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QUIC J. Iyengar, Ed.
Internet-Draft Fastly
Intended status: Standards Track M. Thomson, Ed.
Expires: 21 November 2020 Mozilla
20 May 2020
QUIC: A UDP-Based Multiplexed and Secure Transport
draft-ietf-quic-transport-28
Abstract
This document defines the core of the QUIC transport protocol.
Accompanying documents describe QUIC's loss detection and congestion
control and the use of TLS for key negotiation.
Note to Readers
Discussion of this draft takes place on the QUIC working group
mailing list (quic@ietf.org (mailto:quic@ietf.org)), which is
archived at https://mailarchive.ietf.org/arch/search/?email_list=quic
Working Group information can be found at https://github.com/quicwg;
source code and issues list for this draft can be found at
https://github.com/quicwg/base-drafts/labels/-transport.
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 21 November 2020.
Copyright Notice
Copyright (c) 2020 IETF Trust and the persons identified as the
document authors. All rights reserved.
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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 . . . . . . . . . . . . . . . . . . . . . . . . 6
1.1. Document Structure . . . . . . . . . . . . . . . . . . . 7
1.2. Terms and Definitions . . . . . . . . . . . . . . . . . . 8
1.3. Notational Conventions . . . . . . . . . . . . . . . . . 9
2. Streams . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
2.1. Stream Types and Identifiers . . . . . . . . . . . . . . 11
2.2. Sending and Receiving Data . . . . . . . . . . . . . . . 12
2.3. Stream Prioritization . . . . . . . . . . . . . . . . . . 12
2.4. Required Operations on Streams . . . . . . . . . . . . . 13
3. Stream States . . . . . . . . . . . . . . . . . . . . . . . . 13
3.1. Sending Stream States . . . . . . . . . . . . . . . . . . 14
3.2. Receiving Stream States . . . . . . . . . . . . . . . . . 17
3.3. Permitted Frame Types . . . . . . . . . . . . . . . . . . 19
3.4. Bidirectional Stream States . . . . . . . . . . . . . . . 20
3.5. Solicited State Transitions . . . . . . . . . . . . . . . 21
4. Flow Control . . . . . . . . . . . . . . . . . . . . . . . . 23
4.1. Data Flow Control . . . . . . . . . . . . . . . . . . . . 23
4.2. Flow Credit Increments . . . . . . . . . . . . . . . . . 24
4.3. Handling Stream Cancellation . . . . . . . . . . . . . . 25
4.4. Stream Final Size . . . . . . . . . . . . . . . . . . . . 26
4.5. Controlling Concurrency . . . . . . . . . . . . . . . . . 26
5. Connections . . . . . . . . . . . . . . . . . . . . . . . . . 27
5.1. Connection ID . . . . . . . . . . . . . . . . . . . . . . 27
5.1.1. Issuing Connection IDs . . . . . . . . . . . . . . . 29
5.1.2. Consuming and Retiring Connection IDs . . . . . . . . 30
5.2. Matching Packets to Connections . . . . . . . . . . . . . 31
5.2.1. Client Packet Handling . . . . . . . . . . . . . . . 32
5.2.2. Server Packet Handling . . . . . . . . . . . . . . . 32
5.2.3. Considerations for Simple Load Balancers . . . . . . 33
5.3. Life of a QUIC Connection . . . . . . . . . . . . . . . . 33
5.4. Required Operations on Connections . . . . . . . . . . . 34
6. Version Negotiation . . . . . . . . . . . . . . . . . . . . . 35
6.1. Sending Version Negotiation Packets . . . . . . . . . . . 35
6.2. Handling Version Negotiation Packets . . . . . . . . . . 36
6.2.1. Version Negotiation Between Draft Versions . . . . . 36
6.3. Using Reserved Versions . . . . . . . . . . . . . . . . . 37
7. Cryptographic and Transport Handshake . . . . . . . . . . . . 37
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7.1. Example Handshake Flows . . . . . . . . . . . . . . . . . 38
7.2. Negotiating Connection IDs . . . . . . . . . . . . . . . 39
7.3. Authenticating Connection IDs . . . . . . . . . . . . . . 41
7.4. Transport Parameters . . . . . . . . . . . . . . . . . . 43
7.4.1. Values of Transport Parameters for 0-RTT . . . . . . 43
7.4.2. New Transport Parameters . . . . . . . . . . . . . . 45
7.5. Cryptographic Message Buffering . . . . . . . . . . . . . 45
8. Address Validation . . . . . . . . . . . . . . . . . . . . . 46
8.1. Address Validation During Connection Establishment . . . 46
8.1.1. Token Construction . . . . . . . . . . . . . . . . . 47
8.1.2. Address Validation using Retry Packets . . . . . . . 47
8.1.3. Address Validation for Future Connections . . . . . . 48
8.1.4. Address Validation Token Integrity . . . . . . . . . 51
8.2. Path Validation . . . . . . . . . . . . . . . . . . . . . 51
8.3. Initiating Path Validation . . . . . . . . . . . . . . . 52
8.4. Path Validation Responses . . . . . . . . . . . . . . . . 52
8.5. Successful Path Validation . . . . . . . . . . . . . . . 53
8.6. Failed Path Validation . . . . . . . . . . . . . . . . . 53
9. Connection Migration . . . . . . . . . . . . . . . . . . . . 54
9.1. Probing a New Path . . . . . . . . . . . . . . . . . . . 55
9.2. Initiating Connection Migration . . . . . . . . . . . . . 55
9.3. Responding to Connection Migration . . . . . . . . . . . 56
9.3.1. Peer Address Spoofing . . . . . . . . . . . . . . . . 56
9.3.2. On-Path Address Spoofing . . . . . . . . . . . . . . 57
9.3.3. Off-Path Packet Forwarding . . . . . . . . . . . . . 58
9.4. Loss Detection and Congestion Control . . . . . . . . . . 59
9.5. Privacy Implications of Connection Migration . . . . . . 60
9.6. Server's Preferred Address . . . . . . . . . . . . . . . 61
9.6.1. Communicating a Preferred Address . . . . . . . . . . 61
9.6.2. Responding to Connection Migration . . . . . . . . . 62
9.6.3. Interaction of Client Migration and Preferred
Address . . . . . . . . . . . . . . . . . . . . . . . 62
9.7. Use of IPv6 Flow-Label and Migration . . . . . . . . . . 63
10. Connection Termination . . . . . . . . . . . . . . . . . . . 63
10.1. Closing and Draining Connection States . . . . . . . . . 64
10.2. Idle Timeout . . . . . . . . . . . . . . . . . . . . . . 65
10.3. Immediate Close . . . . . . . . . . . . . . . . . . . . 66
10.3.1. Immediate Close During the Handshake . . . . . . . . 67
10.4. Stateless Reset . . . . . . . . . . . . . . . . . . . . 69
10.4.1. Detecting a Stateless Reset . . . . . . . . . . . . 71
10.4.2. Calculating a Stateless Reset Token . . . . . . . . 72
10.4.3. Looping . . . . . . . . . . . . . . . . . . . . . . 73
11. Error Handling . . . . . . . . . . . . . . . . . . . . . . . 74
11.1. Connection Errors . . . . . . . . . . . . . . . . . . . 74
11.2. Stream Errors . . . . . . . . . . . . . . . . . . . . . 75
12. Packets and Frames . . . . . . . . . . . . . . . . . . . . . 75
12.1. Protected Packets . . . . . . . . . . . . . . . . . . . 76
12.2. Coalescing Packets . . . . . . . . . . . . . . . . . . . 76
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12.3. Packet Numbers . . . . . . . . . . . . . . . . . . . . . 77
12.4. Frames and Frame Types . . . . . . . . . . . . . . . . . 78
13. Packetization and Reliability . . . . . . . . . . . . . . . . 81
13.1. Packet Processing . . . . . . . . . . . . . . . . . . . 82
13.2. Generating Acknowledgements . . . . . . . . . . . . . . 82
13.2.1. Sending ACK Frames . . . . . . . . . . . . . . . . . 83
13.2.2. Managing ACK Ranges . . . . . . . . . . . . . . . . 84
13.2.3. Receiver Tracking of ACK Frames . . . . . . . . . . 85
13.2.4. Limiting ACK Ranges . . . . . . . . . . . . . . . . 85
13.2.5. Measuring and Reporting Host Delay . . . . . . . . . 86
13.2.6. ACK Frames and Packet Protection . . . . . . . . . . 86
13.3. Retransmission of Information . . . . . . . . . . . . . 87
13.4. Explicit Congestion Notification . . . . . . . . . . . . 89
13.4.1. ECN Counts . . . . . . . . . . . . . . . . . . . . . 90
13.4.2. ECN Validation . . . . . . . . . . . . . . . . . . . 90
14. Packet Size . . . . . . . . . . . . . . . . . . . . . . . . . 92
14.1. Path Maximum Transmission Unit (PMTU) . . . . . . . . . 93
14.2. ICMP Packet Too Big Messages . . . . . . . . . . . . . . 94
14.3. Datagram Packetization Layer PMTU Discovery . . . . . . 95
14.3.1. PMTU Probes Containing Source Connection ID . . . . 95
15. Versions . . . . . . . . . . . . . . . . . . . . . . . . . . 96
16. Variable-Length Integer Encoding . . . . . . . . . . . . . . 97
17. Packet Formats . . . . . . . . . . . . . . . . . . . . . . . 97
17.1. Packet Number Encoding and Decoding . . . . . . . . . . 98
17.2. Long Header Packets . . . . . . . . . . . . . . . . . . 99
17.2.1. Version Negotiation Packet . . . . . . . . . . . . . 101
17.2.2. Initial Packet . . . . . . . . . . . . . . . . . . . 103
17.2.3. 0-RTT . . . . . . . . . . . . . . . . . . . . . . . 105
17.2.4. Handshake Packet . . . . . . . . . . . . . . . . . . 106
17.2.5. Retry Packet . . . . . . . . . . . . . . . . . . . . 107
17.3. Short Header Packets . . . . . . . . . . . . . . . . . . 109
17.3.1. Latency Spin Bit . . . . . . . . . . . . . . . . . . 111
18. Transport Parameter Encoding . . . . . . . . . . . . . . . . 112
18.1. Reserved Transport Parameters . . . . . . . . . . . . . 113
18.2. Transport Parameter Definitions . . . . . . . . . . . . 113
19. Frame Types and Formats . . . . . . . . . . . . . . . . . . . 117
19.1. PADDING Frame . . . . . . . . . . . . . . . . . . . . . 117
19.2. PING Frame . . . . . . . . . . . . . . . . . . . . . . . 118
19.3. ACK Frames . . . . . . . . . . . . . . . . . . . . . . . 118
19.3.1. ACK Ranges . . . . . . . . . . . . . . . . . . . . . 120
19.3.2. ECN Counts . . . . . . . . . . . . . . . . . . . . . 121
19.4. RESET_STREAM Frame . . . . . . . . . . . . . . . . . . . 122
19.5. STOP_SENDING Frame . . . . . . . . . . . . . . . . . . . 123
19.6. CRYPTO Frame . . . . . . . . . . . . . . . . . . . . . . 123
19.7. NEW_TOKEN Frame . . . . . . . . . . . . . . . . . . . . 124
19.8. STREAM Frames . . . . . . . . . . . . . . . . . . . . . 125
19.9. MAX_DATA Frame . . . . . . . . . . . . . . . . . . . . . 127
19.10. MAX_STREAM_DATA Frame . . . . . . . . . . . . . . . . . 127
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19.11. MAX_STREAMS Frames . . . . . . . . . . . . . . . . . . . 128
19.12. DATA_BLOCKED Frame . . . . . . . . . . . . . . . . . . . 129
19.13. STREAM_DATA_BLOCKED Frame . . . . . . . . . . . . . . . 130
19.14. STREAMS_BLOCKED Frames . . . . . . . . . . . . . . . . . 130
19.15. NEW_CONNECTION_ID Frame . . . . . . . . . . . . . . . . 131
19.16. RETIRE_CONNECTION_ID Frame . . . . . . . . . . . . . . . 132
19.17. PATH_CHALLENGE Frame . . . . . . . . . . . . . . . . . . 133
19.18. PATH_RESPONSE Frame . . . . . . . . . . . . . . . . . . 134
19.19. CONNECTION_CLOSE Frames . . . . . . . . . . . . . . . . 134
19.20. HANDSHAKE_DONE frame . . . . . . . . . . . . . . . . . . 135
19.21. Extension Frames . . . . . . . . . . . . . . . . . . . . 136
20. Transport Error Codes . . . . . . . . . . . . . . . . . . . . 136
20.1. Application Protocol Error Codes . . . . . . . . . . . . 138
21. Security Considerations . . . . . . . . . . . . . . . . . . . 138
21.1. Handshake Denial of Service . . . . . . . . . . . . . . 138
21.2. Amplification Attack . . . . . . . . . . . . . . . . . . 139
21.3. Optimistic ACK Attack . . . . . . . . . . . . . . . . . 140
21.4. Slowloris Attacks . . . . . . . . . . . . . . . . . . . 140
21.5. Stream Fragmentation and Reassembly Attacks . . . . . . 140
21.6. Stream Commitment Attack . . . . . . . . . . . . . . . . 141
21.7. Peer Denial of Service . . . . . . . . . . . . . . . . . 141
21.8. Explicit Congestion Notification Attacks . . . . . . . . 142
21.9. Stateless Reset Oracle . . . . . . . . . . . . . . . . . 142
21.10. Version Downgrade . . . . . . . . . . . . . . . . . . . 143
21.11. Targeted Attacks by Routing . . . . . . . . . . . . . . 143
21.12. Overview of Security Properties . . . . . . . . . . . . 143
21.12.1. Handshake . . . . . . . . . . . . . . . . . . . . . 144
21.12.2. Protected Packets . . . . . . . . . . . . . . . . . 145
21.12.3. Connection Migration . . . . . . . . . . . . . . . 146
22. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 150
22.1. Registration Policies for QUIC Registries . . . . . . . 150
22.1.1. Provisional Registrations . . . . . . . . . . . . . 150
22.1.2. Selecting Codepoints . . . . . . . . . . . . . . . . 151
22.1.3. Reclaiming Provisional Codepoints . . . . . . . . . 152
22.1.4. Permanent Registrations . . . . . . . . . . . . . . 153
22.2. QUIC Transport Parameter Registry . . . . . . . . . . . 153
22.3. QUIC Frame Type Registry . . . . . . . . . . . . . . . . 154
22.4. QUIC Transport Error Codes Registry . . . . . . . . . . 155
23. References . . . . . . . . . . . . . . . . . . . . . . . . . 157
23.1. Normative References . . . . . . . . . . . . . . . . . . 157
23.2. Informative References . . . . . . . . . . . . . . . . . 158
Appendix A. Sample Packet Number Decoding Algorithm . . . . . . 160
Appendix B. Sample ECN Validation Algorithm . . . . . . . . . . 161
Appendix C. Change Log . . . . . . . . . . . . . . . . . . . . . 162
C.1. Since draft-ietf-quic-transport-27 . . . . . . . . . . . 162
C.2. Since draft-ietf-quic-transport-26 . . . . . . . . . . . 163
C.3. Since draft-ietf-quic-transport-25 . . . . . . . . . . . 163
C.4. Since draft-ietf-quic-transport-24 . . . . . . . . . . . 163
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C.5. Since draft-ietf-quic-transport-23 . . . . . . . . . . . 165
C.6. Since draft-ietf-quic-transport-22 . . . . . . . . . . . 165
C.7. Since draft-ietf-quic-transport-21 . . . . . . . . . . . 166
C.8. Since draft-ietf-quic-transport-20 . . . . . . . . . . . 167
C.9. Since draft-ietf-quic-transport-19 . . . . . . . . . . . 167
C.10. Since draft-ietf-quic-transport-18 . . . . . . . . . . . 168
C.11. Since draft-ietf-quic-transport-17 . . . . . . . . . . . 168
C.12. Since draft-ietf-quic-transport-16 . . . . . . . . . . . 169
C.13. Since draft-ietf-quic-transport-15 . . . . . . . . . . . 170
C.14. Since draft-ietf-quic-transport-14 . . . . . . . . . . . 170
C.15. Since draft-ietf-quic-transport-13 . . . . . . . . . . . 171
C.16. Since draft-ietf-quic-transport-12 . . . . . . . . . . . 172
C.17. Since draft-ietf-quic-transport-11 . . . . . . . . . . . 172
C.18. Since draft-ietf-quic-transport-10 . . . . . . . . . . . 173
C.19. Since draft-ietf-quic-transport-09 . . . . . . . . . . . 173
C.20. Since draft-ietf-quic-transport-08 . . . . . . . . . . . 174
C.21. Since draft-ietf-quic-transport-07 . . . . . . . . . . . 175
C.22. Since draft-ietf-quic-transport-06 . . . . . . . . . . . 176
C.23. Since draft-ietf-quic-transport-05 . . . . . . . . . . . 176
C.24. Since draft-ietf-quic-transport-04 . . . . . . . . . . . 176
C.25. Since draft-ietf-quic-transport-03 . . . . . . . . . . . 177
C.26. Since draft-ietf-quic-transport-02 . . . . . . . . . . . 177
C.27. Since draft-ietf-quic-transport-01 . . . . . . . . . . . 178
C.28. Since draft-ietf-quic-transport-00 . . . . . . . . . . . 180
C.29. Since draft-hamilton-quic-transport-protocol-01 . . . . . 180
Contributors . . . . . . . . . . . . . . . . . . . . . . . . . . 180
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 182
1. Introduction
QUIC is a multiplexed and secure general-purpose transport protocol
that provides:
* Stream multiplexing
* Stream and connection-level flow control
* Low-latency connection establishment
* Connection migration and resilience to NAT rebinding
* Authenticated and encrypted header and payload
QUIC uses UDP as a substrate to avoid requiring changes to legacy
client operating systems and middleboxes. QUIC authenticates all of
its headers and encrypts most of the data it exchanges, including its
signaling, to avoid incurring a dependency on middleboxes.
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1.1. Document Structure
This document describes the core QUIC protocol and is structured as
follows:
* Streams are the basic service abstraction that QUIC provides.
- Section 2 describes core concepts related to streams,
- Section 3 provides a reference model for stream states, and
- Section 4 outlines the operation of flow control.
* Connections are the context in which QUIC endpoints communicate.
- Section 5 describes core concepts related to connections,
- Section 6 describes version negotiation,
- Section 7 details the process for establishing connections,
- Section 8 specifies critical denial of service mitigation
mechanisms,
- Section 9 describes how endpoints migrate a connection to a new
network path,
- Section 10 lists the options for terminating an open
connection, and
- Section 11 provides general guidance for error handling.
* Packets and frames are the basic unit used by QUIC to communicate.
- Section 12 describes concepts related to packets and frames,
- Section 13 defines models for the transmission, retransmission,
and acknowledgement of data, and
- Section 14 specifies rules for managing the size of packets.
* Finally, encoding details of QUIC protocol elements are described
in:
- Section 15 (Versions),
- Section 16 (Integer Encoding),
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- Section 17 (Packet Headers),
- Section 18 (Transport Parameters),
- Section 19 (Frames), and
- Section 20 (Errors).
Accompanying documents describe QUIC's loss detection and congestion
control [QUIC-RECOVERY], and the use of TLS for key negotiation
[QUIC-TLS].
This document defines QUIC version 1, which conforms to the protocol
invariants in [QUIC-INVARIANTS].
1.2. Terms 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.
Commonly used terms in the document are described below.
QUIC: The transport protocol described by this document. QUIC is a
name, not an acronym.
QUIC packet: A complete processable unit of QUIC that can be
encapsulated in a UDP datagram. Multiple QUIC packets can be
encapsulated in a single UDP datagram.
Ack-eliciting Packet: A QUIC packet that contains frames other than
ACK, PADDING, and CONNECTION_CLOSE. These cause a recipient to
send an acknowledgment; see Section 13.2.1.
Out-of-order packet: A packet that does not increase the largest
received packet number for its packet number space (Section 12.3)
by exactly one. A packet can arrive out of order if it is
delayed, if earlier packets are lost or delayed, or if the sender
intentionally skips a packet number.
Endpoint: An entity that can participate in a QUIC connection by
generating, receiving, and processing QUIC packets. There are
only two types of endpoint in QUIC: client and server.
Client: The endpoint initiating a QUIC connection.
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Server: The endpoint accepting incoming QUIC connections.
Address: When used without qualification, the tuple of IP version,
IP address, UDP protocol, and UDP port number that represents one
end of a network path.
Connection ID: An opaque identifier that is used to identify a QUIC
connection at an endpoint. Each endpoint sets a value for its
peer to include in packets sent towards the endpoint.
Stream: A unidirectional or bidirectional channel of ordered bytes
within a QUIC connection. A QUIC connection can carry multiple
simultaneous streams.
Application: An entity that uses QUIC to send and receive data.
1.3. Notational Conventions
Packet and frame diagrams in this document use a bespoke format. The
purpose of this format is to summarize, not define, protocol
elements. Prose defines the complete semantics and details of
structures.
Complex fields are named and then followed by a list of fields
surrounded by a pair of matching braces. Each field in this list is
separated by commas.
Individual fields include length information, plus indications about
fixed value, optionality, or repetitions. Individual fields use the
following notational conventions, with all lengths in bits:
x (A): Indicates that x is A bits long
x (i): Indicates that x uses the variable-length encoding in
Section 16
x (A..B): Indicates that x can be any length from A to B; A can be
omitted to indicate a minimum of zero bits and B can be omitted to
indicate no set upper limit; values in this format always end on
an octet boundary
x (?) = C: Indicates that x has a fixed value of C
x (?) = C..D: Indicates that x has a value in the range from C to D,
inclusive
[x (E)]: Indicates that x is optional (and has length of E)
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x (E) ...: Indicates that x is repeated zero or more times (and that
each instance is length E)
By convention, individual fields reference a complex field by using
the name of the complex field.
For example:
Example Structure {
One-bit Field (1),
7-bit Field with Fixed Value (7) = 61,
Arbitrary-Length Field (..),
Variable-Length Field (8..24),
Field With Minimum Length (16..),
Field With Maximum Length (..128),
[Optional Field (64)],
Repeated Field (8) ...,
}
Figure 1: Example Format
2. Streams
Streams in QUIC provide a lightweight, ordered byte-stream
abstraction to an application. Streams can be unidirectional or
bidirectional. An alternative view of QUIC unidirectional streams is
a "message" abstraction of practically unlimited length.
Streams can be created by sending data. Other processes associated
with stream management - ending, cancelling, and managing flow
control - are all designed to impose minimal overheads. For
instance, a single STREAM frame (Section 19.8) can open, carry data
for, and close a stream. Streams can also be long-lived and can last
the entire duration of a connection.
Streams can be created by either endpoint, can concurrently send data
interleaved with other streams, and can be cancelled. QUIC does not
provide any means of ensuring ordering between bytes on different
streams.
QUIC allows for an arbitrary number of streams to operate
concurrently and for an arbitrary amount of data to be sent on any
stream, subject to flow control constraints and stream limits; see
Section 4.
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2.1. Stream Types and Identifiers
Streams can be unidirectional or bidirectional. Unidirectional
streams carry data in one direction: from the initiator of the stream
to its peer. Bidirectional streams allow for data to be sent in both
directions.
Streams are identified within a connection by a numeric value,
referred to as the stream ID. A stream ID is a 62-bit integer (0 to
2^62-1) that is unique for all streams on a connection. Stream IDs
are encoded as variable-length integers; see Section 16. A QUIC
endpoint MUST NOT reuse a stream ID within a connection.
The least significant bit (0x1) of the stream ID identifies the
initiator of the stream. Client-initiated streams have even-numbered
stream IDs (with the bit set to 0), and server-initiated streams have
odd-numbered stream IDs (with the bit set to 1).
The second least significant bit (0x2) of the stream ID distinguishes
between bidirectional streams (with the bit set to 0) and
unidirectional streams (with the bit set to 1).
The least significant two bits from a stream ID therefore identify a
stream as one of four types, as summarized in Table 1.
+------+----------------------------------+
| Bits | Stream Type |
+======+==================================+
| 0x0 | Client-Initiated, Bidirectional |
+------+----------------------------------+
| 0x1 | Server-Initiated, Bidirectional |
+------+----------------------------------+
| 0x2 | Client-Initiated, Unidirectional |
+------+----------------------------------+
| 0x3 | Server-Initiated, Unidirectional |
+------+----------------------------------+
Table 1: Stream ID Types
Within each type, streams are created with numerically increasing
stream IDs. A stream ID that is used out of order results in all
streams of that type with lower-numbered stream IDs also being
opened.
The first bidirectional stream opened by the client has a stream ID
of 0.
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2.2. Sending and Receiving Data
STREAM frames (Section 19.8) encapsulate data sent by an application.
An endpoint uses the Stream ID and Offset fields in STREAM frames to
place data in order.
Endpoints MUST be able to deliver stream data to an application as an
ordered byte-stream. Delivering an ordered byte-stream requires that
an endpoint buffer any data that is received out of order, up to the
advertised flow control limit.
QUIC makes no specific allowances for delivery of stream data out of
order. However, implementations MAY choose to offer the ability to
deliver data out of order to a receiving application.
An endpoint could receive data for a stream at the same stream offset
multiple times. Data that has already been received can be
discarded. The data at a given offset MUST NOT change if it is sent
multiple times; an endpoint MAY treat receipt of different data at
the same offset within a stream as a connection error of type
PROTOCOL_VIOLATION.
Streams are an ordered byte-stream abstraction with no other
structure visible to QUIC. STREAM frame boundaries are not expected
to be preserved when data is transmitted, retransmitted after packet
loss, or delivered to the application at a receiver.
An endpoint MUST NOT send data on any stream without ensuring that it
is within the flow control limits set by its peer. Flow control is
described in detail in Section 4.
2.3. Stream Prioritization
Stream multiplexing can have a significant effect on application
performance if resources allocated to streams are correctly
prioritized.
QUIC does not provide a mechanism for exchanging prioritization
information. Instead, it relies on receiving priority information
from the application that uses QUIC.
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, the implementation SHOULD use the
information provided by the application.
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2.4. Required Operations on Streams
There are certain operations which an application MUST be able to
perform when interacting with QUIC streams. This document does not
specify an API, but any implementation of this version of QUIC MUST
expose the ability to perform the operations described in this
section on a QUIC stream.
On the sending part of a stream, application protocols need to be
able to:
* write data, understanding when stream flow control credit
(Section 4.1) has successfully been reserved to send the written
data;
* end the stream (clean termination), resulting in a STREAM frame
(Section 19.8) with the FIN bit set; and
* reset the stream (abrupt termination), resulting in a RESET_STREAM
frame (Section 19.4), if the stream was not already in a terminal
state.
On the receiving part of a stream, application protocols need to be
able to:
* read data; and
* abort reading of the stream and request closure, possibly
resulting in a STOP_SENDING frame (Section 19.5).
Applications also need to be informed of state changes on streams,
including when the peer has opened or reset a stream, when a peer
aborts reading on a stream, when new data is available, and when data
can or cannot be written to the stream due to flow control.
3. Stream States
This section describes streams in terms of their send or receive
components. Two state machines are described: one for the streams on
which an endpoint transmits data (Section 3.1), and another for
streams on which an endpoint receives data (Section 3.2).
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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 sides causes the stream to open in
both directions.
An endpoint MUST open streams of the same type in increasing order of
stream ID.
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.
3.1. Sending Stream States
Figure 2 shows the states for the part of a stream that sends data to
a peer.
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o
| Create Stream (Sending)
| Peer Creates Bidirectional Stream
v
+-------+
| Ready | Send RESET_STREAM
| |-----------------------.
+-------+ |
| |
| Send STREAM / |
| STREAM_DATA_BLOCKED |
| |
| Peer Creates |
| Bidirectional Stream |
v |
+-------+ |
| Send | Send RESET_STREAM |
| |---------------------->|
+-------+ |
| |
| Send STREAM + FIN |
v v
+-------+ +-------+
| Data | Send RESET_STREAM | Reset |
| Sent |------------------>| Sent |
+-------+ +-------+
| |
| Recv All ACKs | Recv ACK
v v
+-------+ +-------+
| Data | | Reset |
| Recvd | | Recvd |
+-------+ +-------+
Figure 2: States for Sending Parts of 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. The
"Ready" 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.
Sending the first STREAM or STREAM_DATA_BLOCKED frame causes a
sending part of a stream to enter the "Send" state. An
implementation might choose to defer allocating a stream ID to a
stream until it sends the first STREAM frame and enters this state,
which can allow for better stream prioritization.
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The sending part of a bidirectional stream initiated by a peer (type
0 for a server, type 1 for a client) starts in the "Ready" state when
the receiving part is created.
In the "Send" state, an endpoint transmits - and retransmits as
necessary - stream data in STREAM frames. The endpoint respects the
flow control limits set by its peer, and continues to accept and
process MAX_STREAM_DATA frames. An endpoint in the "Send" state
generates STREAM_DATA_BLOCKED frames if it is blocked from sending by
stream or connection flow control limits Section 4.1.
After the application indicates that all stream data has been sent
and a STREAM frame containing the FIN bit is sent, the sending part
of the stream enters the "Data Sent" state. From this state, the
endpoint only retransmits stream data as necessary. The endpoint
does not need to check flow control limits or send
STREAM_DATA_BLOCKED frames for a stream in this state.
MAX_STREAM_DATA frames might be received until the peer receives the
final stream offset. The endpoint can safely ignore any
MAX_STREAM_DATA frames it receives from its peer for a stream in this
state.
Once all stream data has been successfully acknowledged, the sending
part of the stream enters the "Data Recvd" state, which is a terminal
state.
From any of the "Ready", "Send", or "Data Sent" states, an
application can signal that it wishes to abandon transmission of
stream data. Alternatively, an endpoint might receive a STOP_SENDING
frame from its peer. In either case, the endpoint sends a
RESET_STREAM frame, which causes the stream to enter the "Reset Sent"
state.
An endpoint MAY send a RESET_STREAM as the first frame that mentions
a stream; this causes the sending part of that stream to open and
then immediately transition to the "Reset Sent" state.
Once a packet containing a RESET_STREAM has been acknowledged, the
sending part of the stream enters the "Reset Recvd" state, which is a
terminal state.
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3.2. Receiving Stream States
Figure 3 shows the states for the part of a stream that receives data
from a peer. The states for a receiving part of a stream mirror only
some of the states of the sending part of the stream at the peer.
