, "Network Address
Translation (NAT) Behavioral Requirements for Unicast
UDP", BCP 127, RFC 4787, DOI 10.17487/RFC4787, January
2007, <https://www.rfc-editor.org/info/rfc4787>.
[RFC5869] Krawczyk, H. and P. Eronen, "HMAC-based Extract-and-Expand
Key Derivation Function (HKDF)", RFC 5869,
DOI 10.17487/RFC5869, May 2010,
<https://www.rfc-editor.org/info/rfc5869>.
[RFC6824] Ford, A., Raiciu, C., Handley, M., and O. Bonaventure,
"TCP Extensions for Multipath Operation with Multiple
Addresses", RFC 6824, DOI 10.17487/RFC6824, January 2013,
<https://www.rfc-editor.org/info/rfc6824>.
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[RFC7301] Friedl, S., Popov, A., Langley, A., and E. Stephan,
"Transport Layer Security (TLS) Application-Layer Protocol
Negotiation Extension", RFC 7301, DOI 10.17487/RFC7301,
July 2014, <https://www.rfc-editor.org/info/rfc7301>.
[SLOWLORIS]
RSnake Hansen, R., "Welcome to Slowloris...", June 2009,
<https://web.archive.org/web/20150315054838/
http://ha.ckers.org/slowloris/>.
[SST] Ford, B., "Structured streams", ACM SIGCOMM Computer
Communication Review Vol. 37, pp. 361,
DOI 10.1145/1282427.1282421, October 2007.
15.3. URIs
[1] https://mailarchive.ietf.org/arch/search/?email_list=quic
[2] https://github.com/quicwg
[3] https://github.com/quicwg/base-drafts/labels/-transport
[4] https://github.com/quicwg/base-drafts/wiki/QUIC-Versions
Appendix A. Contributors
The original authors of this specification were Ryan Hamilton, Jana
Iyengar, Ian Swett, and Alyssa Wilk.
The original design and rationale behind this protocol draw
significantly from work by Jim Roskind [EARLY-DESIGN]. In
alphabetical order, the contributors to the pre-IETF QUIC project at
Google are: Britt Cyr, Jeremy Dorfman, Ryan Hamilton, Jana Iyengar,
Fedor Kouranov, Charles Krasic, Jo Kulik, Adam Langley, Jim Roskind,
Robbie Shade, Satyam Shekhar, Cherie Shi, Ian Swett, Raman Tenneti,
Victor Vasiliev, Antonio Vicente, Patrik Westin, Alyssa Wilk, Dale
Worley, Fan Yang, Dan Zhang, Daniel Ziegler.
Appendix B. Acknowledgments
Special thanks are due to the following for helping shape pre-IETF
QUIC and its deployment: Chris Bentzel, Misha Efimov, Roberto Peon,
Alistair Riddoch, Siddharth Vijayakrishnan, and Assar Westerlund.
This document has benefited immensely from various private
discussions and public ones on the quic@ietf.org and proto-
quic@chromium.org mailing lists. Our thanks to all.
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Appendix C. Change Log
*RFC Editor's Note:* Please remove this section prior to
publication of a final version of this document.
Issue and pull request numbers are listed with a leading octothorp.
C.1. Since draft-ietf-quic-transport-07
o Employ variable-length integer encodings throughout (#595)
o Draining period can terminate early (#869)
C.2. Since draft-ietf-quic-transport-06
o Replaced FNV-1a with AES-GCM for all "Cleartext" packets (#554)
o Split error code space between application and transport (#485)
o Stateless reset token moved to end (#820)
o 1-RTT-protected long header types removed (#848)
o No acknowledgments during draining period (#852)
o Remove "application close" as a separate close type (#854)
o Remove timestamps from the ACK frame (#841)
o Require transport parameters to only appear once (#792)
C.3. Since draft-ietf-quic-transport-05
o Stateless token is server-only (#726)
o Refactor section on connection termination (#733, #748, #328,
#177)
o Limit size of Version Negotiation packet (#585)
o Clarify when and what to ack (#736)
o Renamed STREAM_ID_NEEDED to STREAM_ID_BLOCKED
o Clarify Keep-alive requirements (#729)
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C.4. Since draft-ietf-quic-transport-04
o Introduce STOP_SENDING frame, RST_STREAM only resets in one
direction (#165)
o Removed GOAWAY; application protocols are responsible for graceful
shutdown (#696)
o Reduced the number of error codes (#96, #177, #184, #211)
o Version validation fields can't move or change (#121)
o Removed versions from the transport parameters in a
NewSessionTicket message (#547)
o Clarify the meaning of "bytes in flight" (#550)
o Public reset is now stateless reset and not visible to the path
(#215)
o Reordered bits and fields in STREAM frame (#620)
o Clarifications to the stream state machine (#572, #571)
o Increased the maximum length of the Largest Acknowledged field in
ACK frames to 64 bits (#629)
o truncate_connection_id is renamed to omit_connection_id (#659)
o CONNECTION_CLOSE terminates the connection like TCP RST (#330,
#328)
o Update labels used in HKDF-Expand-Label to match TLS 1.3 (#642)
C.5. Since draft-ietf-quic-transport-03
o Change STREAM and RST_STREAM layout
o Add MAX_STREAM_ID settings
C.6. Since draft-ietf-quic-transport-02
o The size of the initial packet payload has a fixed minimum (#267,