The receiving part of a stream does not track states on the sending
part that cannot be observed, such as the "Ready" state. Instead,
the receiving part of a stream tracks the delivery of data to the
application, some of which cannot be observed by the sender.
o
| Recv STREAM / STREAM_DATA_BLOCKED / RESET_STREAM
| Create Bidirectional Stream (Sending)
| Recv MAX_STREAM_DATA / STOP_SENDING (Bidirectional)
| Create Higher-Numbered Stream
v
+-------+
| Recv | Recv RESET_STREAM
| |-----------------------.
+-------+ |
| |
| Recv STREAM + FIN |
v |
+-------+ |
| Size | Recv RESET_STREAM |
| Known |---------------------->|
+-------+ |
| |
| Recv All Data |
v v
+-------+ Recv RESET_STREAM +-------+
| Data |--- (optional) --->| Reset |
| Recvd | Recv All Data | Recvd |
+-------+<-- (optional) ----+-------+
| |
| App Read All Data | App Read RST
v v
+-------+ +-------+
| Data | | Reset |
| Read | | Read |
+-------+ +-------+
Figure 3: States for Receiving Parts of 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) is created when the first STREAM,
STREAM_DATA_BLOCKED, or RESET_STREAM is received for that stream.
For bidirectional streams initiated by a peer, receipt of a
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MAX_STREAM_DATA or STOP_SENDING frame for the sending part of the
stream also creates the receiving part. The initial state for the
receiving part of stream is "Recv".
The receiving part of a 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 "Ready" state.
An endpoint opens a bidirectional stream when a MAX_STREAM_DATA or
STOP_SENDING frame is received from the peer for that stream.
Receiving a MAX_STREAM_DATA frame for an unopened stream indicates
that the remote peer has opened the stream and is providing flow
control credit. Receiving a STOP_SENDING frame for an unopened
stream indicates that the remote peer no longer wishes to receive
data on this stream. Either frame might arrive before a STREAM or
STREAM_DATA_BLOCKED frame if packets are lost or reordered.
Before a stream is created, all streams of the same type with lower-
numbered stream IDs MUST be created. This ensures that the creation
order for streams is consistent on both endpoints.
In the "Recv" state, the endpoint receives STREAM and
STREAM_DATA_BLOCKED frames. Incoming data is buffered and can be
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 size of the
stream is known; see Section 4.4. The receiving part of the stream
then 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 receiving part
enters the "Data Recvd" state. This might happen as a result of
receiving the same STREAM frame that causes the transition to "Size
Known". After all data has been received, any STREAM or
STREAM_DATA_BLOCKED frames for the stream can be discarded.
The "Data Recvd" state persists until stream data has been delivered
to the application. Once stream data has been delivered, the stream
enters the "Data Read" state, which is a terminal state.
Receiving a RESET_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.
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It is possible that all stream data is received when a RESET_STREAM
is received (that is, from the "Data Recvd" state). Similarly, it is
possible for remaining stream data to arrive after receiving a
RESET_STREAM frame (the "Reset Recvd" state). An implementation is
free to manage this situation as it chooses.
Sending RESET_STREAM means that an endpoint cannot guarantee delivery
of stream data; however there is no requirement that stream data not
be delivered if a RESET_STREAM is received. An implementation MAY
interrupt delivery of stream data, discard any data that was not
consumed, and signal the receipt of the RESET_STREAM. A RESET_STREAM
signal might be suppressed or withheld if stream data is completely
received and is buffered to be read by the application. If the
RESET_STREAM is suppressed, the receiving part of the stream remains
in "Data Recvd".
Once the application receives the signal indicating that the stream
was reset, the receiving part of the stream transitions to the "Reset
Read" state, which is a terminal state.
3.3. 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 19.8), STREAM_DATA_BLOCKED (Section 19.13), and RESET_STREAM
(Section 19.4).
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_DATA_BLOCKED after sending a RESET_STREAM; that is, in the
terminal states and in the "Reset Sent" state. A receiver could
receive any of these three frames in any state, due to the
possibility of delayed delivery of packets carrying them.
The receiver of a stream sends MAX_STREAM_DATA (Section 19.10) and
STOP_SENDING frames (Section 19.5).
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 RESET_STREAM frame; that is states other than "Reset Recvd" or
"Reset Read". However there is little value in sending a
STOP_SENDING frame in the "Data Recvd" state, since all stream data
has been received. A sender could receive either of these two frames
in any state as a result of delayed delivery of packets.
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3.4. Bidirectional Stream States
A bidirectional stream is composed of sending and receiving parts.
Implementations may represent states of the bidirectional stream as
composites of sending and receiving stream states. The simplest
model presents the stream as "open" when either sending or receiving
parts are in a non-terminal state and "closed" when both sending and
receiving streams are in terminal states.
Table 2 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 sending or receiving parts of 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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+----------------------+----------------------+-----------------+
| Sending Part | Receiving Part | Composite State |
+======================+======================+=================+
| No Stream/Ready | No Stream/Recv *1 | idle |
+----------------------+----------------------+-----------------+
| Ready/Send/Data Sent | Recv/Size Known | open |
+----------------------+----------------------+-----------------+
| Ready/Send/Data Sent | Data Recvd/Data Read | half-closed |
| | | (remote) |
+----------------------+----------------------+-----------------+
| Ready/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 |
| Recvd | | (local) |
+----------------------+----------------------+-----------------+
| Reset Sent/Reset | Data Recvd/Data Read | closed |
| Recvd | | |
+----------------------+----------------------+-----------------+
| Reset Sent/Reset | Reset Recvd/Reset | closed |
| Recvd | Read | |
+----------------------+----------------------+-----------------+
| Data Recvd | Data Recvd/Data Read | closed |
+----------------------+----------------------+-----------------+
| Data Recvd | Reset Recvd/Reset | closed |
| | Read | |
+----------------------+----------------------+-----------------+
Table 2: 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 receiving part of the stream is in the "Recv"
state without yet having received any frames.
3.5. Solicited State Transitions
If an application is no longer interested in the data it is receiving
on a stream, it can abort reading the stream and specify an
application error code.
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If the stream is in the "Recv" or "Size Known" states, the transport
SHOULD signal this by sending a STOP_SENDING frame 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 it is not a guarantee that incoming data will be
ignored.
STREAM frames received after sending STOP_SENDING are still counted
toward connection and stream flow control, even though these frames
can be discarded upon receipt.
A STOP_SENDING frame requests that the receiving endpoint send a
RESET_STREAM frame. An endpoint that receives a STOP_SENDING frame
MUST send a RESET_STREAM frame if the stream is in the Ready or Send
state. If the stream is in the Data Sent state and any outstanding
data is declared lost, an endpoint SHOULD send a RESET_STREAM frame
in lieu of a retransmission.
An endpoint SHOULD copy the error code from the STOP_SENDING frame to
the RESET_STREAM frame it sends, but MAY use any application error
code. The endpoint that sends a STOP_SENDING frame MAY ignore the
error code carried in any RESET_STREAM frame it receives.
If the STOP_SENDING frame is received on a stream that is already in
the "Data Sent" state, an endpoint that wishes to cease
retransmission of previously-sent STREAM frames on that stream MUST
first send a RESET_STREAM frame.
STOP_SENDING SHOULD only be sent for a stream that has not been reset
by the peer. 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 RESET_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.
An endpoint that wishes to terminate both directions of a
bidirectional stream can terminate one direction by sending a
RESET_STREAM, and it can encourage prompt termination in the opposite
direction by sending a STOP_SENDING frame.
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4. Flow Control
It is necessary to limit the amount of data that a receiver could
buffer, to prevent a fast sender from overwhelming a slow receiver,
or to prevent a malicious sender from consuming a large amount of
memory at a receiver. To enable a receiver to limit memory
commitment to a connection and to apply back pressure on the sender,
streams are flow controlled both individually and as an aggregate. A
QUIC receiver controls the maximum amount of data the sender can send
on a stream at any time, as described in Section 4.1 and Section 4.2
Similarly, to limit concurrency within a connection, a QUIC endpoint
controls the maximum cumulative number of streams that its peer can
initiate, as described in Section 4.5.
Data sent in CRYPTO frames is not flow controlled in the same way as
stream data. QUIC relies on the cryptographic protocol
implementation to avoid excessive buffering of data; see [QUIC-TLS].
The implementation SHOULD provide an interface to QUIC to tell it
about its buffering limits so that there is not excessive buffering
at multiple layers.
4.1. Data Flow Control
QUIC employs a credit-based flow-control scheme similar to that in
HTTP/2 [HTTP2], where a receiver advertises the number of bytes it is
prepared to receive on a given stream and for the entire connection.
This leads to two levels of data flow control in QUIC:
* Stream flow control, which prevents a single stream from consuming
the entire receive buffer for a connection by limiting the amount
of data that can be sent on any stream.
* Connection flow control, which prevents senders from exceeding a
receiver's buffer capacity for the connection, by limiting the
total bytes of stream data sent in STREAM frames on all streams.
A receiver sets initial credits for all streams by sending transport
parameters during the handshake (Section 7.4). A receiver sends
MAX_STREAM_DATA (Section 19.10) or MAX_DATA (Section 19.9) frames to
the sender to advertise additional credit.
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A receiver advertises credit for a stream by sending a
MAX_STREAM_DATA frame with the Stream ID field set appropriately. A
MAX_STREAM_DATA frame indicates the maximum absolute byte offset of a
stream. 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.
A receiver advertises credit for a connection by sending a MAX_DATA
frame, which indicates the maximum of the sum of the absolute byte
offsets of all streams. A receiver maintains a cumulative sum of
bytes received on all streams, which is used to check for flow
control violations. A receiver might use a sum of bytes consumed on
all streams to determine the maximum data limit to be advertised.
A receiver can advertise a larger offset by sending MAX_STREAM_DATA
or MAX_DATA frames. Once a receiver advertises an offset, it MAY
advertise a smaller offset, but this has no effect.
A receiver MUST close the connection with a FLOW_CONTROL_ERROR error
(Section 11) if the sender violates the advertised connection or
stream data limits.
A sender MUST ignore any MAX_STREAM_DATA or MAX_DATA frames that do
not increase flow control limits.
If a sender runs out of flow control credit, it will be unable to
send new data and is considered blocked. A sender SHOULD send a
STREAM_DATA_BLOCKED or DATA_BLOCKED frame to indicate it has data to
write but is blocked by flow control limits. If a sender is blocked
for a period longer than the idle timeout (Section 10.2), the
connection might be closed even when data is available for
transmission. To keep the connection from closing, a sender that is
flow control limited SHOULD periodically send a STREAM_DATA_BLOCKED
or DATA_BLOCKED frame when it has no ack-eliciting packets in flight.
4.2. Flow Credit Increments
Implementations decide when and how much credit to advertise in
MAX_STREAM_DATA and MAX_DATA frames, but this section offers a few
considerations.
To avoid blocking a sender, a receiver can send a MAX_STREAM_DATA or
MAX_DATA frame multiple times within a round trip or send it early
enough to allow for recovery from loss of the frame.
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Control frames contribute to connection overhead. Therefore,
frequently sending MAX_STREAM_DATA and MAX_DATA frames with small
changes is undesirable. On the other hand, if updates are less
frequent, larger increments to limits are necessary to avoid blocking
a sender, requiring larger resource commitments at the receiver.
There is a trade-off between resource commitment and overhead when
determining how large a limit is advertised.
A receiver can use an autotuning mechanism to tune the frequency and
amount of advertised additional credit based on a round-trip time
estimate and the rate at which the receiving application consumes
data, similar to common TCP implementations. As an optimization, an
endpoint could send frames related to flow control only when there
are other frames to send or when a peer is blocked, ensuring that
flow control does not cause extra packets to be sent.
A blocked sender is not required to send STREAM_DATA_BLOCKED or
DATA_BLOCKED frames. Therefore, a receiver MUST NOT wait for a
STREAM_DATA_BLOCKED or DATA_BLOCKED frame before sending a
MAX_STREAM_DATA or MAX_DATA frame; doing so could result in the
sender being blocked for the rest of the connection. Even if the
sender sends these frames, waiting for them will result in the sender
being blocked for at least an entire round trip.
When a sender receives credit after being blocked, it might be able
to send a large amount of data in response, resulting in short-term
congestion; see Section 6.9 in [QUIC-RECOVERY] for a discussion of
how a sender can avoid this congestion.
4.3. Handling Stream Cancellation
Endpoints need to eventually agree on the amount of flow control
credit that has been consumed, to avoid either exceeding flow control
limits or deadlocking.
On receipt of a RESET_STREAM frame, an endpoint will tear down state
for the matching stream and ignore further data arriving on that
stream. Without the offset included in RESET_STREAM, the two
endpoints could disagree on the number of bytes that count towards
connection flow control.
To remedy this issue, a RESET_STREAM frame (Section 19.4) includes
the final size of data sent on the stream. On receiving a
RESET_STREAM frame, a receiver definitively knows how many bytes were
sent on that stream before the RESET_STREAM frame, and the receiver
MUST use the final size of the stream to account for all bytes sent
on the stream in its connection level flow controller.
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RESET_STREAM terminates one direction of a stream abruptly. For a
bidirectional stream, RESET_STREAM has no effect on data flow in the
opposite direction. Both endpoints MUST maintain flow control state
for the stream in the unterminated direction until that direction
enters a terminal state, or until one of the endpoints sends
CONNECTION_CLOSE.
4.4. Stream Final Size
The final size is the amount of flow control credit that is consumed
by a stream. Assuming that every contiguous byte on the stream was
sent once, the final size is the number of bytes sent. More
generally, this is one higher than the offset of the byte with the
largest offset sent on the stream, or zero if no bytes were sent.
For a stream that is reset, the final size is carried explicitly in a
RESET_STREAM frame. Otherwise, the final size is the offset plus the
length of a STREAM frame marked with a FIN flag, or 0 in the case of
incoming unidirectional streams.
An endpoint will know the final size for a stream when the receiving
part of the stream enters the "Size Known" or "Reset Recvd" state
(Section 3).
An endpoint MUST NOT send data on a stream at or beyond the final
size.
Once a final size for a stream is known, it cannot change. If a
RESET_STREAM or STREAM frame is received indicating a change in the
final size for the stream, an endpoint SHOULD respond with a
FINAL_SIZE_ERROR error; see Section 11. A receiver SHOULD treat
receipt of data at or beyond the final size as a FINAL_SIZE_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 size
state for closed streams, which could mean a significant state
commitment.
4.5. Controlling Concurrency
An endpoint limits the cumulative number of incoming streams a peer
can open. Only streams with a stream ID less than (max_stream * 4 +
initial_stream_id_for_type) can be opened; see Table 5. Initial
limits are set in the transport parameters (see Section 18.2) and
subsequently limits are advertised using MAX_STREAMS frames
(Section 19.11). Separate limits apply to unidirectional and
bidirectional streams.
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If a max_streams transport parameter or MAX_STREAMS frame is received
with a value greater than 2^60, this would allow a maximum stream ID
that cannot be expressed as a variable-length integer; see
Section 16. If either is received, the connection MUST be closed
immediately with a connection error of type STREAM_LIMIT_ERROR; see
Section 10.3.
Endpoints MUST NOT exceed the limit set by their peer. An endpoint
that receives a frame with a stream ID exceeding the limit it has
sent MUST treat this as a connection error of type STREAM_LIMIT_ERROR
(Section 11).
Once a receiver advertises a stream limit using the MAX_STREAMS
frame, advertising a smaller limit has no effect. A receiver MUST
ignore any MAX_STREAMS frame that does not increase the stream limit.
As with stream and connection flow control, this document leaves when
and how many streams to advertise to a peer via MAX_STREAMS to
implementations. Implementations might choose to increase limits as
streams close to keep the number of streams available to peers
roughly consistent.
An endpoint that is unable to open a new stream due to the peer's
limits SHOULD send a STREAMS_BLOCKED frame (Section 19.14). This
signal is considered useful for debugging. An endpoint MUST NOT wait
to receive this signal before advertising additional credit, since
doing so will mean that the peer will be blocked for at least an
entire round trip, and potentially for longer if the peer chooses to
not send STREAMS_BLOCKED frames.
5. Connections
QUIC's connection establishment combines version negotiation with the
cryptographic and transport handshakes to reduce connection
establishment latency, as described in Section 7. Once established,
a connection may migrate to a different IP or port at either endpoint
as described in Section 9. Finally, a connection may be terminated
by either endpoint, as described in Section 10.
5.1. Connection ID
Each connection possesses a set of connection identifiers, or
connection IDs, each of which can identify the connection.
Connection IDs are independently selected by endpoints; each endpoint
selects the connection IDs that its peer uses.
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The primary function of a connection ID is to ensure that changes in
addressing at lower protocol layers (UDP, IP) don't cause packets for
a QUIC connection to be delivered to the wrong endpoint. Each
endpoint selects connection IDs using an implementation-specific (and
perhaps deployment-specific) method which will allow packets with
that connection ID to be routed back to the endpoint and identified
by the endpoint upon receipt.
Connection IDs MUST NOT contain any information that can be used by
an external observer (that is, one that does not cooperate with the
issuer) to correlate them with other connection IDs for the same
connection. As a trivial example, this means the same connection ID
MUST NOT be issued more than once on the same connection.
Packets with long headers include Source Connection ID and
Destination Connection ID fields. These fields are used to set the
connection IDs for new connections; see Section 7.2 for details.
Packets with short headers (Section 17.3) only include the
Destination Connection ID and omit the explicit length. The length
of the Destination Connection ID field is expected to be known to
endpoints. Endpoints using a load balancer that routes based on
connection ID could agree with the load balancer on a fixed length
for connection IDs, or agree on an encoding scheme. A fixed portion
could encode an explicit length, which allows the entire connection
ID to vary in length and still be used by the load balancer.
A Version Negotiation (Section 17.2.1) packet echoes the connection
IDs selected by the client, both to ensure correct routing toward the
client and to allow the client to validate that the packet is in
response to an Initial packet.
A zero-length connection ID can be used when a connection ID is not
needed to route to the correct endpoint. However, multiplexing
connections on the same local IP address and port while using zero-
length connection IDs will cause failures in the presence of peer
connection migration, NAT rebinding, and client port reuse; and
therefore MUST NOT be done unless an endpoint is certain that those
protocol features are not in use.
When an endpoint uses a non-zero-length connection ID, it needs to
ensure that the peer has a supply of connection IDs from which to
choose for packets sent to the endpoint. These connection IDs are
supplied by the endpoint using the NEW_CONNECTION_ID frame
(Section 19.15).
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5.1.1. Issuing Connection IDs
Each Connection ID has an associated sequence number to assist in
deduplicating messages. The initial connection ID issued by an
endpoint is sent in the Source Connection ID field of the long packet
header (Section 17.2) during the handshake. The sequence number of
the initial connection ID is 0. If the preferred_address transport
parameter is sent, the sequence number of the supplied connection ID
is 1.
Additional connection IDs are communicated to the peer using
NEW_CONNECTION_ID frames (Section 19.15). The sequence number on
each newly-issued connection ID MUST increase by 1. The connection
ID randomly selected by the client in the Initial packet and any
connection ID provided by a Retry packet are not assigned sequence
numbers unless a server opts to retain them as its initial connection
ID.
When an endpoint issues a connection ID, it MUST accept packets that
carry this connection ID for the duration of the connection or until
its peer invalidates the connection ID via a RETIRE_CONNECTION_ID
frame (Section 19.16). Connection IDs that are issued and not
retired are considered active; any active connection ID is valid for
use with the current connection at any time, in any packet type.
This includes the connection ID issued by the server via the
preferred_address transport parameter.
An endpoint SHOULD ensure that its peer has a sufficient number of
available and unused connection IDs. Endpoints advertise the number
of active connection IDs they are willing to maintain using the
active_connection_id_limit transport parameter. An endpoint MUST NOT
provide more connection IDs than the peer's limit. An endpoint MAY
send connection IDs that temporarily exceed a peer's limit if the
NEW_CONNECTION_ID frame also requires the retirement of any excess,
by including a sufficiently large value in the Retire Prior To field.
A NEW_CONNECTION_ID frame might cause an endpoint to add some active
connection IDs and retire others based on the value of the Retire
Prior To field. After processing a NEW_CONNECTION_ID frame and
adding and retiring active connection IDs, if the number of active
connection IDs exceeds the value advertised in its
active_connection_id_limit transport parameter, an endpoint MUST
close the connection with an error of type CONNECTION_ID_LIMIT_ERROR.
An endpoint SHOULD supply a new connection ID when the peer retires a
connection ID. If an endpoint provided fewer connection IDs than the
peer's active_connection_id_limit, it MAY supply a new connection ID
when it receives a packet with a previously unused connection ID. An
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endpoint MAY limit the frequency or the total number of connection
IDs issued for each connection to avoid the risk of running out of
connection IDs; see Section 10.4.2. An endpoint MAY also limit the
issuance of connection IDs to reduce the amount of per-path state it
maintains, such as path validation status, as its peer might interact
with it over as many paths as there are issued connection IDs.
An endpoint that initiates migration and requires non-zero-length
connection IDs SHOULD ensure that the pool of connection IDs
available to its peer allows the peer to use a new connection ID on
migration, as the peer will close the connection if the pool is
exhausted.
5.1.2. Consuming and Retiring Connection IDs
An endpoint can change the connection ID it uses for a peer to
another available one at any time during the connection. An endpoint
consumes connection IDs in response to a migrating peer; see
Section 9.5 for more.
An endpoint maintains a set of connection IDs received from its peer,
any of which it can use when sending packets. When the endpoint
wishes to remove a connection ID from use, it sends a
RETIRE_CONNECTION_ID frame to its peer. Sending a
RETIRE_CONNECTION_ID frame indicates that the connection ID will not
be used again and requests that the peer replace it with a new
connection ID using a NEW_CONNECTION_ID frame.
As discussed in Section 9.5, endpoints limit the use of a connection
ID to packets sent from a single local address to a single
destination address. Endpoints SHOULD retire connection IDs when
they are no longer actively using either the local or destination
address for which the connection ID was used.
An endpoint might need to stop accepting previously issued connection
IDs in certain circumstances. Such an endpoint can cause its peer to
retire connection IDs by sending a NEW_CONNECTION_ID frame with an
increased Retire Prior To field. The endpoint SHOULD continue to
accept the previously issued connection IDs until they are retired by
the peer. If the endpoint can no longer process the indicated
connection IDs, it MAY close the connection.
Upon receipt of an increased Retire Prior To field, the peer MUST
stop using the corresponding connection IDs and retire them with
RETIRE_CONNECTION_ID frames before adding the newly provided
connection ID to the set of active connection IDs. This ordering
allows an endpoint to replace all active connection IDs without the
possibility of a peer having no available connection IDs and without
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exceeding the limit the peer sets in the active_connection_id_limit
transport parameter; see Section 18.2. Failure to cease using the
connection IDs when requested can result in connection failures, as
the issuing endpoint might be unable to continue using the connection
IDs with the active connection.
An endpoint SHOULD limit the number of connection IDs it has retired
locally and have not yet been acknowledged. An endpoint SHOULD allow
for sending and tracking a number of RETIRE_CONNECTION_ID frames of
at least twice the active_connection_id limit. An endpoint MUST NOT
forget a connection ID without retiring it, though it MAY choose to
treat having connection IDs in need of retirement that exceed this
limit as a connection error of type CONNECTION_ID_LIMIT_ERROR.
Endpoints SHOULD NOT issue updates of the Retire Prior To field
before receiving RETIRE_CONNECTION_ID frames that retire all
connection IDs indicated by the previous Retire Prior To value.
5.2. Matching Packets to Connections
Incoming packets are classified on receipt. Packets can either be
associated with an existing connection, or - for servers -
potentially create a new connection.
Endpoints try to associate a packet with an existing connection. If
the packet has a non-zero-length Destination Connection ID
corresponding to an existing connection, QUIC processes that packet
accordingly. Note that more than one connection ID can be associated
with a connection; see Section 5.1.
If the Destination Connection ID is zero length and the addressing
information in the packet matches the addressing information the
endpoint uses to identify a connection with a zero-length connection
ID, QUIC processes the packet as part of that connection. An
endpoint can use just destination IP and port or both source and
destination addresses for identification, though this makes
connections fragile as described in Section 5.1.
Endpoints can send a Stateless Reset (Section 10.4) for any packets
that cannot be attributed to an existing connection. A stateless
reset allows a peer to more quickly identify when a connection
becomes unusable.
Packets that are matched to an existing connection are discarded if
the packets are inconsistent with the state of that connection. For
example, packets are discarded if they indicate a different protocol
version than that of the connection, or if the removal of packet
protection is unsuccessful once the expected keys are available.
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Invalid packets without packet protection, such as Initial, Retry, or
Version Negotiation, MAY be discarded. An endpoint MUST generate a
connection error if it commits changes to state before discovering an
error.
5.2.1. Client Packet Handling
Valid packets sent to clients always include a Destination Connection
ID that matches a value the client selects. Clients that choose to
receive zero-length connection IDs can use the local address and port
to identify a connection. Packets that don't match an existing
connection are discarded.
Due to packet reordering or loss, a client might receive packets for
a connection that are encrypted with a key it has not yet computed.
The client MAY drop these packets, or MAY buffer them in anticipation
of later packets that allow it to compute the key.
If a client receives a packet that has an unsupported version, it
MUST discard that packet.
5.2.2. Server Packet Handling
If a server receives a packet that has an unsupported version, but
the packet is sufficiently large to initiate a new connection for any
version supported by the server, it SHOULD send a Version Negotiation
packet as described in Section 6.1. Servers MAY rate control these
packets to avoid storms of Version Negotiation packets. Otherwise,
servers MUST drop packets that specify unsupported versions.
The first packet for an unsupported version can use different
semantics and encodings for any version-specific field. In
particular, different packet protection keys might be used for
different versions. Servers that do not support a particular version
are unlikely to be able to decrypt the payload of the packet.
Servers SHOULD NOT attempt to decode or decrypt a packet from an
unknown version, but instead send a Version Negotiation packet,
provided that the packet is sufficiently long.
Packets with a supported version, or no version field, are matched to
a connection using the connection ID or - for packets with zero-
length connection IDs - the local address and port. If the packet
doesn't match an existing connection, the server continues below.
If the packet is an Initial packet fully conforming with the
specification, the server proceeds with the handshake (Section 7).
This commits the server to the version that the client selected.
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If a server isn't currently accepting any new connections, it SHOULD
send an Initial packet containing a CONNECTION_CLOSE frame with error
code SERVER_BUSY.
If the packet is a 0-RTT packet, the server MAY buffer a limited
number of these packets in anticipation of a late-arriving Initial
packet. Clients are not able to send Handshake packets prior to
receiving a server response, so servers SHOULD ignore any such
packets.
Servers MUST drop incoming packets under all other circumstances.
5.2.3. Considerations for Simple Load Balancers
A server deployment could load balance among servers using only
source and destination IP addresses and ports. Changes to the
client's IP address or port could result in packets being forwarded
to the wrong server. Such a server deployment could use one of the
following methods for connection continuity when a client's address
changes.
* Servers could use an out-of-band mechanism to forward packets to
the correct server based on Connection ID.
* If servers can use a dedicated server IP address or port, other
than the one that the client initially connects to, they could use
the preferred_address transport parameter to request that clients
move connections to that dedicated address. Note that clients
could choose not to use the preferred address.
A server in a deployment that does not implement a solution to
maintain connection continuity during connection migration SHOULD
disallow migration using the disable_active_migration transport
parameter.
Server deployments that use this simple form of load balancing MUST
avoid the creation of a stateless reset oracle; see Section 21.9.
5.3. Life of a QUIC Connection
A QUIC connection is a stateful interaction between a client and
server, the primary purpose of which is to support the exchange of
data by an application protocol. Streams (Section 2) are the primary
means by which an application protocol exchanges information.
Each connection starts with a handshake phase, during which client
and server establish a shared secret using the cryptographic
handshake protocol [QUIC-TLS] and negotiate the application protocol.
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The handshake (Section 7) confirms that both endpoints are willing to
communicate (Section 8.1) and establishes parameters for the
connection (Section 7.4).
An application protocol can also operate in a limited fashion during
the handshake phase. 0-RTT allows application messages to be sent by
a client before receiving any messages from the server. However,
0-RTT lacks certain key security guarantees. In particular, there is
no protection against replay attacks in 0-RTT; see [QUIC-TLS].
Separately, a server can also send application data to a client
before it receives the final cryptographic handshake messages that
allow it to confirm the identity and liveness of the client. These
capabilities allow an application protocol to offer the option to
trade some security guarantees for reduced latency.
The use of connection IDs (Section 5.1) allows connections to migrate
to a new network path, both as a direct choice of an endpoint and
when forced by a change in a middlebox. Section 9 describes
mitigations for the security and privacy issues associated with
migration.
For connections that are no longer needed or desired, there are
several ways for a client and server to terminate a connection
(Section 10).
5.4. Required Operations on Connections
There are certain operations which an application MUST be able to
perform when interacting with the QUIC transport. This document does
not specify an API, but any implementation of this version of QUIC
MUST expose the ability to perform the operations described in this
section on a QUIC connection.
When implementing the client role, applications need to be able to:
* open a connection, which begins the exchange described in
Section 7;
* enable 0-RTT when available; and
* be informed when 0-RTT has been accepted or rejected by a server.
When implementing the server role, applications need to be able to:
* listen for incoming connections, which prepares for the exchange
described in Section 7;
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* if Early Data is supported, embed application-controlled data in
the TLS resumption ticket sent to the client; and
* if Early Data is supported, retrieve application-controlled data
from the client's resumption ticket and enable rejecting Early
Data based on that information.
In either role, applications need to be able to:
* configure minimum values for the initial number of permitted
streams of each type, as communicated in the transport parameters
(Section 7.4);
* control resource allocation of various types, including flow
control and the number of permitted streams of each type;
* identify whether the handshake has completed successfully or is
still ongoing;
* keep a connection from silently closing, either by generating PING
frames (Section 19.2) or by requesting that the transport send
additional frames before the idle timeout expires (Section 10.2);
and
* immediately close (Section 10.3) the connection.
6. Version Negotiation
Version negotiation ensures that client and server agree to a QUIC
version that is mutually supported. A server sends a Version
Negotiation packet in response to each packet that might initiate a
new connection; see Section 5.2 for details.
The size of the first packet sent by a client will determine whether
a server sends a Version Negotiation packet. Clients that support
multiple QUIC versions SHOULD pad the first packet they send to the
largest of the minimum packet sizes across all versions they support.
This ensures that the server responds if there is a mutually
supported version.
6.1. 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; see
Section 17.2.1. This includes a list of versions that the server
will accept. An endpoint MUST NOT send a Version Negotiation packet
in response to receiving a Version Negotiation packet.
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This system allows a server to process packets with unsupported
versions without retaining state. Though either the 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. As a result, the
client discards all state for the connection and does not send any
more packets on the connection.
A server MAY limit the number of Version Negotiation packets it
sends. For instance, a server that is able to recognize packets as
0-RTT might choose not to send Version Negotiation packets in
response to 0-RTT packets with the expectation that it will
eventually receive an Initial packet.
6.2. Handling Version Negotiation Packets
Version Negotiation packets are designed to allow future versions of
QUIC to negotiate the version in use between endpoints. Future
versions of QUIC might change how implementations that support
multiple versions of QUIC react to Version Negotiation packets when
attempting to establish a connection using this version.
A client that supports only this version of QUIC MUST abandon the
current connection attempt if it receives a Version Negotiation
packet, with the following two exceptions. A client MUST discard any
Version Negotiation packet if it has received and successfully
processed any other packet, including an earlier Version Negotiation
packet. A client MUST discard a Version Negotiation packet that
lists the QUIC version selected by the client.
How to perform version negotiation is left as future work defined by
future versions of QUIC. In particular, that future work will ensure
robustness against version downgrade attacks; see Section 21.10.
6.2.1. Version Negotiation Between Draft Versions
[[RFC editor: please remove this section before publication.]]
When a draft implementation receives a Version Negotiation packet, it
MAY use it to attempt a new connection with one of the versions
listed in the packet, instead of abandoning the current connection
attempt; see Section 6.2.
The client MUST check that the Destination and Source Connection ID
fields match the Source and Destination Connection ID fields in a
packet that the client sent. If this check fails, the packet MUST be
discarded.
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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 new
connection using that version. The new connection MUST use a new
random Destination Connection ID different from the one it had
previously sent.
Note that this mechanism does not protect against downgrade attacks
and MUST NOT be used outside of draft implementations.
6.3. Using Reserved Versions
For a server to use a new version in the future, clients need to
correctly handle unsupported versions. Some version numbers
(0x?a?a?a?a as defined in Section 15) are reserved for inclusion in
fields that contain version numbers.
Endpoints MAY add reserved versions to any field where unknown or
unsupported versions are ignored to test that a peer correctly
ignores the value. For instance, an endpoint could include a
reserved version in a Version Negotiation packet; see Section 17.2.1.
Endpoints MAY send packets with a reserved version to test that a
peer correctly discards the packet.
7. Cryptographic and Transport Handshake
QUIC relies on a combined cryptographic and transport handshake to
minimize connection establishment latency. QUIC uses the CRYPTO
frame Section 19.6 to transmit the cryptographic handshake. Version
0x00000001 of QUIC uses TLS as described in [QUIC-TLS]; a different
QUIC version number could indicate that a different cryptographic
handshake protocol is in use.
QUIC provides reliable, ordered delivery of the cryptographic
handshake data. QUIC packet protection is used to encrypt as much of
the handshake protocol as possible. The cryptographic handshake MUST
provide the following properties:
* 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
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- 1-RTT keys have forward secrecy
* authenticated values for transport parameters of both endpoints,
and confidentiality protection for server transport parameters
(see Section 7.4)
* authenticated negotiation of an application protocol (TLS uses
ALPN [RFC7301] for this purpose)
An endpoint can verify support for Explicit Congestion Notification
(ECN) in the first packets it sends, as described in Section 13.4.2.
The CRYPTO frame can be sent in different packet number spaces
(Section 12.3). The sequence numbers used by CRYPTO frames to ensure
ordered delivery of cryptographic handshake data start from zero in
each packet number space.
Endpoints MUST explicitly negotiate an application protocol. This
avoids situations where there is a disagreement about the protocol
that is in use.
7.1. Example Handshake Flows
Details of how TLS is integrated with QUIC are provided in
[QUIC-TLS], but some examples are provided here. An extension of
this exchange to support client address validation is shown in
Section 8.1.2.
Once any address validation exchanges are complete, the cryptographic
handshake is used to agree on cryptographic keys. The cryptographic
handshake is carried in Initial (Section 17.2.2) and Handshake
(Section 17.2.4) packets.
Figure 4 provides an overview of the 1-RTT handshake. Each line
shows a QUIC packet with the packet type and packet number shown
first, followed by the frames that are typically contained in those
packets. So, for instance the first packet is of type Initial, with
packet number 0, and contains a CRYPTO frame carrying the
ClientHello.
Note that multiple QUIC packets - even of different packet types -
can be coalesced into a single UDP datagram; see Section 12.2). As a
result, this handshake may consist of as few as 4 UDP datagrams, or
any number more. For instance, the server's first flight contains
Initial packets, Handshake packets, and "0.5-RTT data" in 1-RTT
packets with a short header.
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Client Server
Initial[0]: CRYPTO[CH] ->
Initial[0]: CRYPTO[SH] ACK[0]
Handshake[0]: CRYPTO[EE, CERT, CV, FIN]
<- 1-RTT[0]: STREAM[1, "..."]
Initial[1]: ACK[0]
Handshake[0]: CRYPTO[FIN], ACK[0]
1-RTT[0]: STREAM[0, "..."], ACK[0] ->
Handshake[1]: ACK[0]
<- 1-RTT[1]: STREAM[3, "..."], ACK[0]
Figure 4: Example 1-RTT Handshake
Figure 5 shows an example of a connection with a 0-RTT handshake and
a single packet of 0-RTT data. Note that as described in
Section 12.3, the server acknowledges 0-RTT data in 1-RTT packets,
and the client sends 1-RTT packets in the same packet number space.
Client Server
Initial[0]: CRYPTO[CH]
0-RTT[0]: STREAM[0, "..."] ->
Initial[0]: CRYPTO[SH] ACK[0]
Handshake[0] CRYPTO[EE, FIN]
<- 1-RTT[0]: STREAM[1, "..."] ACK[0]
Initial[1]: ACK[0]
Handshake[0]: CRYPTO[FIN], ACK[0]
1-RTT[1]: STREAM[0, "..."] ACK[0] ->
Handshake[1]: ACK[0]
<- 1-RTT[1]: STREAM[3, "..."], ACK[1]
Figure 5: Example 0-RTT Handshake
7.2. Negotiating Connection IDs
A connection ID is used to ensure consistent routing of packets, as
described in Section 5.1. The long header contains two connection
IDs: the Destination Connection ID is chosen by the recipient of the
packet and is used to provide consistent routing; the Source
Connection ID is used to set the Destination Connection ID used by
the peer.
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During the handshake, packets with the long header (Section 17.2) are
used to establish the connection IDs in each direction. Each
endpoint uses the Source Connection ID field to specify the
connection ID that is used in the Destination Connection ID field of
packets being sent to them. Upon receiving a packet, each endpoint
sets the Destination Connection ID it sends to match the value of the
Source Connection ID that it receives.
When an Initial packet is sent by a client that has not previously
received an Initial or Retry packet from the server, the client
populates the Destination Connection ID field with an unpredictable
value. This Destination Connection ID MUST be at least 8 bytes in
length. Until a packet is received from the server, the client MUST
use the same Destination Connection ID value on all packets in this
connection. This Destination Connection ID is used to determine
packet protection keys for Initial packets.
The client populates the Source Connection ID field with a value of
its choosing and sets the SCID Length field to indicate the length.
The first flight of 0-RTT packets use the same Destination Connection
ID and Source Connection ID values as the client's first Initial
packet.
Upon first receiving an Initial or Retry packet from the server, the
client uses the Source Connection ID supplied by the server as the
Destination Connection ID for subsequent packets, including any 0-RTT
packets. This means that a client might have to change the
connection ID it sets in the Destination Connection ID field twice
during connection establishment: once in response to a Retry, and
once in response to an Initial packet from the server. Once a client
has received a valid Initial packet from the server, it MUST discard
any subsequent packet it receives with a different Source Connection
ID.
A client MUST change the Destination Connection ID it uses for
sending packets in response to only the first received Initial or
Retry packet. A server MUST set the Destination Connection ID it
uses for sending packets based on the first received Initial packet.
Any further changes to the Destination Connection ID are only
permitted if the values are taken from any received NEW_CONNECTION_ID
frames; if subsequent Initial packets include a different Source
Connection ID, they MUST be discarded. This avoids unpredictable
outcomes that might otherwise result from stateless processing of
multiple Initial packets with different Source Connection IDs.
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The Destination Connection ID that an endpoint sends can change over
the lifetime of a connection, especially in response to connection
migration (Section 9); see Section 5.1.1 for details.
7.3. Authenticating Connection IDs
The choice each endpoint makes about connection IDs during the
handshake is authenticated by including all values in transport
parameters; see Section 7.4. This ensures that all connection IDs
used for the handshake are also authenticated by the cryptographic
handshake.
Each endpoint includes the value of the Source Connection ID field
from the first Initial packet it sent in the
initial_source_connection_id transport parameter; see Section 18.2.
A server includes the Destination Connection ID field from the first
Initial packet it received from the client in the
original_destination_connection_id transport parameter; if the server
sent a Retry packet this refers to the first Initial packet received
before sending the Retry packet. If it sends a Retry packet, a
server also includes the Source Connection ID field from the Retry
packet in the retry_source_connection_id transport parameter.
The values provided by a peer for these transport parameters MUST
match the values that an endpoint used in the Destination and Source
Connection ID fields of Initial packets that it sent. Including
connection ID values in transport parameters and verifying them
ensures that that an attacker cannot influence the choice of
connection ID for a successful connection by injecting packets
carrying attacker-chosen connection IDs during the handshake. An
endpoint MUST treat any of the following as a connection error of
type PROTOCOL_VIOLATION:
* absence of the initial_source_connection_id transport parameter
from either endpoint,
* absence of the original_destination_connection_id transport
parameter from the server,
* absence of the retry_source_connection_id transport parameter from
the server after receiving a Retry packet,
* presence of the retry_source_connection_id transport parameter
when no Retry packet was received, or
* a mismatch between values received from a peer in these transport
parameters and the value sent in the corresponding Destination or
Source Connection ID fields of Initial packets.
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If a zero-length connection ID is selected, the corresponding
transport parameter is included with a zero-length value.
Figure 6 shows the connection IDs that are used in a complete
handshake. The exchange of Initial packets is shown, plus the later
exchange of 1-RTT packets that includes the connection ID established
during the handshake.
Client Server
Initial: DCID=S1, SCID=C1 ->
<- Initial: DCID=C1, SCID=S3
...
1-RTT: DCID=S3 ->
<- 1-RTT: DCID=C1
Figure 6: Use of Connection IDs in a Handshake
Figure 7 shows a similar handshake that includes a Retry packet.
Client Server
Initial: DCID=S1, SCID=C1 ->
<- Retry: DCID=C1, SCID=S2
Initial: DCID=S2, SCID=C1 ->
<- Initial: DCID=C1, SCID=S3
...
1-RTT: DCID=S3 ->
<- 1-RTT: DCID=C1
Figure 7: Use of Connection IDs in a Handshake with Retry
For the handshakes in Figure 6 and Figure 7 the client sets the value
of the initial_source_connection_id transport parameter to "C1". In
Figure 7, the server sets original_destination_connection_id to "S1",
retry_source_connection_id to "S2", and initial_source_connection_id
to "S3". In Figure 6, the server sets
original_destination_connection_id to "S1",
initial_source_connection_id to "S3", and does not include
retry_source_connection_id. Each endpoint validates the transport
parameters set by their peer, including the client confirming that
retry_source_connection_id is absent if no Retry packet was
processed.
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7.4. Transport Parameters
During connection establishment, both endpoints make authenticated
declarations of their transport parameters. Endpoints are required
to comply with the restrictions implied by these parameters; the
description of each parameter includes rules for its handling.
Transport parameters are declarations that are made unilaterally by
each endpoint. Each endpoint can choose values for transport
parameters independent of the values chosen by its peer.
The encoding of the transport parameters is detailed in Section 18.
QUIC includes the encoded transport parameters 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.
Definitions for each of the defined transport parameters are included
in Section 18.2.
An endpoint MUST treat receipt of a transport parameter with an
invalid value as a connection error of type
TRANSPORT_PARAMETER_ERROR.
An endpoint MUST NOT send a parameter more than once in a given
transport parameters extension. An endpoint SHOULD treat receipt of
duplicate transport parameters as a connection error of type
TRANSPORT_PARAMETER_ERROR.
Endpoints use transport parameters to authenticate the negotiation of
connection IDs during the handshake; see Section 7.3.
7.4.1. Values of Transport Parameters for 0-RTT
Both endpoints store the value of the server transport parameters
from a connection and apply them to any 0-RTT packets that are sent
in subsequent connections to that peer, except for transport
parameters that are explicitly excluded. Remembered transport
parameters apply to the new connection until the handshake completes
and the client starts sending 1-RTT packets. Once the handshake
completes, the client uses the transport parameters established in
the handshake.
The definition of new transport parameters (Section 7.4.2) MUST
specify whether they MUST, MAY, or MUST NOT be stored for 0-RTT. A
client need not store a transport parameter it cannot process.
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A client MUST NOT use remembered values for the following parameters:
ack_delay_exponent, max_ack_delay, initial_source_connection_id,
original_destination_connection_id, preferred_address,
retry_source_connection_id, and stateless_reset_token. The client
MUST use the server's new values in the handshake instead, and absent
new values from the server, the default value.
A client that attempts to send 0-RTT data MUST remember all other
transport parameters used by the server. The server can remember
these transport parameters, 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.
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 the following parameters (Section 18.2) that
are smaller than the remembered value of the parameters.
* active_connection_id_limit
* initial_max_data
* initial_max_stream_data_bidi_local
* initial_max_stream_data_bidi_remote
* initial_max_stream_data_uni
* initial_max_streams_bidi
* initial_max_streams_uni
Omitting or setting a zero value for certain transport parameters can
result in 0-RTT data being enabled, but not usable. The applicable
subset of transport parameters that permit sending of application
data SHOULD be set to non-zero values for 0-RTT. This includes
initial_max_data and either initial_max_streams_bidi and
initial_max_stream_data_bidi_remote, or initial_max_streams_uni and
initial_max_stream_data_uni.
A server MUST either reject 0-RTT data or abort a handshake if the
implied values for transport parameters cannot be supported.
When sending frames in 0-RTT packets, a client MUST only use
remembered transport parameters; importantly, it MUST NOT use updated
values that it learns from the server's updated transport parameters
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or from frames received in 1-RTT packets. Updated values of
transport parameters from the handshake apply only to 1-RTT packets.
For instance, flow control limits from remembered transport
parameters apply to all 0-RTT packets even if those values are
increased by the handshake or by frames sent in 1-RTT packets. A
server MAY treat use of updated transport parameters in 0-RTT as a
connection error of type PROTOCOL_VIOLATION.
7.4.2. 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. As
described in Section 18.1, some identifiers are reserved in order to
exercise this requirement.
New transport parameters can be registered according to the rules in
Section 22.2.
7.5. Cryptographic Message Buffering
Implementations need to maintain a buffer of CRYPTO data received out
of order. Because there is no flow control of CRYPTO frames, an
endpoint could potentially force its peer to buffer an unbounded
amount of data.
Implementations MUST support buffering at least 4096 bytes of data
received in CRYPTO frames out of order. Endpoints MAY choose to
allow more data to be buffered during the handshake. A larger limit
during the handshake could allow for larger keys or credentials to be
exchanged. An endpoint's buffer size does not need to remain
constant during the life of the connection.
Being unable to buffer CRYPTO frames during the handshake can lead to
a connection failure. If an endpoint's buffer is exceeded during the
handshake, it can expand its buffer temporarily to complete the
handshake. If an endpoint does not expand its buffer, it MUST close
the connection with a CRYPTO_BUFFER_EXCEEDED error code.
Once the handshake completes, if an endpoint is unable to buffer all
data in a CRYPTO frame, it MAY discard that CRYPTO frame and all
CRYPTO frames received in the future, or it MAY close the connection
with a CRYPTO_BUFFER_EXCEEDED error code. Packets containing
discarded CRYPTO frames MUST be acknowledged because the packet has
been received and processed by the transport even though the CRYPTO
frame was discarded.
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8. Address Validation
Address validation is used by QUIC to avoid being used for a 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.
The primary defense against amplification attack is verifying that an
endpoint is able to receive packets at the transport address that it
claims. Address validation is performed both during connection
establishment (see Section 8.1) and during connection migration (see
Section 8.2).
8.1. Address Validation During Connection Establishment
Connection establishment implicitly provides address validation for
both endpoints. In particular, receipt of a packet protected with
Handshake keys confirms that the client received the Initial packet
from the server. Once the server has successfully processed a
Handshake packet from the client, it can consider the client address
to have been validated.
Prior to validating the client address, servers MUST NOT send more
than three times as many bytes as the number of bytes they have
received. This limits the magnitude of any amplification attack that
can be mounted using spoofed source addresses. For the purposes of
avoiding amplification prior to address validation, servers MUST
count all of the payload bytes received in datagrams that are
uniquely attributed to a single connection. This includes datagrams
that contain packets that are successfully processed and datagrams
that contain packets that are all discarded.
Clients MUST ensure that UDP datagrams containing Initial packets
have UDP payloads of at least 1200 bytes, adding padding to packets
in the datagram as necessary. Sending padded datagrams ensures that
the server is not overly constrained by the amplification
restriction.
Loss of an Initial or Handshake packet from the server can cause a
deadlock if the client does not send additional Initial or Handshake
packets. A deadlock could occur when the server reaches its anti-
amplification limit and the client has received acknowledgements for
all the data it has sent. In this case, when the client has no
reason to send additional packets, the server will be unable to send
more data because it has not validated the client's address. To
prevent this deadlock, clients MUST send a packet on a probe timeout
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(PTO, see Section 5.3 of [QUIC-RECOVERY]). Specifically, the client
MUST send an Initial packet in a UDP datagram of at least 1200 bytes
if it does not have Handshake keys, and otherwise send a Handshake
packet.
A server might wish to validate the client address before starting
the cryptographic handshake. QUIC uses a token in the Initial packet
to provide address validation prior to completing the handshake.
This token is delivered to the client during connection establishment
with a Retry packet (see Section 8.1.2) or in a previous connection
using the NEW_TOKEN frame (see Section 8.1.3).
In addition to sending limits imposed prior to address validation,
servers are also constrained in what they can send by the limits set
by the congestion controller. Clients are only constrained by the
congestion controller.
8.1.1. Token Construction
A token sent in a NEW_TOKEN frames or a Retry packet MUST be
constructed in a way that allows the server to identify how it was
provided to a client. These tokens are carried in the same field,
but require different handling from servers.
8.1.2. Address Validation using Retry Packets
Upon receiving the client's Initial packet, the server can request
address validation by sending a Retry packet (Section 17.2.5)
containing a token. This token MUST be repeated by the client in all
Initial packets it sends for that connection after it receives the
Retry packet. In response to processing an Initial containing a
token, a server can either abort the connection or permit it to
proceed.
As long as it is not possible for an attacker to generate a valid
token for its own address (see Section 8.1.4) and the client is able
to return that token, it proves to the server that it received the
token.
A server can also use a Retry packet to defer the state and
processing costs of connection establishment. Requiring the server
to provide a different connection ID, along with the
original_destination_connection_id transport parameter defined in
Section 18.2, forces the server to demonstrate that it, or an entity
it cooperates with, received the original Initial packet from the
client. Providing a different connection ID also grants a server
some control over how subsequent packets are routed. This can be
used to direct connections to a different server instance.
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If a server receives a client Initial that can be unprotected but
contains an invalid Retry token, it knows the client will not accept
another Retry token. The server can discard such a packet and allow
the client to time out to detect handshake failure, but that could
impose a significant latency penalty on the client. Instead, the
server SHOULD immediately close (Section 10.3) the connection with an
INVALID_TOKEN error. Note that a server has not established any
state for the connection at this point and so does not enter the
closing period.
A flow showing the use of a Retry packet is shown in Figure 8.
Client Server
Initial[0]: CRYPTO[CH] ->
<- Retry+Token
Initial+Token[1]: CRYPTO[CH] ->
Initial[0]: CRYPTO[SH] ACK[1]
Handshake[0]: CRYPTO[EE, CERT, CV, FIN]
<- 1-RTT[0]: STREAM[1, "..."]
Figure 8: Example Handshake with Retry
8.1.3. Address Validation for Future Connections
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.
The server uses the NEW_TOKEN frame Section 19.7 to provide the
client with an address validation token that can be used to validate
future connections. The client includes this token in Initial
packets to provide address validation in a future connection. The
client MUST include the token in all Initial packets it sends, unless
a Retry replaces the token with a newer one. The client MUST NOT use
the token provided in a Retry for future connections. Servers MAY
discard any Initial packet that does not carry the expected token.
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Unlike the token that is created for a Retry packet, which is used
immediately, the token sent in the NEW_TOKEN frame might be used
after some period of time has passed. Thus, a token SHOULD have an
expiration time, which could be either an explicit expiration time or
an issued timestamp that can be used to dynamically calculate the
expiration time. A server can store the expiration time or include
it in an encrypted form in the token.
A token issued with NEW_TOKEN MUST NOT include information that would
allow values to be linked by an observer to the connection on which
it was issued, unless the values are encrypted. For example, it
cannot include the previous connection ID or addressing information.
A server MUST ensure that every NEW_TOKEN frame it sends is unique
across all clients, with the exception of those sent to repair losses
of previously sent NEW_TOKEN frames. Information that allows the
server to distinguish between tokens from Retry and NEW_TOKEN MAY be
accessible to entities other than the server.
It is unlikely that the client port number is the same on two
different connections; validating the port is therefore unlikely to
be successful.
A token received in a NEW_TOKEN frame is applicable to any server
that the connection is considered authoritative for (e.g., server
names included in the certificate). When connecting to a server for
which the client retains an applicable and unused token, it SHOULD
include that token in the Token field of its Initial packet.
Including a token might allow the server to validate the client
address without an additional round trip. A client MUST NOT include
a token that is not applicable to the server that it is connecting
to, unless the client has the knowledge that the server that issued
the token and the server the client is connecting to are jointly
managing the tokens. A client MAY use a token from any previous
connection to that server.
A token allows a server to correlate activity between the connection
where the token was issued and any connection where it is used.
Clients that want to break continuity of identity with a server MAY
discard tokens provided using the NEW_TOKEN frame. In comparison, a
token obtained in a Retry packet MUST be used immediately during the
connection attempt and cannot be used in subsequent connection
attempts.
A client SHOULD NOT reuse a NEW_TOKEN token for different connection
attempts. Reusing a token allows connections to be linked by
entities on the network path; see Section 9.5.
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Clients might receive multiple tokens on a single connection. Aside
from preventing linkability, any token can be used in any connection
attempt. Servers can send additional tokens to either enable address
validation for multiple connection attempts or to replace older
tokens that might become invalid. For a client, this ambiguity means
that sending the most recent unused token is most likely to be
effective. Though saving and using older tokens has no negative
consequences, clients can regard older tokens as being less likely be
useful to the server for address validation.
When a server receives an Initial packet with an address validation
token, it MUST attempt to validate the token, unless it has already
completed address validation. If the token is invalid then the
server SHOULD proceed as if the client did not have a validated
address, including potentially sending a Retry. If the validation
succeeds, the server SHOULD then allow the handshake to proceed.
Note: The rationale for treating the client as unvalidated rather
than discarding the packet is that the client might have received
the token in a previous connection using the NEW_TOKEN frame, and
if the server has lost state, it might be unable to validate the
token at all, leading to connection failure if the packet is
discarded. A server SHOULD encode tokens provided with NEW_TOKEN
frames and Retry packets differently, and validate the latter more
strictly.
In a stateless design, a server can use encrypted and authenticated
tokens to pass information to clients that the server can later
recover and use to validate a client address. Tokens are not
integrated into the cryptographic handshake and so they are not
authenticated. For instance, a client might be able to reuse a
token. To avoid attacks that exploit this property, a server can
limit its use of tokens to only the information needed to validate
client addresses.
Clients MAY use tokens obtained on one connection for any connection
attempt using the same version. When selecting a token to use,
clients do not need to consider other properties of the connection
that is being attempted, including the choice of possible application
protocols, session tickets, or other connection properties.
Attackers could replay tokens to use servers as amplifiers in DDoS
attacks. To protect against such attacks, servers SHOULD ensure that
tokens sent in Retry packets are only accepted for a short time.
Tokens that are provided in NEW_TOKEN frames (Section 19.7) need to
be valid for longer, but SHOULD NOT be accepted multiple times in a
short period. Servers are encouraged to allow tokens to be used only
once, if possible.
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8.1.4. 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.
There is no need for a single well-defined format for the token
because the server that generates the token also consumes it. Tokens
sent in Retry packets SHOULD include information that allows the
server to verify that the source IP address and port in client
packets remains constant.
Tokens sent in NEW_TOKEN frames MUST include information that allows
the server to verify that the client IP address has not changed from
when the token was issued. Servers can use tokens from NEW_TOKEN in
deciding not to send a Retry packet, even if the client address has
changed. If the client IP address has changed, the server MUST
adhere to the anti-amplification limits found in Section 8.1. Note
that in the presence of NAT, this requirement might be insufficient
to protect other hosts that share the NAT from amplification attack.
Servers MUST ensure that replay of tokens is prevented or limited.
For instance, servers might limit the time over which a token is
accepted. Tokens provided in NEW_TOKEN frames might need to allow
longer validity periods. Tokens MAY include additional information
about clients to further narrow applicability or reuse.
8.2. Path Validation
Path validation is used during connection migration (see Section 9
and Section 9.6) by the migrating endpoint to verify reachability of
a peer from a new local address. In path validation, endpoints test
reachability between a specific local address and a specific peer
address, where an address is the two-tuple of IP address and port.
Path validation tests that packets (PATH_CHALLENGE) can be both sent
to and received (PATH_RESPONSE) from a peer on the path.
Importantly, it validates that the packets received from the
migrating endpoint do not carry a spoofed source address.
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Path validation can be used at any time by either endpoint. For
instance, an endpoint might check that a peer is still in possession
of its address after a period of quiescence.
Path validation is not designed as a NAT traversal mechanism. Though
the mechanism described here might be effective for the creation of
NAT bindings that support NAT traversal, the expectation is that one
or other peer is able to receive packets without first having sent a
packet on that path. Effective NAT traversal needs additional
synchronization mechanisms that are not provided here.
An endpoint MAY bundle PATH_CHALLENGE and PATH_RESPONSE frames that
are used for path validation with other frames. In particular, an
endpoint may pad a packet carrying a PATH_CHALLENGE for PMTU
discovery, or an endpoint may bundle a PATH_RESPONSE with its own
PATH_CHALLENGE.
When probing a new path, an endpoint might want to ensure that its
peer has an unused connection ID available for responses. The
endpoint can send NEW_CONNECTION_ID and PATH_CHALLENGE frames in the
same packet. This ensures that an unused connection ID will be
available to the peer when sending a response.
8.3. Initiating Path Validation
To initiate path validation, an endpoint sends a PATH_CHALLENGE frame
containing a random payload on the path to be validated.
An endpoint MAY send multiple PATH_CHALLENGE frames to guard against
packet loss. However, an endpoint SHOULD NOT send multiple
PATH_CHALLENGE frames in a single packet. An endpoint SHOULD NOT
send a PATH_CHALLENGE more frequently than it would an Initial
packet, ensuring that connection migration is no more load on a new
path than establishing a new connection.
The endpoint MUST use unpredictable data in every PATH_CHALLENGE
frame so that it can associate the peer's response with the
corresponding PATH_CHALLENGE.
8.4. Path Validation Responses
On receiving a PATH_CHALLENGE frame, an endpoint MUST respond
immediately by echoing the data contained in the PATH_CHALLENGE frame
in a PATH_RESPONSE frame.
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An endpoint MUST NOT send more than one PATH_RESPONSE frame in
response to one PATH_CHALLENGE frame; see Section 13.3. The peer is
expected to send more PATH_CHALLENGE frames as necessary to evoke
additional PATH_RESPONSE frames.
8.5. Successful Path Validation
A new address is considered valid when a PATH_RESPONSE frame is
received that contains the data that was sent in a previous
PATH_CHALLENGE. Receipt of an acknowledgment for a packet containing
a PATH_CHALLENGE frame is not adequate validation, since the
acknowledgment can be spoofed by a malicious peer.
Note that receipt on a different local address does not result in
path validation failure, as it might be a result of a forwarded
packet (see Section 9.3.3) or misrouting. It is possible that a
valid PATH_RESPONSE might be received in the future.
8.6. Failed Path Validation
Path validation only fails when the endpoint attempting to validate
the path abandons its attempt to validate the path.
Endpoints SHOULD abandon path validation based on a timer. When
setting this timer, implementations are cautioned that the new path
could have a longer round-trip time than the original. A value of
three times the larger of the current Probe Timeout (PTO) or the
initial timeout (that is, 2*kInitialRtt) as defined in
[QUIC-RECOVERY] is RECOMMENDED. That is:
validation_timeout = max(3*PTO, 6*kInitialRtt)
Note that the endpoint might receive packets containing other frames
on the new path, but a PATH_RESPONSE frame with appropriate data is
required for path validation to succeed.
When an endpoint abandons path validation, it determines that the
path is unusable. This does not necessarily imply a failure of the
connection - endpoints can continue sending packets over other paths
as appropriate. If no paths are available, an endpoint can wait for
a new path to become available or close the connection.
A path validation might be abandoned for other reasons besides
failure. Primarily, this happens if a connection migration to a new
path is initiated while a path validation on the old path is in
progress.
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9. Connection Migration
The use of a connection ID allows connections to survive changes to
endpoint addresses (IP address and port), such as those caused by an
endpoint migrating to a new network. This section describes the
process by which an endpoint migrates to a new address.
The design of QUIC relies on endpoints retaining a stable address for
the duration of the handshake. An endpoint MUST NOT initiate
connection migration before the handshake is confirmed, as defined in
section 4.1.2 of [QUIC-TLS].
An endpoint also MUST NOT send packets from a different local
address, actively initiating migration, if the peer sent the
disable_active_migration transport parameter during the handshake.
An endpoint which has sent this transport parameter, but detects that
a peer has nonetheless migrated to a different network MUST either
drop the incoming packets on that path without generating a stateless
reset or proceed with path validation and allow the peer to migrate.
Generating a stateless reset or closing the connection would allow
third parties in the network to cause connections to close by
spoofing or otherwise manipulating observed traffic.
Not all changes of peer address are intentional, or active,
migrations. The peer could experience NAT rebinding: a change of
address due to a middlebox, usually a NAT, allocating a new outgoing
port or even a new outgoing IP address for a flow. An endpoint MUST
perform path validation (Section 8.2) if it detects any change to a
peer's address, unless it has previously validated that address.
When an endpoint has no validated path on which to send packets, it
MAY discard connection state. An endpoint capable of connection
migration MAY wait for a new path to become available before
discarding connection state.