#472)
o Define when Version Negotiation packets are ignored (#284, #294,
#241, #143, #474)
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o The 64-bit FNV-1a algorithm is used for integrity protection of
unprotected packets (#167, #480, #481, #517)
o Rework initial packet types to change how the connection ID is
chosen (#482, #442, #493)
o No timestamps are forbidden in unprotected packets (#542, #429)
o Cryptographic handshake is now on stream 0 (#456)
o Remove congestion control exemption for cryptographic handshake
(#248, #476)
o Version 1 of QUIC uses TLS; a new version is needed to use a
different handshake protocol (#516)
o STREAM frames have a reduced number of offset lengths (#543, #430)
o Split some frames into separate connection- and stream- level
frames (#443)
* WINDOW_UPDATE split into MAX_DATA and MAX_STREAM_DATA (#450)
* BLOCKED split to match WINDOW_UPDATE split (#454)
* Define STREAM_ID_NEEDED frame (#455)
o A NEW_CONNECTION_ID frame supports connection migration without
linkability (#232, #491, #496)
o Transport parameters for 0-RTT are retained from a previous
connection (#405, #513, #512)
* A client in 0-RTT no longer required to reset excess streams
(#425, #479)
o Expanded security considerations (#440, #444, #445, #448)
C.7. Since draft-ietf-quic-transport-01
o Defined short and long packet headers (#40, #148, #361)
o Defined a versioning scheme and stable fields (#51, #361)
o Define reserved version values for "greasing" negotiation (#112,
#278)
o The initial packet number is randomized (#35, #283)
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o Narrow the packet number encoding range requirement (#67, #286,
#299, #323, #356)
o Defined client address validation (#52, #118, #120, #275)
o Define transport parameters as a TLS extension (#49, #122)
o SCUP and COPT parameters are no longer valid (#116, #117)
o Transport parameters for 0-RTT are either remembered from before,
or assume default values (#126)
o The server chooses connection IDs in its final flight (#119, #349,
#361)
o The server echoes the Connection ID and packet number fields when
sending a Version Negotiation packet (#133, #295, #244)
o Defined a minimum packet size for the initial handshake packet
from the client (#69, #136, #139, #164)
o Path MTU Discovery (#64, #106)
o The initial handshake packet from the client needs to fit in a
single packet (#338)
o Forbid acknowledgment of packets containing only ACK and PADDING
(#291)
o Require that frames are processed when packets are acknowledged
(#381, #341)
o Removed the STOP_WAITING frame (#66)
o Don't require retransmission of old timestamps for lost ACK frames
(#308)
o Clarified that frames are not retransmitted, but the information
in them can be (#157, #298)
o Error handling definitions (#335)
o Split error codes into four sections (#74)
o Forbid the use of Public Reset where CONNECTION_CLOSE is possible
(#289)
o Define packet protection rules (#336)
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o Require that stream be entirely delivered or reset, including
acknowledgment of all STREAM frames or the RST_STREAM, before it
closes (#381)
o Remove stream reservation from state machine (#174, #280)
o Only stream 1 does not contribute to connection-level flow control
(#204)
o Stream 1 counts towards the maximum concurrent stream limit (#201,
#282)
o Remove connection-level flow control exclusion for some streams
(except 1) (#246)
o RST_STREAM affects connection-level flow control (#162, #163)
o Flow control accounting uses the maximum data offset on each
stream, rather than bytes received (#378)
o Moved length-determining fields to the start of STREAM and ACK
(#168, #277)
o Added the ability to pad between frames (#158, #276)
o Remove error code and reason phrase from GOAWAY (#352, #355)
o GOAWAY includes a final stream number for both directions (#347)
o Error codes for RST_STREAM and CONNECTION_CLOSE are now at a
consistent offset (#249)
o Defined priority as the responsibility of the application protocol
(#104, #303)
C.8. Since draft-ietf-quic-transport-00
o Replaced DIVERSIFICATION_NONCE flag with KEY_PHASE flag
o Defined versioning
o Reworked description of packet and frame layout
o Error code space is divided into regions for each component
o Use big endian for all numeric values
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C.9. Since draft-hamilton-quic-transport-protocol-01
o Adopted as base for draft-ietf-quic-tls
o Updated authors/editors list
o Added IANA Considerations section
o Moved Contributors and Acknowledgments to appendices
Authors' Addresses
Jana Iyengar (editor)
Google
Email: jri@google.com
Martin Thomson (editor)
Mozilla
Email: martin.thomson@gmail.com
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