This document limits migration of connections to new client
addresses, except as described in Section 9.6. Clients are
responsible for initiating all migrations. Servers do not send non-
probing packets (see Section 9.1) toward a client address until they
see a non-probing packet from that address. If a client receives
packets from an unknown server address, the client MUST discard these
packets.
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9.1. Probing a New Path
An endpoint MAY probe for peer reachability from a new local address
using path validation Section 8.2 prior to migrating the connection
to the new local address. Failure of path validation simply means
that the new path is not usable for this connection. Failure to
validate a path does not cause the connection to end unless there are
no valid alternative paths available.
An endpoint uses a new connection ID for probes sent from a new local
address; see Section 9.5 for further discussion. An endpoint that
uses a new local address needs to ensure that at least one new
connection ID is available at the peer. That can be achieved by
including a NEW_CONNECTION_ID frame in the probe.
Receiving a PATH_CHALLENGE frame from a peer indicates that the peer
is probing for reachability on a path. An endpoint sends a
PATH_RESPONSE in response as per Section 8.2.
PATH_CHALLENGE, PATH_RESPONSE, NEW_CONNECTION_ID, and PADDING frames
are "probing frames", and all other frames are "non-probing frames".
A packet containing only probing frames is a "probing packet", and a
packet containing any other frame is a "non-probing packet".
9.2. Initiating Connection Migration
An endpoint can migrate a connection to a new local address by
sending packets containing non-probing frames from that address.
Each endpoint validates its peer's address during connection
establishment. Therefore, a migrating endpoint can send to its peer
knowing that the peer is willing to receive at the peer's current
address. Thus an endpoint can migrate to a new local address without
first validating the peer's address.
When migrating, the new path might not support the endpoint's current
sending rate. Therefore, the endpoint resets its congestion
controller, as described in Section 9.4.
The new path might not have the same ECN capability. Therefore, the
endpoint verifies ECN capability as described in Section 13.4.
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Receiving acknowledgments for data sent on the new path serves as
proof of the peer's reachability from the new address. Note that
since acknowledgments may be received on any path, return
reachability on the new path is not established. To establish return
reachability on the new path, an endpoint MAY concurrently initiate
path validation Section 8.2 on the new path or it MAY choose to wait
for the peer to send the next non-probing frame to its new address.
9.3. Responding to Connection Migration
Receiving a packet from a new peer address containing a non-probing
frame indicates that the peer has migrated to that address.
In response to such a packet, an endpoint MUST start sending
subsequent packets to the new peer address and MUST initiate path
validation (Section 8.2) to verify the peer's ownership of the
unvalidated address.
An endpoint MAY send data to an unvalidated peer address, but it MUST
protect against potential attacks as described in Section 9.3.1 and
Section 9.3.2. An endpoint MAY skip validation of a peer address if
that address has been seen recently. In particular, if an endpoint
returns to a previously-validated path after detecting some form of
spurious migration, skipping address validation and restoring loss
detection and congestion state can reduce the performance impact of
the attack.
An endpoint only changes the address that it sends packets to in
response to the highest-numbered non-probing packet. This ensures
that an endpoint does not send packets to an old peer address in the
case that it receives reordered packets.
After changing the address to which it sends non-probing packets, an
endpoint could abandon any path validation for other addresses.
Receiving a packet from a new peer address might be the result of a
NAT rebinding at the peer.
After verifying a new client address, the server SHOULD send new
address validation tokens (Section 8) to the client.
9.3.1. Peer Address Spoofing
It is possible that a peer is spoofing its source address to cause an
endpoint to send excessive amounts of data to an unwilling host. If
the endpoint sends significantly more data than the spoofing peer,
connection migration might be used to amplify the volume of data that
an attacker can generate toward a victim.
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As described in Section 9.3, an endpoint is required to validate a
peer's new address to confirm the peer's possession of the new
address. Until a peer's address is deemed valid, an endpoint MUST
limit the rate at which it sends data to this address. The endpoint
MUST NOT send more than a minimum congestion window's worth of data
per estimated round-trip time (kMinimumWindow, as defined in
[QUIC-RECOVERY]). In the absence of this limit, an endpoint risks
being used for a denial of service attack against an unsuspecting
victim. Note that since the endpoint will not have any round-trip
time measurements to this address, the estimate SHOULD be the default
initial value; see [QUIC-RECOVERY].
If an endpoint skips validation of a peer address as described in
Section 9.3, it does not need to limit its sending rate.
9.3.2. On-Path Address Spoofing
An on-path attacker could cause a spurious connection migration by
copying and forwarding a packet with a spoofed address such that it
arrives before the original packet. The packet with the spoofed
address will be seen to come from a migrating connection, and the
original packet will be seen as a duplicate and dropped. 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 PATH_CHALLENGE frame
that is sent to it even if it wanted to.
To protect the connection from failing due to such a spurious
migration, an endpoint MUST revert to using the last validated peer
address when validation of a new peer address fails.
If an endpoint has no state about the last validated peer address, it
MUST close the connection silently by discarding 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.
Note that receipt of packets with higher packet numbers from the
legitimate peer address will trigger another connection migration.
This will cause the validation of the address of the spurious
migration to be abandoned.
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9.3.3. Off-Path Packet Forwarding
An off-path attacker that can observe packets might forward copies of
genuine packets to endpoints. If the copied packet arrives before
the genuine packet, this will appear as a NAT rebinding. Any genuine
packet will be discarded as a duplicate. If the attacker is able to
continue forwarding packets, it might be able to cause migration to a
path via the attacker. This places the attacker on path, giving it
the ability to observe or drop all subsequent packets.
Unlike the attack described in Section 9.3.2, the attacker can ensure
that the new path is successfully validated.
This style of attack relies on the attacker using a path that is
approximately as fast as the direct path between endpoints. The
attack is more reliable if relatively few packets are sent or if
packet loss coincides with the attempted attack.
A non-probing packet received on the original path that increases the
maximum received packet number will cause the endpoint to move back
to that path. Eliciting packets on this path increases the
likelihood that the attack is unsuccessful. Therefore, mitigation of
this attack relies on triggering the exchange of packets.
In response to an apparent migration, endpoints MUST validate the
previously active path using a PATH_CHALLENGE frame. This induces
the sending of new packets on that path. If the path is no longer
viable, the validation attempt will time out and fail; if the path is
viable, but no longer desired, the validation will succeed, but only
results in probing packets being sent on the path.
An endpoint that receives a PATH_CHALLENGE on an active path SHOULD
send a non-probing packet in response. If the non-probing packet
arrives before any copy made by an attacker, this results in the
connection being migrated back to the original path. Any subsequent
migration to another path restarts this entire process.
This defense is imperfect, but this is not considered a serious
problem. If the path via the attack is reliably faster than the
original path despite multiple attempts to use that original path, it
is not possible to distinguish between attack and an improvement in
routing.
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An endpoint could also use heuristics to improve detection of this
style of attack. For instance, NAT rebinding is improbable if
packets were recently received on the old path, similarly rebinding
is rare on IPv6 paths. Endpoints can also look for duplicated
packets. Conversely, a change in connection ID is more likely to
indicate an intentional migration rather than an attack.
9.4. Loss Detection and Congestion Control
The capacity available on the new path might not be the same as the
old path. Packets sent on the old path MUST NOT contribute to
congestion control or RTT estimation for the new path.
On confirming a peer's ownership of its new address, an endpoint MUST
immediately reset the congestion controller and round-trip time
estimator for the new path to initial values (see Sections A.3 and
B.3 in [QUIC-RECOVERY]) unless it has knowledge that a previous send
rate or round-trip time estimate is valid for the new path. For
instance, an endpoint might infer that a change in only the client's
port number is indicative of a NAT rebinding, meaning that the new
path is likely to have similar bandwidth and round-trip time.
However, this determination will be imperfect. If the determination
is incorrect, the congestion controller and the RTT estimator are
expected to adapt to the new path. Generally, implementations are
advised to be cautious when using previous values on a new path.
There may be apparent reordering at the receiver when an endpoint
sends data and probes from/to multiple addresses during the migration
period, since the two resulting paths may have different round-trip
times. A receiver of packets on multiple paths will still send ACK
frames covering all received packets.
While multiple paths might be used during connection migration, a
single congestion control context and a single loss recovery context
(as described in [QUIC-RECOVERY]) may be adequate. For instance, an
endpoint might delay switching to a new congestion control context
until it is confirmed that an old path is no longer needed (such as
the case in Section 9.3.3).
A sender can make exceptions for probe packets so that their loss
detection is independent and does not unduly cause the congestion
controller to reduce its sending rate. An endpoint might set a
separate timer when a PATH_CHALLENGE is sent, which is cancelled if
the corresponding PATH_RESPONSE is received. If the timer fires
before the PATH_RESPONSE is received, the endpoint might send a new
PATH_CHALLENGE, and restart the timer for a longer period of time.
This timer SHOULD be set as described in Section 5.3 of
[QUIC-RECOVERY] and MUST NOT be more aggressive.
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9.5. Privacy Implications of Connection Migration
Using a stable connection ID on multiple network paths allows a
passive observer to correlate activity between those paths. An
endpoint that moves between networks might not wish to have their
activity correlated by any entity other than their peer, so different
connection IDs are used when sending from different local addresses,
as discussed in Section 5.1. For this to be effective endpoints need
to ensure that connection IDs they provide cannot be linked by any
other entity.
At any time, endpoints MAY change the Destination Connection ID they
send to a value that has not been used on another path.
An endpoint MUST NOT reuse a connection ID when sending from more
than one local address, for example when initiating connection
migration as described in Section 9.2 or when probing a new network
path as described in Section 9.1.
Similarly, an endpoint MUST NOT reuse a connection ID when sending to
more than one destination address. Due to network changes outside
the control of its peer, an endpoint might receive packets from a new
source address with the same destination connection ID, in which case
it MAY continue to use the current connection ID with the new remote
address while still sending from the same local address.
These requirements regarding connection ID reuse apply only to the
sending of packets, as unintentional changes in path without a change
in connection ID are possible. For example, after a period of
network inactivity, NAT rebinding might cause packets to be sent on a
new path when the client resumes sending. An endpoint responds to
such an event as described in Section 9.3.
Using different connection IDs for packets sent in both directions on
each new network path eliminates the use of the connection ID for
linking packets from the same connection across different network
paths. Header protection ensures that packet numbers cannot be used
to correlate activity. This does not prevent other properties of
packets, such as timing and size, from being used to correlate
activity.
An endpoint SHOULD NOT initiate migration with a peer that has
requested a zero-length connection ID, because traffic over the new
path might be trivially linkable to traffic over the old one. If the
server is able to route packets with a zero-length connection ID to
the right connection, it means that the server is using other
information to demultiplex packets. For example, a server might
provide a unique address to every client, for instance using HTTP
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alternative services [ALTSVC]. Information that might allow correct
routing of packets across multiple network paths will also allow
activity on those paths to be linked by entities other than the peer.
A client might wish to reduce linkability by employing a new
connection ID and source UDP port when sending traffic after a period
of inactivity. Changing the UDP port from which it sends packets at
the same time might cause the packet to appear as a connection
migration. This ensures that the mechanisms that support migration
are exercised even for clients that don't experience NAT rebindings
or genuine migrations. Changing port number can cause a peer to
reset its congestion state (see Section 9.4), so the port SHOULD only
be changed infrequently.
An endpoint that exhausts available connection IDs cannot probe new
paths or initiate migration, nor can it respond to probes or attempts
by its peer to migrate. To ensure that migration is possible and
packets sent on different paths cannot be correlated, endpoints
SHOULD provide new connection IDs before peers migrate; see
Section 5.1.1. If a peer might have exhausted available connection
IDs, a migrating endpoint could include a NEW_CONNECTION_ID frame in
all packets sent on a new network path.
9.6. Server's Preferred Address
QUIC allows servers to accept connections on one IP address and
attempt to transfer these connections to a more preferred address
shortly after the handshake. This is particularly useful when
clients initially connect to an address shared by multiple servers
but would prefer to use a unicast address to ensure connection
stability. This section describes the protocol for migrating a
connection to a preferred server address.
Migrating a connection to a new server address mid-connection is left
for future work. If a client receives packets from a new server
address not indicated by the preferred_address transport parameter,
the client SHOULD discard these packets.
9.6.1. Communicating a Preferred Address
A server conveys a preferred address by including the
preferred_address transport parameter in the TLS handshake.
Servers MAY communicate a preferred address of each address family
(IPv4 and IPv6) to allow clients to pick the one most suited to their
network attachment.
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Once the handshake is confirmed, the client SHOULD select one of the
two server's preferred addresses and initiate path validation (see
Section 8.2) of that address using any previously unused active
connection ID, taken from either the preferred_address transport
parameter or a NEW_CONNECTION_ID frame.
If path validation succeeds, the client SHOULD immediately begin
sending all future packets to the new server address using the new
connection ID and discontinue use of the old server address. If path
validation fails, the client MUST continue sending all future packets
to the server's original IP address.
9.6.2. Responding to Connection Migration
A server might receive a packet addressed to its preferred IP address
at any time after it accepts a connection. If this packet contains a
PATH_CHALLENGE frame, the server sends a PATH_RESPONSE frame as per
Section 8.2. The server MUST send other non-probing frames from its
original address until it receives a non-probing packet from the
client at its preferred address and until the server has validated
the new path.
The server MUST probe on the path toward the client from its
preferred address. This helps to guard against spurious migration
initiated by an attacker.
Once the server has completed its path validation and has received a
non-probing packet with a new largest packet number on its preferred
address, the server begins sending non-probing packets to the client
exclusively from its preferred IP address. It SHOULD drop packets
for this connection received on the old IP address, but MAY continue
to process delayed packets.
The addresses that a server provides in the preferred_address
transport parameter are only valid for the connection in which they
are provided. A client MUST NOT use these for other connections,
including connections that are resumed from the current connection.
9.6.3. Interaction of Client Migration and Preferred Address
A client might need to perform a connection migration before it has
migrated to the server's preferred address. In this case, the client
SHOULD perform path validation to both the original and preferred
server address from the client's new address concurrently.
If path validation of the server's preferred address succeeds, the
client MUST abandon validation of the original address and migrate to
using the server's preferred address. If path validation of the
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server's preferred address fails but validation of the server's
original address succeeds, the client MAY migrate to its new address
and continue sending to the server's original address.
If the connection to the server's preferred address is not from the
same client address, the server MUST protect against potential
attacks as described in Section 9.3.1 and Section 9.3.2. In addition
to intentional simultaneous migration, this might also occur because
the client's access network used a different NAT binding for the
server's preferred address.
Servers SHOULD initiate path validation to the client's new address
upon receiving a probe packet from a different address. Servers MUST
NOT send more than a minimum congestion window's worth of non-probing
packets to the new address before path validation is complete.
A client that migrates to a new address SHOULD use a preferred
address from the same address family for the server.
The connection ID provided in the preferred_address transport
parameter is not specific to the addresses that are provided. This
connection ID is provided to ensure that the client has a connection
ID available for migration, but the client MAY use this connection ID
on any path.
9.7. Use of IPv6 Flow-Label and Migration
Endpoints that send data using IPv6 SHOULD apply an IPv6 flow label
in compliance with [RFC6437], unless the local API does not allow
setting IPv6 flow labels.
The IPv6 flow label SHOULD be a pseudo-random function of the source
and destination addresses, source and destination UDP ports, and the
destination CID. The flow label generation MUST be designed to
minimize the chances of linkability with a previously used flow
label, as this would enable correlating activity on multiple paths;
see Section 9.5.
A possible implementation is to compute the flow label as a
cryptographic hash function of the source and destination addresses,
source and destination UDP ports, destination CID, and a local
secret.
10. Connection Termination
An established QUIC connection can be terminated in one of three
ways:
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* idle timeout (Section 10.2)
* immediate close (Section 10.3)
* stateless reset (Section 10.4)
An endpoint MAY discard connection state if it does not have a
validated path on which it can send packets; see Section 8.2.
10.1. 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 at least three
times the current Probe Timeout (PTO) interval as defined in
[QUIC-RECOVERY].
An endpoint enters a closing period after initiating an immediate
close; Section 10.3. While closing, an endpoint MUST NOT send
packets unless they contain a CONNECTION_CLOSE frame; see
Section 10.3 for details. An endpoint retains only enough
information to generate a packet containing a CONNECTION_CLOSE frame
and to identify packets as belonging to the connection. The
endpoint's selected connection ID and the QUIC version are 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 CONNECTION_CLOSE 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 receives a CONNECTION_CLOSE frame or stateless reset,
both of which indicate that the peer is also closing or draining.
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 generate
an unnecessary stateless reset. Endpoints that have some alternative
means to ensure that late-arriving packets on the connection do not
induce a response, such as those that are able to close the UDP
socket, MAY use an abbreviated draining period which can allow for
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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 10.4) is sent.
An endpoint is not expected to handle key updates when it is closing
or draining. A key update might prevent the endpoint from moving
from the closing state to draining, but it otherwise has no impact.
While in the closing period, an endpoint could receive packets from a
new source address, indicating a connection migration; Section 9. An
endpoint in the closing state MUST strictly limit the number of
packets it sends to this new address until the address is validated;
see Section 8.2. A server in the closing state MAY instead choose to
discard packets received from a new source address.
10.2. Idle Timeout
If a max_idle_timeout is specified by either peer in its transport
parameters (Section 18.2), the connection is silently closed and its
state is discarded when it remains idle for longer than the minimum
of both peers max_idle_timeout values and three times the current
Probe Timeout (PTO).
Each endpoint advertises a max_idle_timeout, but the effective value
at an endpoint is computed as the minimum of the two advertised
values. By announcing a max_idle_timeout, an endpoint commits to
initiating an immediate close (Section 10.3) if it abandons the
connection prior to the effective value.
An endpoint restarts its idle timer when a packet from its peer is
received and processed successfully. An endpoint also restarts its
idle timer when sending an ack-eliciting packet if no other ack-
eliciting packets have been sent since last receiving and processing
a packet. Restarting this timer when sending a packet ensures that
connections are not closed after new activity is initiated.
An endpoint might need to send ack-eliciting packets to avoid an idle
timeout if it is expecting response data, but does not have or is
unable to send application data.
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An endpoint that sends packets close to the effective timeout risks
having them be discarded at the peer, since the peer might enter its
draining state before these packets arrive. An endpoint can send a
PING or another ack-eliciting frame to test the connection for
liveness if the peer could time out soon, such as within a PTO; see
Section 6.6 of [QUIC-RECOVERY]. This is especially useful if any
available application data cannot be safely retried. Note that the
application determines what data is safe to retry.
10.3. Immediate Close
An endpoint sends a CONNECTION_CLOSE frame (Section 19.19) to
terminate the connection immediately. A CONNECTION_CLOSE frame
causes all streams to immediately become closed; open streams can be
assumed to be implicitly reset.
After sending a CONNECTION_CLOSE frame, an endpoint immediately
enters the closing state.
During the closing period, an endpoint that sends a CONNECTION_CLOSE
frame SHOULD respond to any incoming packet that can be decrypted
with another packet containing a CONNECTION_CLOSE frame. Such an
endpoint SHOULD limit the number of packets it generates containing a
CONNECTION_CLOSE frame. For instance, an endpoint could wait for a
progressively increasing number of received packets or amount of time
before responding to a received packet.
An endpoint is allowed to drop the packet protection keys when
entering the closing period (Section 10.1) and send a packet
containing a CONNECTION_CLOSE in response to any UDP datagram that is
received. However, an endpoint without the packet protection keys
cannot identify and discard invalid packets. To avoid creating an
unwitting amplification attack, such endpoints MUST reduce the
frequency with which it sends packets containing a CONNECTION_CLOSE
frame. To minimize the state that an endpoint maintains for a
closing connection, endpoints MAY send the exact same packet.
Note: Allowing retransmission of a closing 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.
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New packets from unverified addresses could be used to create an
amplification attack; see Section 8. To avoid this, endpoints MUST
either limit transmission of CONNECTION_CLOSE frames to validated
addresses or drop packets without response if the response would be
more than three times larger than the received packet.
After receiving a CONNECTION_CLOSE frame, endpoints enter the
draining state. An endpoint that receives a CONNECTION_CLOSE frame
MAY send a single packet containing a CONNECTION_CLOSE 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
CONNECTION_CLOSE 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
to agree to close the connection, after which the application
requests that the connection be closed. The application protocol can
use a CONNECTION_CLOSE frame with an appropriate error code to signal
closure.
10.3.1. Immediate Close During the Handshake
When sending CONNECTION_CLOSE, the goal is to ensure that the peer
will process the frame. Generally, this means sending the frame in a
packet with the highest level of packet protection to avoid the
packet being discarded. After the handshake is confirmed (see
Section 4.1.2 of [QUIC-TLS]), an endpoint MUST send any
CONNECTION_CLOSE frames in a 1-RTT packet. However, prior to
confirming the handshake, it is possible that more advanced packet
protection keys are not available to the peer, so another
CONNECTION_CLOSE frame MAY be sent in a packet that uses a lower
packet protection level. More specifically:
* A client will always know whether the server has Handshake keys
(see Section 17.2.2.1), but it is possible that a server does not
know whether the client has Handshake keys. Under these
circumstances, a server SHOULD send a CONNECTION_CLOSE frame in
both Handshake and Initial packets to ensure that at least one of
them is processable by the client.
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* A client that sends CONNECTION_CLOSE in a 0-RTT packet cannot be
assured of the server has accepted 0-RTT and so sending a
CONNECTION_CLOSE frame in an Initial packet makes it more likely
that the server can receive the close signal, even if the
application error code might not be received.
* Prior to confirming the handshake, a peer might be unable to
process 1-RTT packets, so an endpoint SHOULD send CONNECTION_CLOSE
in both Handshake and 1-RTT packets. A server SHOULD also send
CONNECTION_CLOSE in an Initial packet.
Sending a CONNECTION_CLOSE of type 0x1d in an Initial or Handshake
packet could expose application state or be used to alter application
state. A CONNECTION_CLOSE of type 0x1d MUST be replaced by a
CONNECTION_CLOSE of type 0x1c when sending the frame in Initial or
Handshake packets. Otherwise, information about the application
state might be revealed. Endpoints MUST clear the value of the
Reason Phrase field and SHOULD use the APPLICATION_ERROR code when
converting to a CONNECTION_CLOSE of type 0x1c.
CONNECTION_CLOSE frames sent in multiple packet types can be
coalesced into a single UDP datagram; see Section 12.2.
An endpoint might send a CONNECTION_CLOSE frame in an Initial packet
or in response to unauthenticated information received in Initial or
Handshake packets. Such an immediate close might expose legitimate
connections to a denial of service. QUIC does not include defensive
measures for on-path attacks during the handshake; see Section 21.1.
However, at the cost of reducing feedback about errors for legitimate
peers, some forms of denial of service can be made more difficult for
an attacker if endpoints discard illegal packets rather than
terminating a connection with CONNECTION_CLOSE. For this reason,
endpoints MAY discard packets rather than immediately close if errors
are detected in packets that lack authentication.
An endpoint that has not established state, such as a server that
detects an error in an Initial packet, does not enter the closing
state. An endpoint that has no state for the connection does not
enter a closing or draining period on sending a CONNECTION_CLOSE
frame.
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10.4. Stateless Reset
A stateless reset is provided as an option of last resort for an
endpoint that does not have access to the state of a connection. A
crash or outage might result in peers continuing to send data to an
endpoint that is unable to properly continue the connection. An
endpoint MAY send a stateless reset in response to receiving a packet
that it cannot associate with an active connection.
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 frame if it has sufficient state to do so.
To support this process, a token is sent by endpoints. The token is
carried in the Stateless Reset Token field of a NEW_CONNECTION_ID
frame. Servers can also specify a stateless_reset_token transport
parameter during the handshake that applies to the connection ID that
it selected during the handshake; clients cannot use this transport
parameter because their transport parameters don't have
confidentiality protection. These tokens are protected by
encryption, so only client and server know their value. Tokens are
invalidated when their associated connection ID is retired via a
RETIRE_CONNECTION_ID frame (Section 19.16).
An endpoint that receives packets that it cannot process sends a
packet in the following layout:
Stateless Reset {
Fixed Bits (2) = 1,
Unpredictable Bits (38..),
Stateless Reset Token (128),
}
Figure 9: Stateless Reset Packet
This design ensures that a stateless reset packet is - to the extent
possible - indistinguishable from a regular packet with a short
header.
A stateless reset uses an entire UDP datagram, starting with the
first two bits of the packet header. The remainder of the first byte
and an arbitrary number of bytes following it that are set to
unpredictable values. The last 16 bytes of the datagram contain a
Stateless Reset Token.
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To entities other than its intended recipient, a stateless reset will
appear to be a packet with a short header. For the stateless reset
to appear as a valid QUIC packet, the Unpredictable Bits field needs
to include at least 38 bits of data (or 5 bytes, less the two fixed
bits).
A minimum size of 21 bytes does not guarantee that a stateless reset
is difficult to distinguish from other packets if the recipient
requires the use of a connection ID. To prevent a resulting
stateless reset from being trivially distinguishable from a valid
packet, all packets sent by an endpoint SHOULD be padded to at least
22 bytes longer than the minimum connection ID that the endpoint
might use. An endpoint that sends a stateless reset in response to
packet that is 43 bytes or less in length SHOULD send a stateless
reset that is one byte shorter than the packet it responds to.
These values assume that the Stateless Reset Token is the same as the
minimum expansion of the packet protection AEAD. Additional
unpredictable bytes are necessary if the endpoint could have
negotiated a packet protection scheme with a larger minimum
expansion.
An endpoint MUST NOT send a stateless reset that is three times or
more larger than the packet it receives to avoid being used for
amplification. Section 10.4.3 describes additional limits on
stateless reset size.
Endpoints MUST discard packets that are too small to be valid QUIC
packets. With the set of AEAD functions defined in [QUIC-TLS],
packets that are smaller than 21 bytes are never valid.
Endpoints MUST send stateless reset packets formatted as a packet
with a short header. However, endpoints MUST treat any packet ending
in a valid stateless reset token as a stateless reset, as other QUIC
versions might allow the use of a long header.
An endpoint MAY send a stateless reset in response to a packet with a
long header. Sending a stateless reset is not effective prior to the
stateless reset token being available to a peer. In this QUIC
version, packets with a long header are only used during connection
establishment. Because the stateless reset token is not available
until connection establishment is complete or near completion,
ignoring an unknown packet with a long header might be as effective
as sending a stateless reset.
An endpoint cannot determine the Source Connection ID from a packet
with a short header, therefore it cannot set the Destination
Connection ID in the stateless reset packet. The Destination
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Connection ID will therefore differ from the value used in previous
packets. A random Destination Connection ID makes the connection ID
appear to be the result of moving to a new connection ID that was
provided using a NEW_CONNECTION_ID frame (Section 19.15).
Using a randomized connection ID results in two problems:
* The packet might not reach the peer. If the Destination
Connection ID is critical for routing toward the peer, then this
packet could be incorrectly routed. This might also trigger
another Stateless Reset in response; see Section 10.4.3. A
Stateless Reset that is not correctly routed is an ineffective
error detection and recovery mechanism. In this case, endpoints
will need to rely on other methods - such as timers - to detect
that the connection has failed.
* The randomly generated connection ID can be used by entities other
than the peer to identify this as a potential stateless reset. An
endpoint that occasionally uses different connection IDs might
introduce some uncertainty about this.
This stateless reset design is specific to QUIC version 1. An
endpoint that supports multiple versions of QUIC needs to generate a
stateless reset that will be accepted by peers that support any
version that the endpoint 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 bytes for carrying data.
10.4.1. Detecting a Stateless Reset
An endpoint detects a potential stateless reset using the trailing 16
bytes of the UDP datagram. An endpoint remembers all Stateless Reset
Tokens associated with the connection IDs and remote addresses for
datagrams it has recently sent. This includes Stateless Reset Tokens
from NEW_CONNECTION_ID frames and the server's transport parameters
but excludes Stateless Reset Tokens associated with connection IDs
that are either unused or retired. The endpoint identifies a
received datagram as a stateless reset by comparing the last 16 bytes
of the datagram with all Stateless Reset Tokens associated with the
remote address on which the datagram was received.
This comparison can be performed for every inbound datagram.
Endpoints MAY skip this check if any packet from a datagram is
successfully processed. However, the comparison MUST be performed
when the first packet in an incoming datagram either cannot be
associated with a connection, or cannot be decrypted.
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An endpoint MUST NOT check for any Stateless Reset Tokens associated
with connection IDs it has not used or for connection IDs that have
been retired.
When comparing a datagram to Stateless Reset Token values, endpoints
MUST perform the comparison without leaking information about the
value of the token. For example, performing this comparison in
constant time protects the value of individual Stateless Reset Tokens
from information leakage through timing side channels. Another
approach would be to store and compare the transformed values of
Stateless Reset Tokens instead of the raw token values, where the
transformation is defined as a cryptographically-secure pseudo-random
function using a secret key (e.g., block cipher, HMAC [RFC2104]). An
endpoint is not expected to protect information about whether a
packet was successfully decrypted, or the number of valid Stateless
Reset Tokens.
If the last 16 bytes of the datagram are identical in value to a
Stateless Reset Token, the endpoint MUST enter the draining period
and not send any further packets on this connection.
10.4.2. Calculating a Stateless Reset Token
The stateless reset token MUST be difficult to guess. In order to
create a Stateless Reset Token, an endpoint could randomly generate
[RFC4086] a secret for every connection that it creates. However,
this presents a coordination problem when there are multiple
instances in a cluster or a storage problem for an endpoint 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 a static key and the
connection ID chosen by the endpoint (see Section 5.1) as input. An
endpoint could use HMAC [RFC2104] (for example, HMAC(static_key,
connection_id)) or HKDF [RFC5869] (for example, using the static key
as input keying material, with the connection ID as salt). The
output of this function is truncated to 16 bytes to produce the
Stateless Reset Token for that connection.
An endpoint that loses state can use the same method to generate a
valid Stateless Reset Token. The connection ID comes from the packet
that the endpoint receives.
This design relies on the peer always sending a connection ID in its
packets so that the endpoint can use the connection ID from a packet
to reset the connection. An endpoint that uses this design MUST
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either use the same connection ID length for all connections or
encode the length of the connection ID such that it can be recovered
without state. In addition, it cannot provide a zero-length
connection ID.
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
connection ID and static key MUST NOT be used for another connection.
A denial of service attack is possible if the same connection ID is
used by instances that share a static key, or if an attacker can
cause a packet to be routed to an instance that has no state but the
same static key; see Section 21.9. A connection ID from a connection
that is reset by revealing the Stateless Reset Token MUST NOT be
reused for new connections at nodes that share a static key.
The same Stateless Reset Token MUST NOT be used for multiple
connection IDs. Endpoints are not required to compare new values
against all previous values, but a duplicate value MAY be treated as
a connection error of type PROTOCOL_VIOLATION.
Note that Stateless Reset packets do not have any cryptographic
protection.
10.4.3. Looping
The design of a Stateless Reset is such that without knowing the
stateless reset token it is indistinguishable from a valid packet.
For instance, if a server sends a Stateless Reset to another server
it might receive another Stateless Reset in response, which could
lead to an infinite exchange.
An endpoint MUST ensure that every Stateless Reset that it sends is
smaller than the packet which triggered it, unless it maintains state
sufficient to prevent looping. In the event of a loop, this results
in packets eventually being too small to trigger a response.
An endpoint can remember the number of Stateless Reset packets that
it has sent and stop generating new Stateless Reset packets once a
limit is reached. Using separate limits for different remote
addresses will ensure that Stateless Reset packets can be used to
close connections when other peers or connections have exhausted
limits.
Reducing the size of a Stateless Reset below 41 bytes means that the
packet could reveal to an observer that it is a Stateless Reset,
depending upon the length of the peer's connection IDs. Conversely,
refusing to send a Stateless Reset in response to a small packet
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might result in Stateless Reset not being useful in detecting cases
of broken connections where only very small packets are sent; such
failures might only be detected by other means, such as timers.
11. Error Handling
An endpoint that detects an error SHOULD signal the existence of that
error to its peer. Both transport-level and application-level errors
can affect an entire connection; see Section 11.1. Only application-
level errors can be isolated to a single stream; see Section 11.2.
The most appropriate error code (Section 20) 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; though these are worded as requirements, different
implementation strategies might lead to different errors being
reported. In particular, an endpoint MAY use any applicable error
code when it detects an error condition; a generic error code (such
as PROTOCOL_VIOLATION or INTERNAL_ERROR) can always be used in place
of specific error codes.
A stateless reset (Section 10.4) is not suitable for any error that
can be signaled with a CONNECTION_CLOSE or RESET_STREAM frame. A
stateless reset MUST NOT be used by an endpoint that has the state
necessary to send a frame on the connection.
11.1. 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 frame (Section 19.19). 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-specific variant of the CONNECTION_CLOSE frame.
Errors that are specific to the transport, including all those
described in this document, are carried in the QUIC-specific variant
of the CONNECTION_CLOSE frame.
A CONNECTION_CLOSE frame could be sent in a packet that is lost. An
endpoint SHOULD be prepared to retransmit a packet containing a
CONNECTION_CLOSE frame 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.
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An endpoint that chooses not to retransmit packets containing a
CONNECTION_CLOSE frame 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 10.4).
11.2. Stream Errors
If an application-level error affects a single stream, but otherwise
leaves the connection in a recoverable state, the endpoint can send a
RESET_STREAM frame (Section 19.4) with an appropriate error code to
terminate just the affected stream.
Resetting a stream without the involvement of the application
protocol could cause the application protocol to enter an
unrecoverable state. RESET_STREAM MUST only be instigated by the
application protocol that uses QUIC.
The semantics of the application error code carried in RESET_STREAM
are defined by the application protocol. Only the application
protocol is able to cause a stream to be terminated. A local
instance of the application protocol uses a direct API call and a
remote instance uses the STOP_SENDING frame, which triggers an
automatic RESET_STREAM.
Application protocols SHOULD define rules for handling streams that
are prematurely cancelled by either endpoint.
12. Packets and Frames
QUIC endpoints communicate by exchanging packets. Packets have
confidentiality and integrity protection; see Section 12.1. Packets
are carried in UDP datagrams; see Section 12.2.
This version of QUIC uses the long packet header during connection
establishment; see Section 17.2. Packets with the long header are
Initial (Section 17.2.2), 0-RTT (Section 17.2.3), Handshake
(Section 17.2.4), and Retry (Section 17.2.5). Version negotiation
uses a version-independent packet with a long header; see
Section 17.2.1.
Packets with the short header are designed for minimal overhead and
are used after a connection is established and 1-RTT keys are
available; see Section 17.3.
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12.1. Protected Packets
All QUIC packets except Version Negotiation packets use authenticated
encryption with additional data (AEAD) [RFC5116] to provide
confidentiality and integrity protection. Retry packets use AEAD to
provide integrity protection. Details of packet protection are found
in [QUIC-TLS]; this section includes an overview of the process.
Initial packets are protected using keys that are statically derived.
This packet protection is not effective confidentiality protection.
Initial protection only exists to ensure that the sender of the
packet is on the network path. Any entity that receives the Initial
packet from a client can recover the keys necessary to remove packet
protection or to generate packets that will be successfully
authenticated.
All other packets are protected with keys derived from the
cryptographic handshake. The type of the packet from the long header
or key phase from the short header are used to identify which
encryption keys are used. Packets protected with 0-RTT and 1-RTT
keys are expected to have confidentiality and data origin
authentication; the cryptographic handshake ensures that only the
communicating endpoints receive the corresponding keys.
The packet number field contains a packet number, which has
additional confidentiality protection that is applied after packet
protection is applied; see [QUIC-TLS] for details. The underlying
packet number increases with each packet sent in a given packet
number space; see Section 12.3 for details.
12.2. Coalescing Packets
Initial (Section 17.2.2), 0-RTT (Section 17.2.3), and Handshake
(Section 17.2.4) packets contain a Length field, which determines the
end of the packet. The length includes both the Packet Number and
Payload fields, both of which are confidentiality protected and
initially of unknown length. The length of the Payload field is
learned once header protection is removed.
Using the Length field, a sender can coalesce multiple QUIC packets
into one UDP datagram. This can reduce the number of UDP datagrams
needed to complete the cryptographic handshake and start sending
data. This can also be used to construct PMTU probes; see
Section 14.3.1. Receivers MUST be able to process coalesced packets.
Coalescing packets in order of increasing encryption levels (Initial,
0-RTT, Handshake, 1-RTT; see Section 4.1.4 of [QUIC-TLS]) makes it
more likely the receiver will be able to process all the packets in a
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single pass. A packet with a short header does not include a length,
so it can only be the last packet included in a UDP datagram. An
endpoint SHOULD NOT coalesce multiple packets at the same encryption
level.
Senders MUST NOT coalesce QUIC packets for different connections into
a single UDP datagram. Receivers SHOULD ignore any subsequent
packets with a different Destination Connection ID than the first
packet in the datagram.
Every QUIC packet that is coalesced into a single UDP datagram is
separate and complete. The receiver of coalesced QUIC packets MUST
individually process each QUIC packet and separately acknowledge
them, as if they were received as the payload of different UDP
datagrams. For example, if decryption fails (because the keys are
not available or any other reason), the receiver MAY either discard
or buffer the packet for later processing and MUST attempt to process
the remaining packets.
Retry packets (Section 17.2.5), Version Negotiation packets
(Section 17.2.1), and packets with a short header (Section 17.3) do
not contain a Length field and so cannot be followed by other packets
in the same UDP datagram. Note also that there is no situation where
a Retry or Version Negotiation packet is coalesced with another
packet.
12.3. Packet Numbers
The packet number is an integer in the range 0 to 2^62-1. This
number is used in determining the cryptographic nonce for packet
protection. Each endpoint maintains a separate packet number for
sending and receiving.
Packet numbers are limited to this range because they need to be
representable in whole in the Largest Acknowledged field of an ACK
frame (Section 19.3). When present in a long or short header
however, packet numbers are reduced and encoded in 1 to 4 bytes; see
Section 17.1.
Version Negotiation (Section 17.2.1) and Retry (Section 17.2.5)
packets do not include a packet number.
Packet numbers are divided into 3 spaces in QUIC:
* Initial space: All Initial packets (Section 17.2.2) are in this
space.
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* Handshake space: All Handshake packets (Section 17.2.4) are in
this space.
* Application data space: All 0-RTT and 1-RTT encrypted packets
(Section 12.1) are in this space.
As described in [QUIC-TLS], each packet type uses different
protection keys.
Conceptually, a packet number space is the context in which a packet
can be processed and acknowledged. Initial packets can only be sent
with Initial packet protection keys and acknowledged in packets which
are also Initial packets. Similarly, Handshake packets are sent at
the Handshake encryption level and can only be acknowledged in
Handshake packets.
This enforces cryptographic separation between the data sent in the
different packet sequence number spaces. Packet numbers in each
space start at packet number 0. Subsequent packets sent in the same
packet number space MUST increase the packet number by at least one.
0-RTT and 1-RTT data exist in the same packet number space to make
loss recovery algorithms easier to implement between the two packet
types.
A QUIC endpoint MUST NOT reuse a packet number within the same packet
number space in one connection. 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; an endpoint
MAY send a Stateless Reset (Section 10.4) in response to further
packets that it receives.
A receiver MUST discard a newly unprotected packet unless it is
certain that it has not processed another packet with the same packet
number from the same packet number space. Duplicate suppression MUST
happen after removing packet protection for the reasons described in
Section 9.3 of [QUIC-TLS]. An efficient algorithm for duplicate
suppression can be found in Section 3.4.3 of [RFC4303].
Packet number encoding at a sender and decoding at a receiver are
described in Section 17.1.
12.4. Frames and Frame Types
The payload of QUIC packets, after removing packet protection,
consists of a sequence of complete frames, as shown in Figure 10.
Version Negotiation, Stateless Reset, and Retry packets do not
contain frames.
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Packet Payload {
Frame (..) ...,
}
Figure 10: QUIC Payload
The payload of a packet that contains frames MUST contain at least
one frame, and MAY contain multiple frames and multiple frame types.
Frames always fit within a single QUIC packet and cannot span
multiple packets.
Each frame begins with a Frame Type, indicating its type, followed by
additional type-dependent fields:
Frame {
Frame Type (i),
Type-Dependent Fields (..),
}
Figure 11: Generic Frame Layout
The frame types defined in this specification are listed in Table 3.
The Frame Type in ACK, STREAM, MAX_STREAMS, STREAMS_BLOCKED, and
CONNECTION_CLOSE frames is used to carry other frame-specific flags.
For all other frames, the Frame Type field simply identifies the
frame. These frames are explained in more detail in Section 19.
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+-------------+----------------------+---------------+---------+
| Type Value | Frame Type Name | Definition | Packets |
+=============+======================+===============+=========+
| 0x00 | PADDING | Section 19.1 | IH01 |
+-------------+----------------------+---------------+---------+
| 0x01 | PING | Section 19.2 | IH01 |
+-------------+----------------------+---------------+---------+
| 0x02 - 0x03 | ACK | Section 19.3 | IH_1 |
+-------------+----------------------+---------------+---------+
| 0x04 | RESET_STREAM | Section 19.4 | __01 |
+-------------+----------------------+---------------+---------+
| 0x05 | STOP_SENDING | Section 19.5 | __01 |
+-------------+----------------------+---------------+---------+
| 0x06 | CRYPTO | Section 19.6 | IH_1 |
+-------------+----------------------+---------------+---------+
| 0x07 | NEW_TOKEN | Section 19.7 | ___1 |
+-------------+----------------------+---------------+---------+
| 0x08 - 0x0f | STREAM | Section 19.8 | __01 |
+-------------+----------------------+---------------+---------+
| 0x10 | MAX_DATA | Section 19.9 | __01 |
+-------------+----------------------+---------------+---------+
| 0x11 | MAX_STREAM_DATA | Section 19.10 | __01 |
+-------------+----------------------+---------------+---------+
| 0x12 - 0x13 | MAX_STREAMS | Section 19.11 | __01 |
+-------------+----------------------+---------------+---------+
| 0x14 | DATA_BLOCKED | Section 19.12 | __01 |
+-------------+----------------------+---------------+---------+
| 0x15 | STREAM_DATA_BLOCKED | Section 19.13 | __01 |
+-------------+----------------------+---------------+---------+
| 0x16 - 0x17 | STREAMS_BLOCKED | Section 19.14 | __01 |
+-------------+----------------------+---------------+---------+
| 0x18 | NEW_CONNECTION_ID | Section 19.15 | __01 |
+-------------+----------------------+---------------+---------+
| 0x19 | RETIRE_CONNECTION_ID | Section 19.16 | __01 |
+-------------+----------------------+---------------+---------+
| 0x1a | PATH_CHALLENGE | Section 19.17 | __01 |
+-------------+----------------------+---------------+---------+
| 0x1b | PATH_RESPONSE | Section 19.18 | __01 |
+-------------+----------------------+---------------+---------+
| 0x1c - 0x1d | CONNECTION_CLOSE | Section 19.19 | ih01 |
+-------------+----------------------+---------------+---------+
| 0x1e | HANDSHAKE_DONE | Section 19.20 | ___1 |
+-------------+----------------------+---------------+---------+
Table 3: Frame Types
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The "Packets" column in Table 3 does not form part of the IANA
registry; see Section 22.3. This column lists the types of packets
that each frame type could appear in, indicated by the following
characters:
I: Initial (Section 17.2.2)
H: Handshake (Section 17.2.4)
0: 0-RTT (Section 17.2.3)
1: 1-RTT (Section 17.3)
ih: A CONNECTION_CLOSE frame of type 0x1d cannot appear in Initial
or Handshake packets.
Section 4 of [QUIC-TLS] provides more detail about these
restrictions. Note that all frames can appear in 1-RTT packets.
An endpoint MUST treat the receipt of a frame of unknown type as a
connection error of type FRAME_ENCODING_ERROR.
All QUIC frames are idempotent in this version of QUIC. That is, a
valid frame does not cause undesirable side effects or errors when
received more than once.
The Frame Type field uses a variable length integer encoding (see
Section 16) with one exception. To ensure simple and efficient
implementations of frame parsing, a frame type MUST use the shortest
possible encoding. For frame types defined in this document, this
means a single-byte encoding, even though it is possible to encode
these values as a two-, four- or eight-byte variable length integer.
For instance, though 0x4001 is a legitimate two-byte encoding for a
variable-length integer with a value of 1, PING frames are always
encoded as a single byte with the value 0x01. This rule applies to
all current and future QUIC frame types. An endpoint MAY treat the
receipt of a frame type that uses a longer encoding than necessary as
a connection error of type PROTOCOL_VIOLATION.
13. Packetization and Reliability
A sender bundles one or more frames in a QUIC packet; see
Section 12.4.
A sender can 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
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large numbers of small packets. An implementation MAY use knowledge
about application sending behavior or heuristics 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.
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.
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. Implementations are advised to bundle as few streams as
necessary in outgoing packets without losing transmission efficiency
to underfilled packets.
13.1. Packet Processing
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 enqueued
in preparation to be received by the application protocol, but it
does not require that data is delivered and consumed.
Once the packet has been fully processed, a receiver acknowledges
receipt by sending one or more ACK frames containing the packet
number of the received packet.
13.2. Generating Acknowledgements
Endpoints acknowledge all packets they receive and process. However,
only ack-eliciting packets cause an ACK frame to be sent within the
maximum ack delay. Packets that are not ack-eliciting are only
acknowledged when an ACK frame is sent for other reasons.
When sending a packet for any reason, an endpoint SHOULD attempt to
bundle an ACK frame if one has not been sent recently. Doing so
helps with timely loss detection at the peer.
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In general, frequent feedback from a receiver improves loss and
congestion response, but this has to be balanced against excessive
load generated by a receiver that sends an ACK frame in response to
every ack-eliciting packet. The guidance offered below seeks to
strike this balance.
13.2.1. Sending ACK Frames
Every packet SHOULD be acknowledged at least once, and ack-eliciting
packets MUST be acknowledged at least once within the maximum ack
delay. An endpoint communicates its maximum delay using the
max_ack_delay transport parameter; see Section 18.2. max_ack_delay
declares an explicit contract: an endpoint promises to never
intentionally delay acknowledgments of an ack-eliciting packet by
more than the indicated value. If it does, any excess accrues to the
RTT estimate and could result in spurious or delayed retransmissions
from the peer. For Initial and Handshake packets, a max_ack_delay of
0 is used. The sender uses the receiver's max_ack_delay value in
determining timeouts for timer-based retransmission, as detailed in
Section 5.2.1 of [QUIC-RECOVERY].
An ACK frame SHOULD be generated for at least every second ack-
eliciting packet. This recommendation is in keeping with standard
practice for TCP [RFC5681]. A receiver could decide to send an ACK
frame less frequently if it has information about how frequently the
sender's congestion controller needs feedback, or if the receiver is
CPU or bandwidth constrained.
In order to assist loss detection at the sender, an endpoint SHOULD
send an ACK frame immediately on receiving an ack-eliciting packet
that is out of order. The endpoint SHOULD NOT continue sending ACK
frames immediately unless more ack-eliciting packets are received out
of order. If every subsequent ack-eliciting packet arrives out of
order, then an ACK frame SHOULD be sent immediately for every
received ack-eliciting packet.
Similarly, packets marked with the ECN Congestion Experienced (CE)
codepoint in the IP header SHOULD be acknowledged immediately, to
reduce the peer's response time to congestion events.
As an optimization, a receiver MAY process multiple packets before
sending any ACK frames in response. In this case the receiver can
determine whether an immediate or delayed acknowledgement should be
generated after processing incoming packets.
Packets containing PADDING frames are considered to be in flight for
congestion control purposes [QUIC-RECOVERY]. Sending only PADDING
frames might cause the sender to become limited by the congestion
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controller with no acknowledgments forthcoming from the receiver.
Therefore, a sender SHOULD ensure that other frames are sent in
addition to PADDING frames to elicit acknowledgments from the
receiver.
An endpoint that is only sending ACK frames will not receive
acknowledgments from its peer unless those acknowledgements are
included in packets with ack-eliciting frames. An endpoint SHOULD
bundle ACK frames with other frames when there are new ack-eliciting
packets to acknowledge. When only non-ack-eliciting packets need to
be acknowledged, an endpoint MAY wait until an ack-eliciting packet
has been received to bundle an ACK frame with outgoing frames.
The algorithms in [QUIC-RECOVERY] are resilient to receivers that do
not follow guidance offered above. However, an implementor should
only deviate from these requirements after careful consideration of
the performance implications of doing so.
Packets containing only ACK frames are not congestion controlled, so
there are limits on how frequently they can be sent. An endpoint
MUST NOT send more than one ACK-frame-only packet in response to
receiving an ack-eliciting packet. An endpoint MUST NOT send a non-
ack-eliciting packet in response to a non-ack-eliciting packet, even
if there are packet gaps which precede the received packet. Limiting
ACK frames avoids an infinite feedback loop of acknowledgements,
which could prevent the connection from ever becoming idle. However,
the endpoint acknowledges non-ACK-eliciting packets when it sends an
ACK frame.
An endpoint SHOULD treat receipt of an acknowledgment for a packet it
did not send as a connection error of type PROTOCOL_VIOLATION, if it
is able to detect the condition.
13.2.2. Managing ACK Ranges
When an ACK frame is sent, one or more ranges of acknowledged packets
are included. Including older packets reduces the chance of spurious
retransmits caused by losing previously sent ACK frames, at the cost
of larger ACK frames.
ACK frames SHOULD always acknowledge the most recently received
packets, and the more out-of-order the packets are, the more
important it is to send an updated ACK frame quickly, to prevent the
peer from declaring a packet as lost and spuriously retransmitting
the frames it contains. An ACK frame is expected to fit within a
single QUIC packet. If it does not, then older ranges (those with
the smallest packet numbers) are omitted.
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Section 13.2.3 and Section 13.2.4 describe an exemplary approach for
determining what packets to acknowledge in each ACK frame. Though
the goal of these algorithms is to generate an acknowledgment for
every packet that is processed, it is still possible for
acknowledgments to be lost. A sender cannot expect to receive an
acknowledgment for every packet that the receiver processes.
13.2.3. Receiver Tracking of ACK Frames
When a packet containing an ACK frame is sent, the largest
acknowledged in that frame may be saved. When a packet containing an
ACK frame is acknowledged, the receiver can stop acknowledging
packets less than or equal to the largest acknowledged in the sent
ACK frame.
In cases without ACK frame loss, this algorithm allows for a minimum
of 1 RTT of reordering. In cases with ACK frame loss and reordering,
this approach does not guarantee that every acknowledgement is seen
by the sender before it is no longer included in the ACK frame.
Packets could be received out of order and all subsequent ACK frames
containing them could be lost. In this case, the loss recovery
algorithm could cause spurious retransmits, but the sender will
continue making forward progress.
13.2.4. Limiting ACK Ranges
A receiver limits the number of ACK Ranges (Section 19.3.1) it
remembers and sends in ACK frames, both to limit the size of ACK
frames and to avoid resource exhaustion. After receiving
acknowledgments for an ACK frame, the receiver SHOULD stop tracking
those acknowledged ACK Ranges.
It is possible that retaining many ACK Ranges could cause an ACK
frame to become too large. A receiver can discard unacknowledged ACK
Ranges to limit ACK frame size, at the cost of increased
retransmissions from the sender. This is necessary if an ACK frame
would be too large to fit in a packet, however receivers MAY also
limit ACK frame size further to preserve space for other frames.
A receiver MUST retain an ACK Range unless it can ensure that it will
not subsequently accept packets with numbers in that range.
Maintaining a minimum packet number that increases as ranges are
discarded is one way to achieve this with minimal state.
Receivers can discard all ACK Ranges, but they MUST retain the
largest packet number that has been successfully processed as that is
used to recover packet numbers from subsequent packets; see
Section 17.1.
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A receiver SHOULD include an ACK Range containing the largest
received packet number in every ACK frame. The Largest Acknowledged
field is used in ECN validation at a sender and including a lower
value than what was included in a previous ACK frame could cause ECN
to be unnecessarily disabled; see Section 13.4.2.
A receiver that sends only non-ack-eliciting packets, such as ACK
frames, might not receive an acknowledgement for a long period of
time. This could cause the receiver to maintain state for a large
number of ACK frames for a long period of time, and ACK frames it
sends could be unnecessarily large. In such a case, a receiver could
bundle a PING or other small ack-eliciting frame occasionally, such
as once per round trip, to elicit an ACK from the peer.
A receiver MUST NOT bundle an ack-eliciting frame with all packets
that would otherwise be non-ack-eliciting, to avoid an infinite
feedback loop of acknowledgements.
13.2.5. Measuring and Reporting Host Delay
An endpoint measures the delays intentionally introduced between the
time the packet with the largest packet number is received and the
time an acknowledgment is sent. The endpoint encodes this delay in
the Ack Delay field of an ACK frame; see Section 19.3. This allows
the receiver of the ACK to adjust for any intentional delays, which
is important for getting a better estimate of the path RTT when
acknowledgments are delayed. A packet might be held in the OS kernel
or elsewhere on the host before being processed. An endpoint MUST
NOT include delays that it does not control when populating the Ack
Delay field in an ACK frame.
13.2.6. ACK Frames and Packet Protection
ACK frames MUST only be carried in a packet that has the same packet
number space as the packet being ACKed; see Section 12.1. For
instance, packets that are protected with 1-RTT keys MUST be
acknowledged in packets that are also protected with 1-RTT keys.
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.
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13.3. Retransmission of Information
QUIC packets that are determined to be lost are not retransmitted
whole. The same applies to the frames that are contained within lost
packets. Instead, the information that might be carried in frames is
sent again in new frames as needed.
New frames and packets are used to carry information that is
determined to have been lost. In general, information is sent again
when a packet containing that information is determined to be lost
and sending ceases when a packet containing that information is
acknowledged.
* Data sent in CRYPTO frames is retransmitted according to the rules
in [QUIC-RECOVERY], until all data has been acknowledged. Data in
CRYPTO frames for Initial and Handshake packets is discarded when
keys for the corresponding packet number space are discarded.
* Application data sent in STREAM frames is retransmitted in new
STREAM frames unless the endpoint has sent a RESET_STREAM for that
stream. Once an endpoint sends a RESET_STREAM frame, no further
STREAM frames are needed.
* ACK frames carry the most recent set of acknowledgements and the
Ack Delay from the largest acknowledged packet, as described in
Section 13.2.1. Delaying the transmission of packets containing
ACK frames or sending old ACK frames can cause the peer to
generate an inflated RTT sample or unnecessarily disable ECN.
* Cancellation of stream transmission, as carried in a RESET_STREAM
frame, is sent until acknowledged or until all stream data is
acknowledged by the peer (that is, either the "Reset Recvd" or
"Data Recvd" state is reached on the sending part of the stream).
The content of a RESET_STREAM frame MUST NOT change when it is
sent again.
* Similarly, a request to cancel stream transmission, as encoded in
a STOP_SENDING frame, is sent until the receiving part of the
stream enters either a "Data Recvd" or "Reset Recvd" state; see
Section 3.5.
* Connection close signals, including packets that contain
CONNECTION_CLOSE frames, are not sent again when packet loss is
detected, but as described in Section 10.
* The current connection maximum data is sent in MAX_DATA frames.
An updated value is sent in a MAX_DATA frame if the packet
containing the most recently sent MAX_DATA frame is declared lost,
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or when the endpoint decides to update the limit. Care is
necessary to avoid sending this frame too often as the limit can
increase frequently and cause an unnecessarily large number of
MAX_DATA frames to be sent.
* The current maximum stream data offset is sent in MAX_STREAM_DATA
frames. Like MAX_DATA, an updated value is sent when the packet
containing the most recent MAX_STREAM_DATA frame for a stream is
lost or when the limit is updated, with care taken to prevent the
frame from being sent too often. An endpoint SHOULD stop sending
MAX_STREAM_DATA frames when the receiving part of the stream
enters a "Size Known" state.
* The limit on streams of a given type is sent in MAX_STREAMS
frames. Like MAX_DATA, an updated value is sent when a packet
containing the most recent MAX_STREAMS for a stream type frame is
declared lost or when the limit is updated, with care taken to
prevent the frame from being sent too often.
* Blocked signals are carried in DATA_BLOCKED, STREAM_DATA_BLOCKED,
and STREAMS_BLOCKED frames. DATA_BLOCKED frames have connection
scope, STREAM_DATA_BLOCKED frames have stream scope, and
STREAMS_BLOCKED frames are scoped to a specific stream type. New
frames are sent if packets containing the most recent frame for a
scope is lost, but only while the endpoint is blocked on the
corresponding limit. These frames always include the limit that
is causing blocking at the time that they are transmitted.
* A liveness or path validation check using PATH_CHALLENGE frames is
sent periodically until a matching PATH_RESPONSE frame is received
or until there is no remaining need for liveness or path
validation checking. PATH_CHALLENGE frames include a different
payload each time they are sent.
* Responses to path validation using PATH_RESPONSE frames are sent
just once. The peer is expected to send more PATH_CHALLENGE
frames as necessary to evoke additional PATH_RESPONSE frames.
* New connection IDs are sent in NEW_CONNECTION_ID frames and
retransmitted if the packet containing them is lost.
Retransmissions of this frame carry the same sequence number
value. Likewise, retired connection IDs are sent in
RETIRE_CONNECTION_ID frames and retransmitted if the packet
containing them is lost.
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* NEW_TOKEN frames are retransmitted if the packet containing them
is lost. No special support is made for detecting reordered and
duplicated NEW_TOKEN frames other than a direct comparison of the
frame contents.
* PING and PADDING frames contain no information, so lost PING or
PADDING frames do not require repair.
* The HANDSHAKE_DONE frame MUST be retransmitted until it is
acknowledged.
Endpoints SHOULD prioritize retransmission of data over sending new
data, unless priorities specified by the application indicate
otherwise; see Section 2.3.
Even though a sender is encouraged to assemble frames containing up-
to-date information every time it sends a packet, it is not forbidden
to retransmit copies of frames from lost packets. A sender that
retransmits copies of frames needs to handle decreases in available
payload size due to change in packet number length, connection ID
length, and path MTU. A receiver MUST accept packets containing an
outdated frame, such as a MAX_DATA frame carrying a smaller maximum
data than one found in an older packet.
Upon detecting losses, a sender MUST take appropriate congestion
control action. The details of loss detection and congestion control
are described in [QUIC-RECOVERY].
13.4. Explicit Congestion Notification
QUIC endpoints can use Explicit Congestion Notification (ECN)
[RFC3168] to detect and respond to network congestion. ECN allows a
network node to indicate congestion in the network by setting a
codepoint in the IP header of a packet instead of dropping it.
Endpoints react to congestion by reducing their sending rate in
response, as described in [QUIC-RECOVERY].
To use ECN, QUIC endpoints first determine whether a path supports
ECN marking and the peer is able to access the ECN codepoint in the
IP header. A network path does not support ECN if ECN marked packets
get dropped or ECN markings are rewritten on the path. An endpoint
validates the use of ECN on the path, both during connection
establishment and when migrating to a new path (Section 9).
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13.4.1. ECN Counts
On receiving a QUIC packet with an ECT or CE codepoint, an ECN-
enabled endpoint that can access the ECN codepoints from the
enclosing IP packet increases the corresponding ECT(0), ECT(1), or CE
count, and includes these counts in subsequent ACK frames; see
Section 13.2 and Section 19.3. Note that this requires being able to
read the ECN codepoints from the enclosing IP packet, which is not
possible on all platforms.
A packet detected by a receiver as a duplicate does not affect the
receiver's local ECN codepoint counts; see (Section 21.8) for
relevant security concerns.
If an endpoint receives a QUIC packet without an ECT or CE codepoint
in the IP packet header, it responds per Section 13.2 with an ACK
frame without increasing any ECN counts. If an endpoint does not
implement ECN support or does not have access to received ECN
codepoints, it does not increase ECN counts.
Coalesced packets (see Section 12.2) mean that several packets can
share the same IP header. The ECN counter for the ECN codepoint
received in the associated IP header are incremented once for each
QUIC packet, not per enclosing IP packet or UDP datagram.
Each packet number space maintains separate acknowledgement state and
separate ECN counts. For example, if one each of an Initial, 0-RTT,
Handshake, and 1-RTT QUIC packet are coalesced, the corresponding
counts for the Initial and Handshake packet number space will be
incremented by one and the counts for the 1-RTT packet number space
will be increased by two.
13.4.2. ECN Validation
It is possible for faulty network devices to corrupt or erroneously
drop packets with ECN markings. To provide robust connectivity in
the presence of such devices, each endpoint independently validates
ECN counts and disables ECN if errors are detected.
Endpoints validate ECN for packets sent on each network path
independently. An endpoint thus validates ECN on new connection
establishment, when switching to a new server preferred address, and
on active connection migration to a new path. Appendix B describes
one possible algorithm for testing paths for ECN support.
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Even if an endpoint does not use ECN markings on packets it
transmits, the endpoint MUST provide feedback about ECN markings
received from the peer if they are accessible. Failing to report ECN
counts will cause the peer to disable ECN marking.
13.4.2.1. Sending ECN Markings
To start ECN validation, an endpoint SHOULD do the following when
sending packets on a new path to a peer:
* Set the ECT(0) codepoint in the IP header of early outgoing
packets sent on a new path to the peer [RFC8311].
* If all packets that were sent with the ECT(0) codepoint are
eventually deemed lost [QUIC-RECOVERY], validation is deemed to
have failed.
To reduce the chances of misinterpreting congestive loss as packets
dropped by a faulty network element, an endpoint could set the ECT(0)
codepoint in the first ten outgoing packets on a path, or for a
period of three RTTs, whichever occurs first.
Implementations MAY experiment with and use other strategies for use
of ECN. Other methods of probing paths for ECN support are possible,
as are different marking strategies. Implementations can also use
the ECT(1) codepoint, as specified in [RFC8311].
13.4.2.2. Receiving ACK Frames
An endpoint that sets ECT(0) or ECT(1) codepoints on packets it
transmits MUST use the following steps on receiving an ACK frame to
validate ECN.
* If this ACK frame newly acknowledges a packet that the endpoint
sent with either ECT(0) or ECT(1) codepoints set, and if no ECN
feedback is present in the ACK frame, validation fails. This step
protects against both a network element that zeroes out ECN bits
and a peer that is unable to access ECN markings, since the peer
could respond without ECN feedback in these cases.
* For validation to succeed, the total increase in ECT(0), ECT(1),
and CE counts MUST be no smaller than the total number of QUIC
packets sent with an ECT codepoint that are newly acknowledged in
this ACK frame. This step detects any network remarking from
ECT(0), ECT(1), or CE codepoints to Not-ECT.
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* Any increase in either ECT(0) or ECT(1) counts, plus any increase
in the CE count, MUST be no smaller than the number of packets
sent with the corresponding ECT codepoint that are newly
acknowledged in this ACK frame. This step detects any erroneous
network remarking from ECT(0) to ECT(1) (or vice versa).
Processing ECN counts out of order can result in validation failure.
An endpoint SHOULD NOT perform this validation if this ACK frame does
not advance the largest packet number acknowledged in this
connection.
An endpoint could miss acknowledgements for a packet when ACK frames
are lost. It is therefore possible for the total increase in ECT(0),
ECT(1), and CE counts to be greater than the number of packets
acknowledged in an ACK frame. When this happens, and if validation
succeeds, the local reference counts MUST be increased to match the
counts in the ACK frame.
13.4.2.3. Validation Outcomes
If validation fails, then the endpoint stops sending ECN markings in
subsequent IP packets with the expectation that either the network
path or the peer does not support ECN.
Upon successful validation, an endpoint can continue to set ECT
codepoints in subsequent packets with the expectation that the path
is ECN-capable. Network routing and path elements can change mid-
connection however; an endpoint MUST disable ECN if validation fails
at any point in the connection.
Even if validation fails, an endpoint MAY revalidate ECN on the same
path at any later time in the connection.
14. Packet Size
The QUIC packet size includes the QUIC header and protected payload,
but not the UDP or IP header.
A client MUST expand the payload of all UDP datagrams carrying
Initial packets to at least 1200 bytes, by adding PADDING frames to
the Initial packet or by coalescing the Initial packet; see
Section 12.2. Sending a UDP datagram of this size ensures that the
network path from the client to the server supports a reasonable
Maximum Transmission Unit (MTU). Padding datagrams also helps reduce
the amplitude of amplification attacks caused by server responses
toward an unverified client address; see Section 8.
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Enforcement of the max_udp_payload_size transport parameter
(Section 18.2) might act as an additional limit on packet size.
Exceeding this limit can be avoided once the value is known.
However, prior to learning the value of the transport parameter,
endpoints risk datagrams being lost if they send packets larger than
1200 bytes.
Datagrams containing Initial packets MAY exceed 1200 bytes if the
client believes that the network path and peer both support the size
that it chooses.
UDP datagrams MUST NOT be fragmented at the IP layer. In IPv4
[IPv4], the DF bit MUST be set to prevent fragmentation on the path.
A server MUST discard an Initial packet that is carried in a UDP
datagram with a payload that is less than 1200 bytes. A server MAY
also immediately close the connection by sending a CONNECTION_CLOSE
frame with an error code of PROTOCOL_VIOLATION; see Section 10.3.1.
The server MUST also limit the number of bytes it sends before
validating the address of the client; see Section 8.
14.1. Path Maximum Transmission Unit (PMTU)
The Path Maximum Transmission Unit (PMTU) is the maximum size of the
entire IP packet including the IP header, UDP header, and UDP
payload. The UDP payload includes the QUIC packet header, protected
payload, and any authentication fields. The PMTU can depend on path
characteristics, and can therefore change over time. The largest UDP
payload an endpoint sends at any given time is referred to as the
endpoint's maximum packet size.
QUIC depends on a PMTU of at least 1280 bytes. This is the IPv6
minimum size [RFC8200] and is also supported by most modern IPv4
networks. All QUIC packets (except for PMTU probe packets) SHOULD be
sized to fit within the maximum packet size to avoid the packet being
fragmented or dropped [RFC8085].
An endpoint SHOULD use Datagram Packetization Layer PMTU Discovery
([DPLPMTUD]) or implement Path MTU Discovery (PMTUD) [RFC1191]
[RFC8201] to determine whether the path to a destination will support
a desired message size without fragmentation.
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In the absence of these mechanisms, QUIC endpoints SHOULD NOT send IP
packets larger than 1280 bytes. Assuming the minimum IP header size,
this results in a QUIC maximum packet size of 1232 bytes for IPv6 and
1252 bytes for IPv4. A QUIC implementation MAY be more conservative
in computing the QUIC maximum packet size to allow for unknown tunnel
overheads or IP header options/extensions.
Each pair of local and remote addresses could have a different PMTU.
QUIC implementations that implement any kind of PMTU discovery
therefore SHOULD maintain a maximum packet size for each combination
of local and remote IP addresses.
If a QUIC endpoint determines that the PMTU between any pair of local
and remote IP addresses has fallen below the size needed to support
the smallest allowed maximum packet size, it MUST immediately cease
sending QUIC packets, except for PMTU probe packets, on the affected
path. An endpoint MAY terminate the connection if an alternative
path cannot be found.
14.2. ICMP Packet Too Big Messages
PMTU discovery [RFC1191] [RFC8201] relies on reception of ICMP
messages (e.g., IPv6 Packet Too Big messages) that indicate when a
packet is dropped because it is larger than the local router MTU.
DPLPMTUD can also optionally use these messages. This use of ICMP
messages is potentially vulnerable to off-path attacks that
successfully guess the addresses used on the path and reduce the PMTU
to a bandwidth-inefficient value.
An endpoint MUST ignore an ICMP message that claims the PMTU has
decreased below 1280 bytes.
The requirements for generating ICMP ([RFC1812], [RFC4443]) state
that the quoted packet should contain as much of the original packet
as possible without exceeding the minimum MTU for the IP version.
The size of the quoted packet can actually be smaller, or the
information unintelligible, as described in Section 1.1 of
[DPLPMTUD].
QUIC endpoints SHOULD validate ICMP messages to protect from off-path
injection as specified in [RFC8201] and Section 5.2 of [RFC8085].
This validation SHOULD use the quoted packet supplied in the payload
of an ICMP message to associate the message with a corresponding
transport connection [DPLPMTUD].
ICMP message validation MUST include matching IP addresses and UDP
ports [RFC8085] and, when possible, connection IDs to an active QUIC
session.
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The endpoint SHOULD ignore all ICMP messages that fail validation.
An endpoint MUST NOT increase PMTU based on ICMP messages; see
Section 3, clause 6 of [DPLPMTUD]. Any reduction in the QUIC maximum
packet size in response to ICMP messages MAY be provisional until
QUIC's loss detection algorithm determines that the quoted packet has
actually been lost.
14.3. Datagram Packetization Layer PMTU Discovery
Section 6.3 of [DPLPMTUD] provides considerations for implementing
Datagram Packetization Layer PMTUD (DPLPMTUD) with QUIC.
When implementing the algorithm in Section 5 of [DPLPMTUD], the
initial value of BASE_PMTU SHOULD be consistent with the minimum QUIC
packet size (1232 bytes for IPv6 and 1252 bytes for IPv4).
PING and PADDING frames can be used to generate PMTU probe packets.
These frames might not be retransmitted if a probe packet containing
them is lost. However, these frames do consume congestion window,
which could delay the transmission of subsequent application data.
A PING frame can be included in a PMTU probe to ensure that a valid
probe is acknowledged.
The considerations for processing ICMP messages in the previous
section also apply if these messages are used by DPLPMTUD.
14.3.1. PMTU Probes Containing Source Connection ID
Endpoints that rely on the destination connection ID for routing
incoming QUIC packets are likely to require that the connection ID be
included in PMTU probe packets to route any resulting ICMP messages
(Section 14.2) back to the correct endpoint. However, only long
header packets (Section 17.2) contain source connection IDs, and long
header packets are not decrypted or acknowledged by the peer once the
handshake is complete.
One way to construct a probe for the path MTU is to coalesce (see
Section 12.2) a Handshake packet (Section 17.2.4) with a short header
packet in a single UDP datagram. If the UDP datagram reaches the
endpoint, the Handshake packet will be ignored, but the short header
packet will be acknowledged. If the UDP datagram causes an ICMP
message to be sent, the first part of the datagram will be quoted in
that message. If the source connection ID is within the quoted
portion of the UDP datagram, that could be used for routing.
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15. Versions
QUIC versions are identified using a 32-bit unsigned number.
The version 0x00000000 is reserved to represent version negotiation.
This version of the specification is identified by the number
0x00000001.
Other versions of QUIC might have different properties to this
version. The properties of QUIC that are guaranteed to be consistent
across all versions of the protocol are described in
[QUIC-INVARIANTS].
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 bytes is 1010 (in binary). A
client or server MAY advertise support for any of these reserved
versions.
Reserved version numbers will 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 at
https://github.com/quicwg/base-drafts/wiki/QUIC-Versions.
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16. Variable-Length Integer Encoding
QUIC packets and frames commonly use a variable-length encoding for
non-negative integer values. This encoding ensures that smaller
integer values need fewer bytes to encode.
The QUIC variable-length integer encoding reserves the two most
significant bits of the first byte to encode the base 2 logarithm of
the integer encoding length in bytes. 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 bytes 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 byte sequence c2 19 7c 5e ff 14 e8 8c (in
hexadecimal) decodes to the decimal value 151288809941952652; the
four byte sequence 9d 7f 3e 7d decodes to 494878333; the two byte
sequence 7b bd decodes to 15293; and the single byte 25 decodes to 37
(as does the two byte sequence 40 25).
Error codes (Section 20) and versions (Section 15) are described
using integers, but do not use this encoding.
17. Packet Formats
All numeric values are encoded in network byte order (that is, big-
endian) and all field sizes are in bits. Hexadecimal notation is
used for describing the value of fields.
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17.1. Packet Number Encoding and Decoding
Packet numbers are integers in the range 0 to 2^62-1 (Section 12.3).
When present in long or short packet headers, they are encoded in 1
to 4 bytes. The number of bits required to represent the packet
number is reduced by including the least significant bits of the
packet number.
The encoded packet number is protected as described in Section 5.4 of
[QUIC-TLS].
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
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
bit 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
0xabe8bc, sending a packet with a number of 0xac5c02 requires a
packet number encoding with 16 bits or more; whereas the 24-bit
packet number encoding is needed to send a packet with a number of
0xace8fe.
At a receiver, protection of the packet number is removed prior to
recovering the full packet number. The full packet number is then
reconstructed based on the number of significant bits present, the
value of those bits, and the largest packet number received on a
successfully authenticated packet. Recovering the full packet number
is necessary to successfully remove packet protection.
Once header protection is removed, the 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 0xa82f30ea, then a packet
containing a 16-bit value of 0x9b32 will be decoded as 0xa82f9b32.
Example pseudo-code for packet number decoding can be found in
Appendix A.
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17.2. Long Header Packets
Long Header Packet {
Header Form (1) = 1,
Fixed Bit (1) = 1,
Long Packet Type (2),
Type-Specific Bits (4),
Version (32),
DCID Length (8),
Destination Connection ID (0..160),
SCID Length (8),
Source Connection ID (0..160),
}
Figure 12: Long Header Packet Format
Long headers are used for packets that are sent prior to the
establishment of 1-RTT keys. Once 1-RTT keys are available, a sender
switches to sending packets using the short header (Section 17.3).
The long form allows for special packets - such as the Version
Negotiation packet - to be represented in this uniform fixed-length
packet format. Packets that use the long header contain the
following fields:
Header Form: The most significant bit (0x80) of byte 0 (the first
byte) is set to 1 for long headers.
Fixed Bit: The next bit (0x40) of byte 0 is set to 1. Packets
containing a zero value for this bit are not valid packets in this
version and MUST be discarded.
Long Packet Type: The next two bits (those with a mask of 0x30) of
byte 0 contain a packet type. Packet types are listed in Table 5.
Type-Specific Bits: The lower four bits (those with a mask of 0x0f)
of byte 0 are type-specific.
Version: The QUIC Version is a 32-bit field that follows the first
byte. This field indicates which version of QUIC is in use and
determines how the rest of the protocol fields are interpreted.
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DCID Length: The byte following the version contains the length in
bytes of the Destination Connection ID field that follows it.
This length is encoded as an 8-bit unsigned integer. In QUIC
version 1, this value MUST NOT exceed 20. Endpoints that receive
a version 1 long header with a value larger than 20 MUST drop the
packet. Servers SHOULD be able to read longer connection IDs from
other QUIC versions in order to properly form a version
negotiation packet.
Destination Connection ID: The Destination Connection ID field
follows the DCID Length field and is between 0 and 20 bytes in
length. Section 7.2 describes the use of this field in more
detail.
SCID Length: The byte following the Destination Connection ID
contains the length in bytes of the Source Connection ID field
that follows it. This length is encoded as a 8-bit unsigned
integer. In QUIC version 1, this value MUST NOT exceed 20 bytes.
Endpoints that receive a version 1 long header with a value larger
than 20 MUST drop the packet. Servers SHOULD be able to read
longer connection IDs from other QUIC versions in order to
properly form a version negotiation packet.
Source Connection ID: The Source Connection ID field follows the
SCID Length field and is between 0 and 20 bytes in length.
Section 7.2 describes the use of this field in more detail.
In this version of QUIC, the following packet types with the long
header are defined:
+------+-----------+----------------+
| Type | Name | Section |
+======+===========+================+
| 0x0 | Initial | Section 17.2.2 |
+------+-----------+----------------+
| 0x1 | 0-RTT | Section 17.2.3 |
+------+-----------+----------------+
| 0x2 | Handshake | Section 17.2.4 |
+------+-----------+----------------+
| 0x3 | Retry | Section 17.2.5 |
+------+-----------+----------------+
Table 5: Long Header Packet Types
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The header form bit, connection ID lengths byte, Destination and
Source Connection ID fields, and Version fields of a long header
packet are version-independent. The other fields in the first byte
are version-specific. See [QUIC-INVARIANTS] 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. While type-specific semantics for this
version are described in the following sections, several long-header
packets in this version of QUIC contain these additional fields:
Reserved Bits: Two bits (those with a mask of 0x0c) of byte 0 are
reserved across multiple packet types. These bits are protected
using header protection; see Section 5.4 of [QUIC-TLS]. The value
included prior to protection MUST be set to 0. An endpoint MUST
treat receipt of a packet that has a non-zero value for these
bits, after removing both packet and header protection, as a
connection error of type PROTOCOL_VIOLATION. Discarding such a
packet after only removing header protection can expose the
endpoint to attacks; see Section 9.3 of [QUIC-TLS].
Packet Number Length: In packet types which contain a Packet Number
field, the least significant two bits (those with a mask of 0x03)
of byte 0 contain the length of the packet number, encoded as an
unsigned, two-bit integer that is one less than the length of the
packet number field in bytes. That is, the length of the packet
number field is the value of this field, plus one. These bits are
protected using header protection; see Section 5.4 of [QUIC-TLS].
Length: The length of the remainder of the packet (that is, the
Packet Number and Payload fields) in bytes, encoded as a variable-
length integer (Section 16).
Packet Number: The packet number field is 1 to 4 bytes long. The
packet number has confidentiality protection separate from packet
protection, as described in Section 5.4 of [QUIC-TLS]. The length
of the packet number field is encoded in the Packet Number Length
bits of byte 0; see above.
17.2.1. Version Negotiation Packet
A Version Negotiation packet is inherently not version-specific.
Upon receipt by a client, it will be identified as a Version
Negotiation packet based on the Version field having a value of 0.
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.
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The layout of a Version Negotiation packet is:
Version Negotiation Packet {
Header Form (1) = 1,
Unused (7),
Version (32) = 0,
DCID Length (8),
Destination Connection ID (0..2040),
SCID Length (8),
Source Connection ID (0..2040),
Supported Version (32) ...,
}
Figure 13: Version Negotiation Packet
The value in the Unused field is selected randomly by the server.
Clients MUST ignore the value of this field. Servers SHOULD set the
most significant bit of this field (0x40) to 1 so that Version
Negotiation packets appear to have the Fixed Bit field.
The Version field of a Version Negotiation packet MUST be set to
0x00000000.
The server MUST include the value from the Source Connection ID field
of the packet it receives in the Destination Connection ID field.
The value for Source Connection ID MUST be copied from the
Destination Connection ID of the received packet, which is initially
randomly selected by a client. Echoing both connection IDs gives
clients some assurance that the server received the packet and that
the Version Negotiation packet was not generated by an off-path
attacker.
As future versions of QUIC may support Connection IDs larger than the
version 1 limit, Version Negotiation packets could carry Connection
IDs that are longer than 20 bytes.
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.
The Version Negotiation packet does not include the Packet Number and
Length fields present in other packets that use the long header form.
Consequently, a Version Negotiation packet consumes an entire UDP
datagram.
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A server MUST NOT send more than one Version Negotiation packet in
response to a single UDP datagram.
See Section 6 for a description of the version negotiation process.
17.2.2. Initial Packet
An Initial packet uses long headers with a type value of 0x0. It
carries the first CRYPTO frames sent by the client and server to
perform key exchange, and carries ACKs in either direction.
Initial Packet {
Header Form (1) = 1,
Fixed Bit (1) = 1,
Long Packet Type (2) = 0,
Reserved Bits (2),
Packet Number Length (2),
Version (32),
DCID Length (8),
Destination Connection ID (0..160),
SCID Length (8),
Source Connection ID (0..160),
Token Length (i),
Token (..),
Length (i),
Packet Number (8..32),
Packet Payload (..),
}
Figure 14: Initial Packet
The Initial packet contains a long header as well as the Length and
Packet Number fields. The first byte contains the Reserved and
Packet Number Length bits. Between the SCID and Length fields, there
are two additional fields specific to the Initial packet.
Token Length: A variable-length integer specifying the length of the
Token field, in bytes. This value is zero if no token is present.
Initial packets sent by the server MUST set the Token Length field
to zero; clients that receive an Initial packet with a non-zero
Token Length field MUST either discard the packet or generate a
connection error of type PROTOCOL_VIOLATION.
Token: The value of the token that was previously provided in a
Retry packet or NEW_TOKEN frame.
Packet Payload: The payload of the packet.
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In order to prevent tampering by version-unaware middleboxes, Initial
packets are protected with connection- and version-specific keys
(Initial keys) 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.
The client and server use the Initial packet type for any packet that
contains an initial cryptographic handshake message. This includes
all cases where a new packet containing the initial cryptographic
message needs to be created, such as the packets sent after receiving
a Retry packet (Section 17.2.5).
A server sends its first Initial packet in response to a client
Initial. A server may send multiple Initial packets. The
cryptographic key exchange could require multiple round trips or
retransmissions of this data.
The payload of an Initial packet includes a CRYPTO frame (or frames)
containing a cryptographic handshake message, ACK frames, or both.
PING, PADDING, and CONNECTION_CLOSE frames are also permitted. An
endpoint that receives an Initial packet containing other frames can
either discard the packet as spurious or treat it as a connection
error.
The first packet sent by a client always includes a CRYPTO frame that
contains the start or all of the first cryptographic handshake
message. The first CRYPTO frame sent always begins at an offset of
0; see Section 7.
Note that if the server sends a HelloRetryRequest, the client will
send another series of Initial packets. These Initial packets will
continue the cryptographic handshake and will contain CRYPTO frames
starting at an offset matching the size of the CRYPTO frames sent in
the first flight of Initial packets.
17.2.2.1. Abandoning Initial Packets
A client stops both sending and processing Initial packets when it
sends its first Handshake packet. A server stops sending and
processing Initial packets when it receives its first Handshake
packet. Though packets might still be in flight or awaiting
acknowledgment, no further Initial packets need to be exchanged
beyond this point. Initial packet protection keys are discarded (see
Section 4.10.1 of [QUIC-TLS]) along with any loss recovery and
congestion control state; see Section 6.5 of [QUIC-RECOVERY].
Any data in CRYPTO frames is discarded - and no longer retransmitted
- when Initial keys are discarded.
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17.2.3. 0-RTT
A 0-RTT packet uses long headers with a type value of 0x1, followed
by the Length and Packet Number fields. The first byte contains the
Reserved and Packet Number Length bits. It is used to carry "early"
data from the client to the server as part of the first flight, prior
to handshake completion. As part of the TLS handshake, the server
can accept or reject this early data.
See Section 2.3 of [TLS13] for a discussion of 0-RTT data and its
limitations.
0-RTT Packet {
Header Form (1) = 1,
Fixed Bit (1) = 1,
Long Packet Type (2) = 1,
Reserved Bits (2),
Packet Number Length (2),
Version (32),
DCID Length (8),
Destination Connection ID (0..160),
SCID Length (8),
Source Connection ID (0..160),
Length (i),
Packet Number (8..32),
Packet Payload (..),
}
Figure 15: 0-RTT Packet
Packet numbers for 0-RTT protected packets use the same space as
1-RTT protected packets.
After a client receives a Retry packet, 0-RTT packets are likely to
have been lost or discarded by the server. A client SHOULD attempt
to resend data in 0-RTT packets after it sends a new Initial packet.
A client MUST NOT reset the packet number it uses for 0-RTT packets,
since the keys used to protect 0-RTT packets will not change as a
result of responding to a Retry packet. Sending packets with the
same packet number in that case is likely to compromise the packet
protection for all 0-RTT packets because the same key and nonce could
be used to protect different content.
A client only receives acknowledgments for its 0-RTT packets once the
handshake is complete. Consequently, a server might expect 0-RTT
packets to start with a packet number of 0. Therefore, in
determining the length of the packet number encoding for 0-RTT
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packets, a client MUST assume that all packets up to the current
packet number are in flight, starting from a packet number of 0.
Thus, 0-RTT packets could need to use a longer packet number
encoding.
A client MUST NOT send 0-RTT packets once it starts processing 1-RTT
packets from the server. This means that 0-RTT packets cannot
contain any response to frames from 1-RTT packets. For instance, a
client cannot send an ACK frame in a 0-RTT packet, because that can
only acknowledge a 1-RTT packet. An acknowledgment for a 1-RTT
packet MUST be carried in a 1-RTT packet.
A server SHOULD treat a violation of remembered limits as a
connection error of an appropriate type (for instance, a
FLOW_CONTROL_ERROR for exceeding stream data limits).
17.2.4. Handshake Packet
A Handshake packet uses long headers with a type value of 0x2,
followed by the Length and Packet Number fields. The first byte
contains the Reserved and Packet Number Length bits. It is used to
carry acknowledgments and cryptographic handshake messages from the
server and client.
Handshake Packet {
Header Form (1) = 1,
Fixed Bit (1) = 1,
Long Packet Type (2) = 2,
Reserved Bits (2),
Packet Number Length (2),
Version (32),
DCID Length (8),
Destination Connection ID (0..160),
SCID Length (8),
Source Connection ID (0..160),
Length (i),
Packet Number (8..32),
Packet Payload (..),
}
Figure 16: Handshake Protected Packet
Once a client has received a Handshake packet from a server, it uses
Handshake packets to send subsequent cryptographic handshake messages
and acknowledgments to the server.
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The Destination Connection ID field in a Handshake packet contains a
connection ID that is chosen by the recipient of the packet; the
Source Connection ID includes the connection ID that the sender of
the packet wishes to use; see Section 7.2.
Handshake packets are their own packet number space, and thus the
first Handshake packet sent by a server contains a packet number of
0.
The payload of this packet contains CRYPTO frames and could contain
PING, PADDING, or ACK frames. Handshake packets MAY contain
CONNECTION_CLOSE frames. Endpoints MUST treat receipt of Handshake
packets with other frames as a connection error.
Like Initial packets (see Section 17.2.2.1), data in CRYPTO frames
for Handshake packets is discarded - and no longer retransmitted -
when Handshake protection keys are discarded.
17.2.5. Retry Packet
A Retry packet uses a long packet header with a type value of 0x3.
It carries an address validation token created by the server. It is
used by a server that wishes to perform a retry; see Section 8.1.
Retry Packet {
Header Form (1) = 1,
Fixed Bit (1) = 1,
Long Packet Type (2) = 3,
Unused (4),
Version (32),
DCID Length (8),
Destination Connection ID (0..160),
SCID Length (8),
Source Connection ID (0..160),
Retry Token (..),
Retry Integrity Tag (128),
}
Figure 17: Retry Packet
A Retry packet (shown in Figure 17) does not contain any protected
fields. The value in the Unused field is selected randomly by the
server. In addition to the fields from the long header, it contains
these additional fields:
Retry Token: An opaque token that the server can use to validate the
client's address.
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Retry Integrity Tag: See the Retry Packet Integrity section of
[QUIC-TLS].
17.2.5.1. Sending a Retry Packet
The server populates the Destination Connection ID with the
connection ID that the client included in the Source Connection ID of
the Initial packet.
The server includes a connection ID of its choice in the Source
Connection ID field. This value MUST not be equal to the Destination
Connection ID field of the packet sent by the client. A client MUST
discard a Retry packet that contains a Source Connection ID field
that is identical to the Destination Connection ID field of its
Initial packet. The client MUST use the value from the Source
Connection ID field of the Retry packet in the Destination Connection
ID field of subsequent packets that it sends.
A server MAY send Retry packets in response to Initial and 0-RTT
packets. A server can either discard or buffer 0-RTT packets that it
receives. A server can send multiple Retry packets as it receives
Initial or 0-RTT packets. A server MUST NOT send more than one Retry
packet in response to a single UDP datagram.
17.2.5.2. Handling a Retry Packet
A client MUST accept and process at most one Retry packet for each
connection attempt. After the client has received and processed an
Initial or Retry packet from the server, it MUST discard any
subsequent Retry packets that it receives.
Clients MUST discard Retry packets that have a Retry Integrity Tag
that cannot be validated, see the Retry Packet Integrity section of
[QUIC-TLS]. This diminishes an off-path attacker's ability to inject
a Retry packet and protects against accidental corruption of Retry
packets. A client MUST discard a Retry packet with a zero-length
Retry Token field.
The client responds to a Retry packet with an Initial packet that
includes the provided Retry Token to continue connection
establishment.
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A client sets the Destination Connection ID field of this Initial
packet to the value from the Source Connection ID in the Retry
packet. Changing Destination Connection ID also results in a change
to the keys used to protect the Initial packet. It also sets the
Token field to the token provided in the Retry. The client MUST NOT
change the Source Connection ID because the server could include the
connection ID as part of its token validation logic; see
Section 8.1.4.
A Retry packet does not include a packet number and cannot be
explicitly acknowledged by a client.
17.2.5.3. Continuing a Handshake After Retry
The next Initial packet from the client uses the connection ID and
token values from the Retry packet; see Section 7.2. Aside from
this, the Initial packet sent by the client is subject to the same
restrictions as the first Initial packet. A client MUST use the same
cryptographic handshake message it includes in this packet. A server
MAY treat a packet that contains a different cryptographic handshake
message as a connection error or discard it.
A client MAY attempt 0-RTT after receiving a Retry packet by sending
0-RTT packets to the connection ID provided by the server. A client
MUST NOT change the cryptographic handshake message it sends in
response to receiving a Retry.
A client MUST NOT reset the packet number for any packet number space
after processing a Retry packet; Section 17.2.3 contains more
information on this.
A server acknowledges the use of a Retry packet for a connection
using the retry_source_connection_id transport parameter; see
Section 18.2. If the server sends a Retry packet, it also
subsequently includes the value of the Source Connection ID field
from the Retry packet in its retry_source_connection_id transport
parameter.
If the client received and processed a Retry packet, it MUST validate
that the retry_source_connection_id transport parameter is present
and correct; otherwise, it MUST validate that the transport parameter
is absent. A client MUST treat a failed validation as a connection
error of type PROTOCOL_VIOLATION.
17.3. Short Header Packets
This version of QUIC defines a single packet type which uses the
short packet header.
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Short Header Packet {
Header Form (1) = 0,
Fixed Bit (1) = 1,
Spin Bit (1),
Reserved Bits (2),
Key Phase (1),
Packet Number Length (2),
Destination Connection ID (0..160),
Packet Number (8..32),
Packet Payload (..),
}
Figure 18: Short Header Packet Format
The short header can be used after the version and 1-RTT keys are
negotiated. Packets that use the short header contain the following
fields:
Header Form: The most significant bit (0x80) of byte 0 is set to 0
for the short header.
Fixed Bit: The next bit (0x40) of byte 0 is set to 1. Packets
containing a zero value for this bit are not valid packets in this
version and MUST be discarded.
Spin Bit: The third most significant bit (0x20) of byte 0 is the
latency spin bit, set as described in Section 17.3.1.
Reserved Bits: The next two bits (those with a mask of 0x18) of byte
0 are reserved. These bits are protected using header protection;
see Section 5.4 of [QUIC-TLS]. The value included prior to
protection MUST be set to 0. An endpoint MUST treat receipt of a
packet that has a non-zero value for these bits, after removing
both packet and header protection, as a connection error of type
PROTOCOL_VIOLATION. Discarding such a packet after only removing
header protection can expose the endpoint to attacks; see
Section 9.3 of [QUIC-TLS].
Key Phase: The next bit (0x04) of byte 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. This bit is protected using header
protection; see Section 5.4 of [QUIC-TLS].
Packet Number Length: The least significant two bits (those with a
mask of 0x03) of byte 0 contain the length of the packet number,
encoded as an unsigned, two-bit integer that is one less than the
length of the packet number field in bytes. That is, the length
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of the packet number field is the value of this field, plus one.
These bits are protected using header protection; see Section 5.4
of [QUIC-TLS].
Destination Connection ID: The Destination Connection ID is a
connection ID that is chosen by the intended recipient of the
packet. See Section 5.1 for more details.
Packet Number: The packet number field is 1 to 4 bytes long. The
packet number has confidentiality protection separate from packet
protection, as described in Section 5.4 of [QUIC-TLS]. The length
of the packet number field is encoded in Packet Number Length
field. See Section 17.1 for details.
Packet Payload: Packets with a short header always include a 1-RTT
protected payload.
The header form bit and the connection ID field of a short header
packet are version-independent. The remaining fields are specific to
the selected QUIC version. See [QUIC-INVARIANTS] for details on how
packets from different versions of QUIC are interpreted.
17.3.1. Latency Spin Bit
The latency spin bit enables passive latency monitoring from
observation points on the network path throughout the duration of a
connection. The spin bit is only present in the short packet header,
since it is possible to measure the initial RTT of a connection by
observing the handshake. Therefore, the spin bit is available after
version negotiation and connection establishment are completed. On-
path measurement and use of the latency spin bit is further discussed
in [QUIC-MANAGEABILITY].
The spin bit is an OPTIONAL feature of QUIC. A QUIC stack that
chooses to support the spin bit MUST implement it as specified in
this section.
Each endpoint unilaterally decides if the spin bit is enabled or
disabled for a connection. Implementations MUST allow administrators
of clients and servers to disable the spin bit either globally or on
a per-connection basis. Even when the spin bit is not disabled by
the administrator, endpoints MUST disable their use of the spin bit
for a random selection of at least one in every 16 network paths, or
for one in every 16 connection IDs. As each endpoint disables the
spin bit independently, this ensures that the spin bit signal is
disabled on approximately one in eight network paths.
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When the spin bit is disabled, endpoints MAY set the spin bit to any
value, and MUST ignore any incoming value. It is RECOMMENDED that
endpoints set the spin bit to a random value either chosen
independently for each packet or chosen independently for each
connection ID.
If the spin bit is enabled for the connection, the endpoint maintains
a spin value and sets the spin bit in the short header to the
currently stored value when a packet with a short header is sent out.
The spin value is initialized to 0 in the endpoint at connection
start. Each endpoint also remembers the highest packet number seen
from its peer on the connection.
When a server receives a short header packet that increments the
highest packet number seen by the server from the client, it sets the
spin value to be equal to the spin bit in the received packet.
When a client receives a short header packet that increments the
highest packet number seen by the client from the server, it sets the
spin value to the inverse of the spin bit in the received packet.
An endpoint resets its spin value to zero when sending the first
packet of a given connection with a new connection ID. This reduces
the risk that transient spin bit state can be used to link flows
across connection migration or ID change.
With this mechanism, the server reflects the spin value received,
while the client 'spins' it after one RTT. On-path observers can
measure the time between two spin bit toggle events to estimate the
end-to-end RTT of a connection.
18. Transport Parameter Encoding
The extension_data field of the quic_transport_parameters extension
defined in [QUIC-TLS] contains the QUIC transport parameters. They
are encoded as a sequence of transport parameters, as shown in
Figure 19:
Transport Parameters {
Transport Parameter (..) ...,
}
Figure 19: Sequence of Transport Parameters
Each transport parameter is encoded as an (identifier, length, value)
tuple, as shown in Figure 20:
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Transport Parameter {
Transport Parameter ID (i),
Transport Parameter Length (i),
Transport Parameter Value (..),
}
Figure 20: Transport Parameter Encoding
The Transport Parameter Length field contains the length of the
Transport Parameter Value field.
QUIC encodes transport parameters into a sequence of bytes, which are
then included in the cryptographic handshake.
18.1. Reserved Transport Parameters
Transport parameters with an identifier of the form "31 * N + 27" for
integer values of N are reserved to exercise the requirement that
unknown transport parameters be ignored. These transport parameters
have no semantics, and may carry arbitrary values.
18.2. Transport Parameter Definitions
This section details the transport parameters defined in this
document.
Many transport parameters listed here have integer values. Those
transport parameters that are identified as integers use a variable-
length integer encoding; see Section 16. Transport parameters have a
default value of 0 if the transport parameter is absent unless
otherwise stated.
The following transport parameters are defined:
original_destination_connection_id (0x00): The value of the
Destination Connection ID field from the first Initial packet sent
by the client; see Section 7.3. This transport parameter is only
sent by a server.
max_idle_timeout (0x01): The max idle timeout is a value in
milliseconds that is encoded as an integer; see (Section 10.2).
Idle timeout is disabled when both endpoints omit this transport
parameter or specify a value of 0.
stateless_reset_token (0x02): A stateless reset token is used in
verifying a stateless reset; see Section 10.4. This parameter is
a sequence of 16 bytes. This transport parameter MUST NOT be sent
by a client, but MAY be sent by a server. A server that does not
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send this transport parameter cannot use stateless reset
(Section 10.4) for the connection ID negotiated during the
handshake.
max_udp_payload_size (0x03): The maximum UDP payload size parameter
is an integer value that limits the size of UDP payloads that the
endpoint is willing to receive. UDP packets with payloads larger
than this limit are not likely to be processed by the receiver.
The default for this parameter is the maximum permitted UDP
payload of 65527. Values below 1200 are invalid.
This limit does act as an additional constraint on datagram size
in the same way as the path MTU, but it is a property of the
endpoint and not the path. It is expected that this is the space
an endpoint dedicates to holding incoming packets.
initial_max_data (0x04): The initial maximum data parameter is an
integer value that contains the initial value for the maximum
amount of data that can be sent on the connection. This is
equivalent to sending a MAX_DATA (Section 19.9) for the connection
immediately after completing the handshake.
initial_max_stream_data_bidi_local (0x05): This parameter is an
integer value specifying the initial flow control limit for
locally-initiated bidirectional streams. This limit applies to
newly created bidirectional streams opened by the endpoint that
sends the transport parameter. In client transport parameters,
this applies to streams with an identifier with the least
significant two bits set to 0x0; in server transport parameters,
this applies to streams with the least significant two bits set to
0x1.
initial_max_stream_data_bidi_remote (0x06): This parameter is an
integer value specifying the initial flow control limit for peer-
initiated bidirectional streams. This limit applies to newly
created bidirectional streams opened by the endpoint that receives
the transport parameter. In client transport parameters, this
applies to streams with an identifier with the least significant
two bits set to 0x1; in server transport parameters, this applies
to streams with the least significant two bits set to 0x0.
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initial_max_stream_data_uni (0x07): This parameter is an integer
value specifying the initial flow control limit for unidirectional
streams. This limit applies to newly created unidirectional
streams opened by the endpoint that receives the transport
parameter. In client transport parameters, this applies to
streams with an identifier with the least significant two bits set
to 0x3; in server transport parameters, this applies to streams
with the least significant two bits set to 0x2.
initial_max_streams_bidi (0x08): The initial maximum bidirectional
streams parameter is an integer value that contains the initial
maximum number of bidirectional streams the peer may initiate. If
this parameter is absent or zero, the peer cannot open
bidirectional streams until a MAX_STREAMS frame is sent. Setting
this parameter is equivalent to sending a MAX_STREAMS
(Section 19.11) of the corresponding type with the same value.
initial_max_streams_uni (0x09): The initial maximum unidirectional
streams parameter is an integer value that contains the initial
maximum number of unidirectional streams the peer may initiate.
If this parameter is absent or zero, the peer cannot open
unidirectional streams until a MAX_STREAMS frame is sent. Setting
this parameter is equivalent to sending a MAX_STREAMS
(Section 19.11) of the corresponding type with the same value.
ack_delay_exponent (0x0a): The ACK delay exponent is an integer
value indicating an exponent used to decode the ACK Delay field in
the ACK frame (Section 19.3). If this value is absent, a default
value of 3 is assumed (indicating a multiplier of 8). Values
above 20 are invalid.
max_ack_delay (0x0b): The maximum ACK delay is an integer value
indicating the maximum amount of time in milliseconds by which the
endpoint will delay sending acknowledgments. This value SHOULD
include the receiver's expected delays in alarms firing. For
example, if a receiver sets a timer for 5ms and alarms commonly
fire up to 1ms late, then it should send a max_ack_delay of 6ms.
If this value is absent, a default of 25 milliseconds is assumed.
Values of 2^14 or greater are invalid.
disable_active_migration (0x0c): The disable active migration
transport parameter is included if the endpoint does not support
active connection migration (Section 9). Peers of an endpoint
that sets this transport parameter MUST NOT send any packets,
including probing packets (Section 9.1), from a local address or
port other than that used to perform the handshake. This
parameter is a zero-length value.
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preferred_address (0x0d): The server's preferred address is used to
effect a change in server address at the end of the handshake, as
described in Section 9.6. The format of this transport parameter
is shown in Figure 21. This transport parameter is only sent by a
server. Servers MAY choose to only send a preferred address of
one address family by sending an all-zero address and port
(0.0.0.0:0 or ::.0) for the other family. IP addresses are
encoded in network byte order.
The Connection ID field and the Stateless Reset Token field
contain an alternative connection ID that has a sequence number of
1; see Section 5.1.1. Having these values bundled with the
preferred address ensures that there will be at least one unused
active connection ID when the client initiates migration to the
preferred address.
The Connection ID and Stateless Reset Token fields of a preferred
address are identical in syntax and semantics to the corresponding
fields of a NEW_CONNECTION_ID frame (Section 19.15). A server
that chooses a zero-length connection ID MUST NOT provide a
preferred address. Similarly, a server MUST NOT include a zero-
length connection ID in this transport parameter. A client MUST
treat violation of these requirements as a connection error of
type TRANSPORT_PARAMETER_ERROR.
Preferred Address {
IPv4 Address (32),
IPv4 Port (16),
IPv6 Address (128),
IPv6 Port (16),
CID Length (8),
Connection ID (..),
Stateless Reset Token (128),
}
Figure 21: Preferred Address format
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active_connection_id_limit (0x0e): The active connection ID limit is
an integer value specifying the maximum number of connection IDs
from the peer that an endpoint is willing to store. This value
includes the connection ID received during the handshake, that
received in the preferred_address transport parameter, and those
received in NEW_CONNECTION_ID frames. The value of the
active_connection_id_limit parameter MUST be at least 2. An
endpoint that receives a value less than 2 MUST close the
connection with an error of type TRANSPORT_PARAMETER_ERROR. If
this transport parameter is absent, a default of 2 is assumed. If
an endpoint issues a zero-length connection ID, it will never send
a NEW_CONNECTION_ID frame and therefore ignores the
active_connection_id_limit value received from its peer.
initial_source_connection_id (0x0f): The value that the endpoint
included in the Source Connection ID field of the first Initial
packet it sends for the connection; see Section 7.3.
retry_source_connection_id (0x10): The value that the the server
included in the Source Connection ID field of a Retry packet; see
Section 7.3. This transport parameter is only sent by a server.
If present, transport parameters that set initial flow control limits
(initial_max_stream_data_bidi_local,
initial_max_stream_data_bidi_remote, and initial_max_stream_data_uni)
are equivalent to sending a MAX_STREAM_DATA frame (Section 19.10) on
every stream of the corresponding type immediately after opening. If
the transport parameter is absent, streams of that type start with a
flow control limit of 0.
A client MUST NOT include any server-only transport parameter:
original_destination_connection_id, preferred_address,
retry_source_connection_id, or stateless_reset_token. A server MUST
treat receipt of any of these transport parameters as a connection
error of type TRANSPORT_PARAMETER_ERROR.
19. Frame Types and Formats
As described in Section 12.4, packets contain one or more frames.
This section describes the format and semantics of the core QUIC
frame types.
19.1. 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 byte that identifies the frame as a PADDING frame.
19.2. PING Frame
Endpoints can use PING frames (type=0x01) to verify that their peers
are still alive or to check reachability to the peer. The PING frame
contains no additional fields.
The receiver of a PING frame simply needs to acknowledge the packet
containing this frame.
The 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.
A connection will time out if no packets are sent or received for a
period longer than the time negotiated using the max_idle_timeout
transport parameter; see Section 10. 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.
19.3. ACK Frames
Receivers send ACK frames (types 0x02 and 0x03) to inform senders of
packets they have received and processed. The ACK frame contains one
or more ACK Ranges. ACK Ranges identify acknowledged packets. If
the frame type is 0x03, ACK frames also contain the sum of QUIC
packets with associated ECN marks received on the connection up until
this point. QUIC implementations MUST properly handle both types
and, if they have enabled ECN for packets they send, they SHOULD use
the information in the ECN section to manage their congestion state.
QUIC acknowledgements are irrevocable. Once acknowledged, a packet
remains acknowledged, even if it does not appear in a future ACK
frame. This is unlike TCP SACKs ([RFC2018]).
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Packets from different packet number spaces can be identified using
the same numeric value. An acknowledgment for a packet needs to
indicate both a packet number and a packet number space. This is
accomplished by having each ACK frame only acknowledge packet numbers
in the same space as the packet in which the ACK frame is contained.
Version Negotiation and Retry packets cannot be acknowledged because
they do not contain a packet number. Rather than relying on ACK
frames, these packets are implicitly acknowledged by the next Initial
packet sent by the client.
An ACK frame is shown in Figure 22.
ACK Frame {
Type (i) = 0x02..0x03,
Largest Acknowledged (i),
ACK Delay (i),
ACK Range Count (i),
First ACK Range (i),
ACK Range (..) ...,
[ECN Counts (..)],
}
Figure 22: ACK Frame Format
ACK frames contain the following fields:
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. Unlike the packet number in the QUIC
long or short header, the value in an ACK frame is not truncated.
ACK Delay: A variable-length integer representing the time delta in
microseconds between when this ACK was sent and when the largest
acknowledged packet, as indicated in the Largest Acknowledged
field, was received by this peer. The value of the ACK Delay
field is scaled by multiplying the encoded value by 2 to the power
of the value of the ack_delay_exponent transport parameter set by
the sender of the ACK frame; see Section 18.2. Scaling in this
fashion allows for a larger range of values with a shorter
encoding at the cost of lower resolution. Because the receiver
doesn't use the ACK Delay for Initial and Handshake packets, a
sender SHOULD send a value of 0.
ACK Range Count: A variable-length integer specifying the number of
Gap and ACK Range fields in the frame.
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First ACK Range: A variable-length integer indicating the number of
contiguous packets preceding the Largest Acknowledged that are
being acknowledged. The First ACK Range is encoded as an ACK
Range; see Section 19.3.1 starting from the Largest Acknowledged.
That is, the smallest packet acknowledged in the range is
determined by subtracting the First ACK Range value from the
Largest Acknowledged.
ACK Ranges: Contains additional ranges of packets which are
alternately not acknowledged (Gap) and acknowledged (ACK Range);
see Section 19.3.1.
ECN Counts: The three ECN Counts; see Section 19.3.2.
19.3.1. ACK Ranges
Each ACK Range consists of alternating Gap and ACK Range values in
descending packet number order. ACK Ranges can be repeated. The
number of Gap and ACK Range values is determined by the ACK Range
Count field; one of each value is present for each value in the ACK
Range Count field.
ACK Ranges are structured as shown in Figure 23.
ACK Range {
Gap (i),
ACK Range Length (i),
}
Figure 23: ACK Ranges
The fields that form each ACK Range are:
Gap: A variable-length integer indicating the number of contiguous
unacknowledged packets preceding the packet number one lower than
the smallest in the preceding ACK Range.
ACK Range Length: A variable-length integer indicating the number of
contiguous acknowledged packets preceding the largest packet
number, as determined by the preceding Gap.
Gap and ACK Range value use a relative integer encoding for
efficiency. Though each encoded value is positive, the values are
subtracted, so that each ACK Range describes progressively lower-
numbered packets.
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Each ACK Range acknowledges a contiguous range of packets by
indicating the number of acknowledged packets that precede the
largest packet number in that range. A value of zero indicates that
only the largest packet number is acknowledged. Larger ACK Range
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 range, the smallest value is determined by the
formula:
smallest = largest - ack_range
An ACK Range acknowledges all packets between the smallest packet
number and the largest, inclusive.
The largest value for an ACK Range is determined by cumulatively
subtracting the size of all preceding ACK Ranges 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 subsequent ACK Range using the following formula:
largest = previous_smallest - gap - 2
If any computed packet number is negative, an endpoint MUST generate
a connection error of type FRAME_ENCODING_ERROR.
19.3.2. ECN Counts
The ACK frame uses the least significant bit (that is, type 0x03) to
indicate ECN feedback and report receipt of QUIC packets with
associated ECN codepoints of ECT(0), ECT(1), or CE in the packet's IP
header. ECN Counts are only present when the ACK frame type is 0x03.
ECN Counts are only parsed when the ACK frame type is 0x03. There
are 3 ECN counts, as shown in Figure 24.
ECN Counts {
ECT0 Count (i),
ECT1 Count (i),
ECN-CE Count (i),
}
Figure 24: ECN Count Format
The three ECN Counts are:
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ECT0 Count: A variable-length integer representing the total number
of packets received with the ECT(0) codepoint in the packet number
space of the ACK frame.
ECT1 Count: A variable-length integer representing the total number
of packets received with the ECT(1) codepoint in the packet number
space of the ACK frame.
CE Count: A variable-length integer representing the total number of
packets received with the CE codepoint in the packet number space
of the ACK frame.
ECN counts are maintained separately for each packet number space.
19.4. RESET_STREAM Frame
An endpoint uses a RESET_STREAM frame (type=0x04) to abruptly
terminate the sending part of a stream.
After sending a RESET_STREAM, an endpoint ceases transmission and
retransmission of STREAM frames on the identified stream. A receiver
of RESET_STREAM can discard any data that it already received on that
stream.
An endpoint that receives a RESET_STREAM frame for a send-only stream
MUST terminate the connection with error STREAM_STATE_ERROR.
The RESET_STREAM frame is shown in Figure 25.
RESET_STREAM Frame {
Type (i) = 0x04,
Stream ID (i),
Application Protocol Error Code (i),
Final Size (i),
}
Figure 25: RESET_STREAM Frame Format
RESET_STREAM frames contain the following fields:
Stream ID: A variable-length integer encoding of the Stream ID of
the stream being terminated.
Application Protocol Error Code: A variable-length integer
containing the application protocol error code (see Section 20.1)
which indicates why the stream is being closed.
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Final Size: A variable-length integer indicating the final size of
the stream by the RESET_STREAM sender, in unit of bytes.
19.5. STOP_SENDING Frame
An endpoint uses a STOP_SENDING frame (type=0x05) to communicate that
incoming data is being discarded on receipt at application request.
STOP_SENDING requests that a peer cease transmission on a stream.
A STOP_SENDING frame can be sent for streams in the Recv or Size
Known states; see Section 3.1. Receiving a STOP_SENDING frame for a
locally-initiated stream that has not yet been created MUST be
treated as a connection error of type STREAM_STATE_ERROR. An
endpoint that receives a STOP_SENDING frame for a receive-only stream
MUST terminate the connection with error STREAM_STATE_ERROR.
The STOP_SENDING frame is shown in Figure 26.
STOP_SENDING Frame {
Type (i) = 0x05,
Stream ID (i),
Application Protocol Error Code (i),
}
Figure 26: STOP_SENDING Frame Format
STOP_SENDING frames contain the following fields:
Stream ID: A variable-length integer carrying the Stream ID of the
stream being ignored.
Application Protocol Error Code: A variable-length integer
containing the application-specified reason the sender is ignoring
the stream; see Section 20.1.
19.6. CRYPTO Frame
The CRYPTO frame (type=0x06) is used to transmit cryptographic
handshake messages. It can be sent in all packet types except 0-RTT.
The CRYPTO frame offers the cryptographic protocol an in-order stream
of bytes. CRYPTO frames are functionally identical to STREAM frames,
except that they do not bear a stream identifier; they are not flow
controlled; and they do not carry markers for optional offset,
optional length, and the end of the stream.
The CRYPTO frame is shown in Figure 27.
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CRYPTO Frame {
Type (i) = 0x06,
Offset (i),
Length (i),
Crypto Data (..),
}
Figure 27: CRYPTO Frame Format
CRYPTO frames contain the following fields:
Offset: A variable-length integer specifying the byte offset in the
stream for the data in this CRYPTO frame.
Length: A variable-length integer specifying the length of the
Crypto Data field in this CRYPTO frame.
Crypto Data: The cryptographic message data.
There is a separate flow of cryptographic handshake data in each
encryption level, each of which starts at an offset of 0. This
implies that each encryption level is treated as a separate CRYPTO
stream of data.
The largest offset delivered on a stream - the sum of the offset and
data length - cannot exceed 2^62-1. Receipt of a frame that exceeds
this limit MUST be treated as a connection error of type
FRAME_ENCODING_ERROR or CRYPTO_BUFFER_EXCEEDED.
Unlike STREAM frames, which include a Stream ID indicating to which
stream the data belongs, the CRYPTO frame carries data for a single
stream per encryption level. The stream does not have an explicit
end, so CRYPTO frames do not have a FIN bit.
19.7. NEW_TOKEN Frame
A server sends a NEW_TOKEN frame (type=0x07) to provide the client
with a token to send in the header of an Initial packet for a future
connection.
The NEW_TOKEN frame is shown in Figure 28.
NEW_TOKEN Frame {
Type (i) = 0x07,
Token Length (i),
Token (..),
}
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Figure 28: NEW_TOKEN Frame Format
NEW_TOKEN frames contain the following fields:
Token Length: A variable-length integer specifying the length of the
token in bytes.
Token: An opaque blob that the client may use with a future Initial
packet. The token MUST NOT be empty. An endpoint MUST treat
receipt of a NEW_TOKEN frame with an empty Token field as a
connection error of type FRAME_ENCODING_ERROR.
An endpoint might receive multiple NEW_TOKEN frames that contain the
same token value if packets containing the frame are incorrectly
determined to be lost. Endpoints are responsible for discarding
duplicate values, which might be used to link connection attempts;
see Section 8.1.3.
Clients MUST NOT send NEW_TOKEN frames. Servers MUST treat receipt
of a NEW_TOKEN frame as a connection error of type
PROTOCOL_VIOLATION.
19.8. STREAM Frames
STREAM frames implicitly create a stream and carry stream data. The
STREAM frame takes the form 0b00001XXX (or the set of values from
0x08 to 0x0f). The value of the three low-order bits of the frame
type determines the fields that are present in the frame.
* 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 bytes of the stream, or the end of a stream that includes no
data).
* 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.
* The FIN bit (0x01) of the frame type is set only on frames that
contain the final size of the stream. Setting this bit indicates
that the frame marks the end of the stream.
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An endpoint MUST terminate the connection with error
STREAM_STATE_ERROR if it receives a STREAM frame for a locally-
initiated stream that has not yet been created, or for a send-only
stream.
The STREAM frames are shown in Figure 29.
STREAM Frame {
Type (i) = 0x08..0x0f,
Stream ID (i),
[Offset (i)],
[Length (i)],
Stream Data (..),
}
Figure 29: STREAM Frame Format
STREAM frames contain the following fields:
Stream ID: A variable-length integer indicating the stream ID of the
stream; see Section 2.1.
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.
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 bytes in the packet.
Stream Data: The bytes from the designated stream to be delivered.
When a Stream Data field has a length of 0, 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 offset and data length -
cannot exceed 2^62-1, as it is not possible to provide flow control
credit for that data. Receipt of a frame that exceeds this limit
MUST be treated as a connection error of type FRAME_ENCODING_ERROR or
FLOW_CONTROL_ERROR.
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19.9. MAX_DATA Frame
The MAX_DATA frame (type=0x10) 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 MAX_DATA frame is shown in Figure 30.
MAX_DATA Frame {
Type (i) = 0x10,
Maximum Data (i),
}
Figure 30: MAX_DATA Frame Format
MAX_DATA frames contain the following fields:
Maximum Data: A variable-length integer indicating the maximum
amount of data that can be sent on the entire connection, in units
of bytes.
All data sent in STREAM frames counts toward this limit. The sum of
the largest received offsets on all streams - including streams in
terminal states - MUST NOT exceed the value advertised by a receiver.
An endpoint MUST terminate a connection with a FLOW_CONTROL_ERROR
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 7.4.1.
19.10. MAX_STREAM_DATA Frame
The MAX_STREAM_DATA frame (type=0x11) is used in flow control to
inform a peer of the maximum amount of data that can be sent on a
stream.
A MAX_STREAM_DATA frame can be sent for streams in the Recv state;
see Section 3.1. Receiving a MAX_STREAM_DATA frame for a locally-
initiated stream that has not yet been created MUST be treated as a
connection error of type STREAM_STATE_ERROR. An endpoint that
receives a MAX_STREAM_DATA frame for a receive-only stream MUST
terminate the connection with error STREAM_STATE_ERROR.
The MAX_STREAM_DATA frame is shown in Figure 31.
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MAX_STREAM_DATA Frame {
Type (i) = 0x11,
Stream ID (i),
Maximum Stream Data (i),
}
Figure 31: MAX_STREAM_DATA Frame Format
MAX_STREAM_DATA frames contain the following fields:
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 bytes.
When counting data toward this limit, an endpoint accounts for the
largest received offset of data that is sent or received on the
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 7.4.1.
19.11. MAX_STREAMS Frames
The MAX_STREAMS frames (type=0x12 and 0x13) inform the peer of the
cumulative number of streams of a given type it is permitted to open.
A MAX_STREAMS frame with a type of 0x12 applies to bidirectional
streams, and a MAX_STREAMS frame with a type of 0x13 applies to
unidirectional streams.
The MAX_STREAMS frames are shown in Figure 32;
MAX_STREAMS Frame {
Type (i) = 0x12..0x13,
Maximum Streams (i),
}
Figure 32: MAX_STREAMS Frame Format
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MAX_STREAMS frames contain the following fields:
Maximum Streams: A count of the cumulative number of streams of the
corresponding type that can be opened over the lifetime of the
connection. This value cannot exceed 2^60, as it is not possible
to encode stream IDs larger than 2^62-1. Receipt of a frame that
permits opening of a stream larger than this limit MUST be treated
as a FRAME_ENCODING_ERROR.
Loss or reordering can cause a MAX_STREAMS frame to be received which
states a lower stream limit than an endpoint has previously received.
MAX_STREAMS frames which do not increase the stream limit MUST be
ignored.
An endpoint MUST NOT open more streams than permitted by the current
stream limit set by its peer. For instance, a server that receives a
unidirectional stream limit of 3 is permitted to open stream 3, 7,
and 11, but not stream 15. An endpoint MUST terminate a connection
with a STREAM_LIMIT_ERROR error if a peer opens more streams than was
permitted.
Note that these frames (and the corresponding transport parameters)
do not describe the number of streams that can be opened
concurrently. The limit includes streams that have been closed as
well as those that are open.
19.12. DATA_BLOCKED Frame
A sender SHOULD send a DATA_BLOCKED frame (type=0x14) when it wishes
to send data, but is unable to due to connection-level flow control;
see Section 4. DATA_BLOCKED frames can be used as input to tuning of
flow control algorithms; see Section 4.2.
The DATA_BLOCKED frame is shown in Figure 33.
DATA_BLOCKED Frame {
Type (i) = 0x14,
Maximum Data (i),
}
Figure 33: DATA_BLOCKED Frame Format
DATA_BLOCKED frames contain the following fields:
Maximum Data: A variable-length integer indicating the connection-
level limit at which blocking occurred.
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19.13. STREAM_DATA_BLOCKED Frame
A sender SHOULD send a STREAM_DATA_BLOCKED frame (type=0x15) when it
wishes to send data, but is unable to due to stream-level flow
control. This frame is analogous to DATA_BLOCKED (Section 19.12).
An endpoint that receives a STREAM_DATA_BLOCKED frame for a send-only
stream MUST terminate the connection with error STREAM_STATE_ERROR.
The STREAM_DATA_BLOCKED frame is shown in Figure 34.
STREAM_DATA_BLOCKED Frame {
Type (i) = 0x15,
Stream ID (i),
Maximum Stream Data (i),
}
Figure 34: STREAM_DATA_BLOCKED Frame Format
STREAM_DATA_BLOCKED frames contain the following fields:
Stream ID: A variable-length integer indicating the stream which is
flow control blocked.
Maximum Stream Data: A variable-length integer indicating the offset
of the stream at which the blocking occurred.
19.14. STREAMS_BLOCKED Frames
A sender SHOULD send a STREAMS_BLOCKED frame (type=0x16 or 0x17) when
it wishes to open a stream, but is unable to due to the maximum
stream limit set by its peer; see Section 19.11. A STREAMS_BLOCKED
frame of type 0x16 is used to indicate reaching the bidirectional
stream limit, and a STREAMS_BLOCKED frame of type 0x17 indicates
reaching the unidirectional stream limit.
A STREAMS_BLOCKED frame does not open the stream, but informs the
peer that a new stream was needed and the stream limit prevented the
creation of the stream.
The STREAMS_BLOCKED frames are shown in Figure 35.
STREAMS_BLOCKED Frame {
Type (i) = 0x16..0x17,
Maximum Streams (i),
}
Figure 35: STREAMS_BLOCKED Frame Format
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STREAMS_BLOCKED frames contain the following fields:
Maximum Streams: A variable-length integer indicating the maximum
number of streams allowed at the time the frame was sent. This
value cannot exceed 2^60, as it is not possible to encode stream
IDs larger than 2^62-1. Receipt of a frame that encodes a larger
stream ID MUST be treated as a STREAM_LIMIT_ERROR or a
FRAME_ENCODING_ERROR.
19.15. NEW_CONNECTION_ID Frame
An endpoint sends a NEW_CONNECTION_ID frame (type=0x18) to provide
its peer with alternative connection IDs that can be used to break
linkability when migrating connections; see Section 9.5.
The NEW_CONNECTION_ID frame is shown in Figure 36.
NEW_CONNECTION_ID Frame {
Type (i) = 0x18,
Sequence Number (i),
Retire Prior To (i),
Length (8),
Connection ID (8..160),
Stateless Reset Token (128),
}
Figure 36: NEW_CONNECTION_ID Frame Format
NEW_CONNECTION_ID frames contain the following fields:
Sequence Number: The sequence number assigned to the connection ID
by the sender. See Section 5.1.1.
Retire Prior To: A variable-length integer indicating which
connection IDs should be retired; see Section 5.1.2.
Length: An 8-bit unsigned integer containing the length of the
connection ID. Values less than 1 and greater than 20 are invalid
and MUST be treated as a connection error of type
FRAME_ENCODING_ERROR.
Connection ID: A connection ID of the specified length.
Stateless Reset Token: A 128-bit value that will be used for a
stateless reset when the associated connection ID is used; see
Section 10.4.
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An endpoint MUST NOT send this frame if it currently requires that
its peer send packets with a zero-length Destination Connection ID.
Changing the length of a connection ID to or from zero-length makes
it difficult to identify when the value of the connection ID changed.
An endpoint that is sending packets with a zero-length Destination
Connection ID MUST treat receipt of a NEW_CONNECTION_ID frame as a
connection error of type PROTOCOL_VIOLATION.
Transmission errors, timeouts and retransmissions might cause the
same NEW_CONNECTION_ID frame to be received multiple times. Receipt
of the same frame multiple times MUST NOT be treated as a connection
error. A receiver can use the sequence number supplied in the
NEW_CONNECTION_ID frame to identify new connection IDs from old ones.
If an endpoint receives a NEW_CONNECTION_ID frame that repeats a
previously issued connection ID with a different Stateless Reset
Token or a different sequence number, or if a sequence number is used
for different connection IDs, the endpoint MAY treat that receipt as
a connection error of type PROTOCOL_VIOLATION.
The Retire Prior To field counts connection IDs established during
connection setup and the preferred_address transport parameter; see
Section 5.1.2. The Retire Prior To field MUST be less than or equal
to the Sequence Number field. Receiving a value greater than the
Sequence Number MUST be treated as a connection error of type
FRAME_ENCODING_ERROR.
Once a sender indicates a Retire Prior To value, smaller values sent
in subsequent NEW_CONNECTION_ID frames have no effect. A receiver
MUST ignore any Retire Prior To fields that do not increase the
largest received Retire Prior To value.
An endpoint that receives a NEW_CONNECTION_ID frame with a sequence
number smaller than the Retire Prior To field of a previously
received NEW_CONNECTION_ID frame MUST send a corresponding
RETIRE_CONNECTION_ID frame that retires the newly received connection
ID, unless it has already done so for that sequence number.
19.16. RETIRE_CONNECTION_ID Frame
An endpoint sends a RETIRE_CONNECTION_ID frame (type=0x19) to
indicate that it will no longer use a connection ID that was issued
by its peer. This may include the connection ID provided during the
handshake. Sending a RETIRE_CONNECTION_ID frame also serves as a
request to the peer to send additional connection IDs for future use;
see Section 5.1. New connection IDs can be delivered to a peer using
the NEW_CONNECTION_ID frame (Section 19.15).
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Retiring a connection ID invalidates the stateless reset token
associated with that connection ID.
The RETIRE_CONNECTION_ID frame is shown in Figure 37.
RETIRE_CONNECTION_ID Frame {
Type (i) = 0x19,
Sequence Number (i),
}
Figure 37: RETIRE_CONNECTION_ID Frame Format
RETIRE_CONNECTION_ID frames contain the following fields:
Sequence Number: The sequence number of the connection ID being
retired; see Section 5.1.2.
Receipt of a RETIRE_CONNECTION_ID frame containing a sequence number
greater than any previously sent to the peer MUST be treated as a
connection error of type PROTOCOL_VIOLATION.
The sequence number specified in a RETIRE_CONNECTION_ID frame MUST
NOT refer to the Destination Connection ID field of the packet in
which the frame is contained. The peer MAY treat this as a
connection error of type FRAME_ENCODING_ERROR.
An endpoint cannot send this frame if it was provided with a zero-
length connection ID by its peer. An endpoint that provides a zero-
length connection ID MUST treat receipt of a RETIRE_CONNECTION_ID
frame as a connection error of type PROTOCOL_VIOLATION.
19.17. PATH_CHALLENGE Frame
Endpoints can use PATH_CHALLENGE frames (type=0x1a) to check
reachability to the peer and for path validation during connection
migration.
The PATH_CHALLENGE frame is shown in Figure 38.
PATH_CHALLENGE Frame {
Type (i) = 0x1a,
Data (64),
}
Figure 38: PATH_CHALLENGE Frame Format
PATH_CHALLENGE frames contain the following fields:
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Data: This 8-byte field contains arbitrary data.
A PATH_CHALLENGE frame containing 8 bytes that are hard to guess is
sufficient to ensure that it is easier to receive the packet than it
is to guess the value correctly.
The recipient of this frame MUST generate a PATH_RESPONSE frame
(Section 19.18) containing the same Data.
19.18. PATH_RESPONSE Frame
The PATH_RESPONSE frame (type=0x1b) is sent in response to a
PATH_CHALLENGE frame. Its format, shown in Figure 39 is identical to
the PATH_CHALLENGE frame (Section 19.17).
PATH_RESPONSE Frame {
Type (i) = 0x1b,
Data (64),
}
Figure 39: PATH_RESPONSE Frame Format
If the content of a PATH_RESPONSE frame does not match the content of
a PATH_CHALLENGE frame previously sent by the endpoint, the endpoint
MAY generate a connection error of type PROTOCOL_VIOLATION.
19.19. CONNECTION_CLOSE Frames
An endpoint sends a CONNECTION_CLOSE frame (type=0x1c or 0x1d) to
notify its peer that the connection is being closed. The
CONNECTION_CLOSE with a frame type of 0x1c is used to signal errors
at only the QUIC layer, or the absence of errors (with the NO_ERROR
code). The CONNECTION_CLOSE frame with a type of 0x1d is used to
signal an error with the application that uses QUIC.
If there are open streams that haven't been explicitly closed, they
are implicitly closed when the connection is closed.
The CONNECTION_CLOSE frames are shown in Figure 40.
CONNECTION_CLOSE Frame {
Type (i) = 0x1c..0x1d,
Error Code (i),
[Frame Type (i)],
Reason Phrase Length (i),
Reason Phrase (..),
}
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Figure 40: CONNECTION_CLOSE Frame Format
CONNECTION_CLOSE frames contain the following fields:
Error Code: A variable length integer error code which indicates the
reason for closing this connection. A CONNECTION_CLOSE frame of
type 0x1c uses codes from the space defined in Section 20. A
CONNECTION_CLOSE frame of type 0x1d uses codes from the
application protocol error code space; see Section 20.1
Frame Type: A variable-length integer encoding the type of frame
that triggered the error. A value of 0 (equivalent to the mention
of the PADDING frame) is used when the frame type is unknown. The
application-specific variant of CONNECTION_CLOSE (type 0x1d) does
not include this field.
Reason Phrase Length: A variable-length integer specifying the
length of the reason phrase in bytes. Because a CONNECTION_CLOSE
frame cannot be split between packets, 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].
The application-specific variant of CONNECTION_CLOSE (type 0x1d) can
only be sent using 0-RTT or 1-RTT packets ([QUIC-TLS], Section 4).
When an application wishes to abandon a connection during the
handshake, an endpoint can send a CONNECTION_CLOSE frame (type 0x1c)
with an error code of APPLICATION_ERROR in an Initial or a Handshake
packet.
19.20. HANDSHAKE_DONE frame
The server uses the HANDSHAKE_DONE frame (type=0x1e) to signal
confirmation of the handshake to the client. The HANDSHAKE_DONE
frame contains no additional fields.
This frame can only be sent by the server. Servers MUST NOT send a
HANDSHAKE_DONE frame before completing the handshake. A server MUST
treat receipt of a HANDSHAKE_DONE frame as a connection error of type
PROTOCOL_VIOLATION.
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19.21. Extension Frames
QUIC frames do not use a self-describing encoding. An endpoint
therefore needs to understand the syntax of all frames before it can
successfully process a packet. This allows for efficient encoding of
frames, but it means that an endpoint cannot send a frame of a type
that is unknown to its peer.
An extension to QUIC that wishes to use a new type of frame MUST
first ensure that a peer is able to understand the frame. An
endpoint can use a transport parameter to signal its willingness to
receive one or more extension frame types with the one transport
parameter.
Extensions that modify or replace core protocol functionality
(including frame types) will be difficult to combine with other
extensions which modify or replace the same functionality unless the
behavior of the combination is explicitly defined. Such extensions
SHOULD define their interaction with previously-defined extensions
modifying the same protocol components.
Extension frames MUST be congestion controlled and MUST cause an ACK
frame to be sent. The exception is extension frames that replace or
supplement the ACK frame. Extension frames are not included in flow
control unless specified in the extension.
An IANA registry is used to manage the assignment of frame types; see
Section 22.3.
20. Transport Error Codes
QUIC error codes are 62-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.
INTERNAL_ERROR (0x1): The endpoint encountered an internal error and
cannot continue with the connection.
SERVER_BUSY (0x2): The server is currently busy and does not accept
any new connections.
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FLOW_CONTROL_ERROR (0x3): An endpoint received more data than it
permitted in its advertised data limits; see Section 4.
STREAM_LIMIT_ERROR (0x4): An endpoint received a frame for a stream
identifier that exceeded its advertised stream limit for the
corresponding stream type.
STREAM_STATE_ERROR (0x5): An endpoint received a frame for a stream
that was not in a state that permitted that frame; see Section 3.
FINAL_SIZE_ERROR (0x6): An endpoint received a STREAM frame
containing data that exceeded the previously established final
size. Or an endpoint received a STREAM frame or a RESET_STREAM
frame containing a final size that was lower than the size of
stream data that was already received. Or an endpoint received a
STREAM frame or a RESET_STREAM frame containing a different final
size to the one already established.
FRAME_ENCODING_ERROR (0x7): An endpoint received a frame that was
badly formatted. For instance, a frame of an unknown type, or an
ACK frame that has more acknowledgment ranges than the remainder
of the packet could carry.
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.
CONNECTION_ID_LIMIT_ERROR (0x9): The number of connection IDs
provided by the peer exceeds the advertised
active_connection_id_limit.
PROTOCOL_VIOLATION (0xA): An endpoint detected an error with
protocol compliance that was not covered by more specific error
codes.
INVALID_TOKEN (0xB): A server received a Retry Token in a client
Initial that is invalid.
APPLICATION_ERROR (0xC): The application or application protocol
caused the connection to be closed.
CRYPTO_BUFFER_EXCEEDED (0xD): An endpoint has received more data in
CRYPTO frames than it can buffer.
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CRYPTO_ERROR (0x1XX): The cryptographic handshake failed. A range
of 256 values is reserved for carrying error codes specific to the
cryptographic handshake that is used. Codes for errors occurring
when TLS is used for the crypto handshake are described in
Section 4.8 of [QUIC-TLS].
See Section 22.4 for details of registering new error codes.
In defining these error codes, several principles are applied. Error
conditions that might require specific action on the part of a
recipient are given unique codes. Errors that represent common
conditions are given specific codes. Absent either of these
conditions, error codes are used to identify a general function of
the stack, like flow control or transport parameter handling.
Finally, generic errors are provided for conditions where
implementations are unable or unwilling to use more specific codes.
20.1. Application Protocol Error Codes
Application protocol error codes are 62-bit unsigned integers, but
the management of application error codes is left to application
protocols. Application protocol error codes are used for the
RESET_STREAM frame (Section 19.4), the STOP_SENDING frame
(Section 19.5), and the CONNECTION_CLOSE frame with a type of 0x1d
(Section 19.19).
21. Security Considerations
21.1. Handshake Denial of Service
As an encrypted and authenticated transport QUIC provides a range of
protections against denial of service. Once the cryptographic
handshake is complete, QUIC endpoints discard most packets that are
not authenticated, greatly limiting the ability of an attacker to
interfere with existing connections.
Once a connection is established QUIC endpoints might accept some
unauthenticated ICMP packets (see Section 14.2), but the use of these
packets is extremely limited. The only other type of packet that an
endpoint might accept is a stateless reset (Section 10.4) which
relies on the token being kept secret until it is used.
During the creation of a connection, QUIC only provides protection
against attack from off the network path. All QUIC packets contain
proof that the recipient saw a preceding packet from its peer.
Addresses cannot change during the handshake, so endpoints can
discard packets that are received on a different network path.
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The Source and Destination Connection ID fields are the primary means
of protection against off-path attack during the handshake. These
are required to match those set by a peer. Except for an Initial and
stateless reset packets, an endpoint only accepts packets that
include a Destination Connection ID field that matches a value the
endpoint previously chose. This is the only protection offered for
Version Negotiation packets.
The Destination Connection ID field in an Initial packet is selected
by a client to be unpredictable, which serves an additional purpose.
The packets that carry the cryptographic handshake are protected with
a key that is derived from this connection ID and salt specific to
the QUIC version. This allows endpoints to use the same process for
authenticating packets that they receive as they use after the
cryptographic handshake completes. Packets that cannot be
authenticated are discarded. Protecting packets in this fashion
provides a strong assurance that the sender of the packet saw the
Initial packet and understood it.
These protections are not intended to be effective against an
attacker that is able to receive QUIC packets prior to the connection
being established. Such an attacker can potentially send packets
that will be accepted by QUIC endpoints. This version of QUIC
attempts to detect this sort of attack, but it expects that endpoints
will fail to establish a connection rather than recovering. For the
most part, the cryptographic handshake protocol [QUIC-TLS] is
responsible for detecting tampering during the handshake.
Endpoints are permitted to use other methods to detect and attempt to
recover from interference with the handshake. Invalid packets may be
identified and discarded using other methods, but no specific method
is mandated in this document.
21.2. Amplification Attack
An attacker might be able to receive an address validation token
(Section 8) from a server and then release the IP address it used to
acquire that token. At a later time, the attacker may initiate a
0-RTT connection with a server by spoofing this same address, which
might now address a different (victim) endpoint. The attacker can
thus potentially cause the server to send an initial congestion
window's worth of data towards the victim.
Servers SHOULD provide mitigations for this attack by limiting the
usage and lifetime of address validation tokens; see Section 8.1.3.
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21.3. Optimistic ACK Attack
An endpoint that acknowledges packets it has not received might cause
a congestion controller to permit sending at rates beyond what the
network supports. An endpoint MAY skip packet numbers when sending
packets to detect this behavior. An endpoint can then immediately
close the connection with a connection error of type
PROTOCOL_VIOLATION; see Section 10.3.
21.4. 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.
21.5. Stream Fragmentation and Reassembly Attacks
An adversarial sender might intentionally send fragments of stream
data in order to cause disproportionate receive buffer memory
commitment and/or creation of a large and inefficient data structure.
An adversarial receiver might intentionally not acknowledge packets
containing stream data in order to force the sender to store the
unacknowledged stream data for retransmission.
The attack on receivers 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.
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QUIC deployments SHOULD provide mitigations against stream
fragmentation attacks. Mitigations could consist of avoiding over-
committing memory, limiting the size of tracking data structures,
delaying reassembly of STREAM frames, implementing heuristics based
on the age and duration of reassembly holes, or some combination.
21.6. 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 2.1. However, when several streams are initiated at short
intervals, loss or reordering may cause STREAM frames that open
streams to be received out of sequence. On receiving a higher-
numbered stream ID, a receiver is required to open all intervening
streams of the same type; see Section 3.2. Thus, on a new
connection, opening stream 4000000 opens 1 million and 1 client-
initiated bidirectional streams.
The number of active streams is limited by the
initial_max_streams_bidi and initial_max_streams_uni transport
parameters, as explained in Section 4.5. If chosen judiciously,
these limits mitigate 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.
21.7. Peer Denial of Service
QUIC and TLS both contain messages that have legitimate uses in some
contexts, but that can be abused to cause a peer to expend processing
resources without having any observable impact on the state of the
connection.
Messages can also be used to change and revert state in small or
inconsequential ways, such as by sending small increments to flow
control limits.
If processing costs are disproportionately large in comparison to
bandwidth consumption or effect on state, then this could allow a
malicious peer to exhaust processing capacity.
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While there are legitimate uses for all messages, implementations
SHOULD track cost of processing relative to progress and treat
excessive quantities of any non-productive packets as indicative of
an attack. Endpoints MAY respond to this condition with a connection
error, or by dropping packets.
21.8. Explicit Congestion Notification Attacks
An on-path attacker could manipulate the value of ECN codepoints in
the IP header to influence the sender's rate. [RFC3168] discusses
manipulations and their effects in more detail.
An on-the-side attacker can duplicate and send packets with modified
ECN codepoints to affect the sender's rate. If duplicate packets are
discarded by a receiver, an off-path attacker will need to race the
duplicate packet against the original to be successful in this
attack. Therefore, QUIC endpoints ignore the ECN codepoint field on
an IP packet unless at least one QUIC packet in that IP packet is
successfully processed; see Section 13.4.
21.9. Stateless Reset Oracle
Stateless resets create a possible denial of service attack analogous
to a TCP reset injection. This attack is possible if an attacker is
able to cause a stateless reset token to be generated for a
connection with a selected connection ID. An attacker that can cause
this token to be generated can reset an active connection with the
same connection ID.
If a packet can be routed to different instances that share a static
key, for example by changing an IP address or port, then an attacker
can cause the server to send a stateless reset. To defend against
this style of denial service, endpoints that share a static key for
stateless reset (see Section 10.4.2) MUST be arranged so that packets
with a given connection ID always arrive at an instance that has
connection state, unless that connection is no longer active.
In the case of a cluster that uses dynamic load balancing, it's
possible that a change in load balancer configuration could happen
while an active instance retains connection state; even if an
instance retains connection state, the change in routing and
resulting stateless reset will result in the connection being
terminated. If there is no chance in the packet being routed to the
correct instance, it is better to send a stateless reset than wait
for connections to time out. However, this is acceptable only if the
routing cannot be influenced by an attacker.
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21.10. Version Downgrade
This document defines QUIC Version Negotiation packets in Section 6,
which can be used to negotiate the QUIC version used between two
endpoints. However, this document does not specify how this
negotiation will be performed between this version and subsequent
future versions. In particular, Version Negotiation packets do not
contain any mechanism to prevent version downgrade attacks. Future
versions of QUIC that use Version Negotiation packets MUST define a
mechanism that is robust against version downgrade attacks.
21.11. Targeted Attacks by Routing
Deployments should limit the ability of an attacker to target a new
connection to a particular server instance. This means that client-
controlled fields, such as the initial Destination Connection ID used
on Initial and 0-RTT packets SHOULD NOT be used by themselves to make
routing decisions. Ideally, routing decisions are made independently
of client-selected values; a Source Connection ID can be selected to
route later packets to the same server.
21.12. Overview of Security Properties
A complete security analysis of QUIC is outside the scope of this
document. This section provides an informal description of the
desired security properties as an aid to implementors and to help
guide protocol analysis.
QUIC assumes the threat model described in [SEC-CONS] and provides
protections against many of the attacks that arise from that model.
For this purpose, attacks are divided into passive and active
attacks. Passive attackers have the capability to read packets from
the network, while active attackers also have the capability to write
packets into the network. However, a passive attack may involve an
attacker with the ability to cause a routing change or other
modification in the path taken by packets that comprise a connection.
Attackers are additionally categorized as either on-path attackers or
off-path attackers; see Section 3.5 of [SEC-CONS]. An on-path
attacker can read, modify, or remove any packet it observes such that
it no longer reaches its destination, while an off-path attacker
observes the packets, but cannot prevent the original packet from
reaching its intended destination. An off-path attacker can also
transmit arbitrary packets.
Properties of the handshake, protected packets, and connection
migration are considered separately.
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21.12.1. Handshake
The QUIC handshake incorporates the TLS 1.3 handshake and inherits
the cryptographic properties described in Appendix E.1 of [TLS13].
Many of the security properties of QUIC depend on the TLS handshake
providing these properties. Any attack on the TLS handshake could
affect QUIC.
Any attack on the TLS handshake that compromises the secrecy or
uniqueness of session keys affects other security guarantees provided
by QUIC that depends on these keys. For instance, migration
(Section 9) depends on the efficacy of confidentiality protections,
both for the negotiation of keys using the TLS handshake and for QUIC
packet protection, to avoid linkability across network paths.
An attack on the integrity of the TLS handshake might allow an
attacker to affect the selection of application protocol or QUIC
version.
In addition to the properties provided by TLS, the QUIC handshake
provides some defense against DoS attacks on the handshake.
21.12.1.1. Anti-Amplification
Address validation (Section 8) is used to verify that an entity that
claims a given address is able to receive packets at that address.
Address validation limits amplification attack targets to addresses
for which an attacker is either on-path or off-path.
Prior to validation, endpoints are limited in what they are able to
send. During the handshake, a server cannot send more than three
times the data it receives; clients that initiate new connections or
migrate to a new network path are limited.
21.12.1.2. Server-Side DoS
Computing the server's first flight for a full handshake is
potentially expensive, requiring both a signature and a key exchange
computation. In order to prevent computational DoS attacks, the
Retry packet provides a cheap token exchange mechanism which allows
servers to validate a client's IP address prior to doing any
expensive computations at the cost of a single round trip. After a
successful handshake, servers can issue new tokens to a client which
will allow new connection establishment without incurring this cost.
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21.12.1.3. On-Path Handshake Termination
An on-path or off-path attacker can force a handshake to fail by
replacing or racing Initial packets. Once valid Initial packets have
been exchanged, subsequent Handshake packets are protected with the
handshake keys and an on-path attacker cannot force handshake failure
other than by dropping packets to cause endpoints to abandon the
attempt.
An on-path attacker can also replace the addresses of packets on
either side and therefore cause the client or server to have an
incorrect view of the remote addresses. Such an attack is
indistinguishable from the functions performed by a NAT.
21.12.1.4. Parameter Negotiation
The entire handshake is cryptographically protected, with the Initial
packets being encrypted with per-version keys and the Handshake and
later packets being encrypted with keys derived from the TLS key
exchange. Further, parameter negotiation is folded into the TLS
transcript and thus provides the same integrity guarantees as
ordinary TLS negotiation. An attacker can observe the client's
transport parameters (as long as it knows the version-specific keys)
but cannot observe the server's transport parameters and cannot
influence parameter negotiation.
Connection IDs are unencrypted but integrity protected in all
packets.
This version of QUIC does not incorporate a version negotiation
mechanism; implementations of incompatible versions will simply fail
to establish a connection.
21.12.2. Protected Packets
Packet protection (Section 12.1) provides authentication and
encryption of all packets except Version Negotiation packets, though
Initial and Retry packets have limited encryption and authentication
based on version-specific keys; see [QUIC-TLS] for more details.
This section considers passive and active attacks against protected
packets.
Both on-path and off-path attackers can mount a passive attack in
which they save observed packets for an offline attack against packet
protection at a future time; this is true for any observer of any
packet on any network.
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A blind attacker, one who injects packets without being able to
observe valid packets for a connection, is unlikely to be successful,
since packet protection ensures that valid packets are only generated
by endpoints which possess the key material established during the
handshake; see Section 7 and Section 21.12.1. Similarly, any active
attacker that observes packets and attempts to insert new data or
modify existing data in those packets should not be able to generate
packets deemed valid by the receiving endpoint.
A spoofing attack, in which an active attacker rewrites unprotected
parts of a packet that it forwards or injects, such as the source or
destination address, is only effective if the attacker can forward
packets to the original endpoint. Packet protection ensures that the
packet payloads can only be processed by the endpoints that completed
the handshake, and invalid packets are ignored by those endpoints.
An attacker can also modify the boundaries between packets and UDP
datagrams, causing multiple packets to be coalesced into a single
datagram, or splitting coalesced packets into multiple datagrams.
Aside from datagrams containing Initial packets, which require
padding, modification of how packets are arranged in datagrams has no
functional effect on a connection, although it might change some
performance characteristics.
21.12.3. Connection Migration
Connection Migration (Section 9) provides endpoints with the ability
to transition between IP addresses and ports on multiple paths, using
one path at a time for transmission and receipt of non-probing
frames. Path validation (Section 8.2) establishes that a peer is
both willing and able to receive packets sent on a particular path.
This helps reduce the effects of address spoofing by limiting the
number of packets sent to a spoofed address.
This section describes the intended security properties of connection
migration when under various types of DoS attacks.
21.12.3.1. On-Path Active Attacks
An attacker that can cause a packet it observes to no longer reach
its intended destination is considered an on-path attacker. When an
attacker is present between a client and server, endpoints are
required to send packets through the attacker to establish
connectivity on a given path.
An on-path attacker can:
* Inspect packets
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* Modify IP and UDP packet headers
* Inject new packets
* Delay packets
* Reorder packets
* Drop packets
* Split and merge datagrams along packet boundaries
An on-path attacker cannot:
* Modify an authenticated portion of a packet and cause the
recipient to accept that packet
An on-path attacker has the opportunity to modify the packets that it
observes, however any modifications to an authenticated portion of a
packet will cause it to be dropped by the receiving endpoint as
invalid, as packet payloads are both authenticated and encrypted.
In the presence of an on-path attacker, QUIC aims to provide the
following properties:
1. An on-path attacker can prevent use of a path for a connection,
causing it to fail if it cannot use a different path that does
not contain the attacker. This can be achieved by dropping all
packets, modifying them so that they fail to decrypt, or other
methods.
2. An on-path attacker can prevent migration to a new path for which
the attacker is also on-path by causing path validation to fail
on the new path.
3. An on-path attacker cannot prevent a client from migrating to a
path for which the attacker is not on-path.
4. An on-path attacker can reduce the throughput of a connection by
delaying packets or dropping them.
5. An on-path attacker cannot cause an endpoint to accept a packet
for which it has modified an authenticated portion of that
packet.
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21.12.3.2. Off-Path Active Attacks
An off-path attacker is not directly on the path between a client and
server, but could be able to obtain copies of some or all packets
sent between the client and the server. It is also able to send
copies of those packets to either endpoint.
An off-path attacker can:
* Inspect packets
* Inject new packets
* Reorder injected packets
An off-path attacker cannot:
* Modify any part of a packet
* Delay packets
* Drop packets
* Reorder original packets
An off-path attacker can modify packets that it has observed and
inject them back into the network, potentially with spoofed source
and destination addresses.
For the purposes of this discussion, it is assumed that an off-path
attacker has the ability to observe, modify, and re-inject a packet
into the network that will reach the destination endpoint prior to
the arrival of the original packet observed by the attacker. In
other words, an attacker has the ability to consistently "win" a race
with the legitimate packets between the endpoints, potentially
causing the original packet to be ignored by the recipient.
It is also assumed that an attacker has the resources necessary to
affect NAT state, potentially both causing an endpoint to lose its
NAT binding, and an attacker to obtain the same port for use with its
traffic.
In the presence of an off-path attacker, QUIC aims to provide the
following properties:
1. An off-path attacker can race packets and attempt to become a
"limited" on-path attacker.
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2. An off-path attacker can cause path validation to succeed for
forwarded packets with the source address listed as the off-path
attacker as long as it can provide improved connectivity between
the client and the server.
3. An off-path attacker cannot cause a connection to close once the
handshake has completed.
4. An off-path attacker cannot cause migration to a new path to fail
if it cannot observe the new path.
5. An off-path attacker can become a limited on-path attacker during
migration to a new path for which it is also an off-path
attacker.
6. An off-path attacker can become a limited on-path attacker by
affecting shared NAT state such that it sends packets to the
server from the same IP address and port that the client
originally used.
21.12.3.3. Limited On-Path Active Attacks
A limited on-path attacker is an off-path attacker that has offered
improved routing of packets by duplicating and forwarding original
packets between the server and the client, causing those packets to
arrive before the original copies such that the original packets are
dropped by the destination endpoint.
A limited on-path attacker differs from an on-path attacker in that
it is not on the original path between endpoints, and therefore the
original packets sent by an endpoint are still reaching their
destination. This means that a future failure to route copied
packets to the destination faster than their original path will not
prevent the original packets from reaching the destination.
A limited on-path attacker can:
* Inspect packets
* Inject new packets
* Modify unencrypted packet headers
* Reorder packets
A limited on-path attacker cannot:
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* Delay packets so that they arrive later than packets sent on the
original path
* Drop packets
* Modify the authenticated and encrypted portion of a packet and
cause the recipient to accept that packet
A limited on-path attacker can only delay packets up to the point
that the original packets arrive before the duplicate packets,
meaning that it cannot offer routing with worse latency than the
original path. If a limited on-path attacker drops packets, the
original copy will still arrive at the destination endpoint.
In the presence of a limited on-path attacker, QUIC aims to provide
the following properties:
1. A limited on-path attacker cannot cause a connection to close
once the handshake has completed.
2. A limited on-path attacker cannot cause an idle connection to
close if the client is first to resume activity.
3. A limited on-path attacker can cause an idle connection to be
deemed lost if the server is the first to resume activity.
Note that these guarantees are the same guarantees provided for any
NAT, for the same reasons.
22. IANA Considerations
This document establishes several registries for the management of
codepoints in QUIC. These registries operate on a common set of
policies as defined in Section 22.1.
22.1. Registration Policies for QUIC Registries
All QUIC registries allow for both provisional and permanent
registration of codepoints. This section documents policies that are
common to these registries.
22.1.1. Provisional Registrations
Provisional registration of codepoints are intended to allow for
private use and experimentation with extensions to QUIC. Provisional
registrations only require the inclusion of the codepoint value and
contact information. However, provisional registrations could be
reclaimed and reassigned for another purpose.
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Provisional registrations require Expert Review, as defined in
Section 4.5 of [RFC8126]. Designated expert(s) are advised that only
registrations for an excessive proportion of remaining codepoint
space or the very first unassigned value (see Section 22.1.2) can be
rejected.
Provisional registrations will include a date field that indicates
when the registration was last updated. A request to update the date
on any provisional registration can be made without review from the
designated expert(s).
All QUIC registries include the following fields to support
provisional registration:
Value: The assigned codepoint.
Status: "Permanent" or "Provisional".
Specification: A reference to a publicly available specification for
the value.
Date: The date of last update to the registration.
Contact: Contact details for the registrant.
Notes: Supplementary notes about the registration.
Provisional registrations MAY omit the Specification and Notes
fields, plus any additional fields that might be required for a
permanent registration. The Date field is not required as part of
requesting a registration as it is set to the date the registration
is created or updated.
22.1.2. Selecting Codepoints
New uses of codepoints from QUIC registries SHOULD use a randomly
selected codepoint that excludes both existing allocations and the
first unallocated codepoint in the selected space. Requests for
multiple codepoints MAY use a contiguous range. This minimizes the
risk that differing semantics are attributed to the same codepoint by
different implementations. Use of the first codepoint in a range is
intended for use by specifications that are developed through the
standards process [STD] and its allocation MUST be negotiated with
IANA before use.
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For codepoints that are encoded in variable-length integers
(Section 16), such as frame types, codepoints that encode to four or
eight bytes (that is, values 2^14 and above) SHOULD be used unless
the usage is especially sensitive to having a longer encoding.
Applications to register codepoints in QUIC registries MAY include a
codepoint as part of the registration. IANA MUST allocate the
selected codepoint unless that codepoint is already assigned or the
codepoint is the first unallocated codepoint in the registry.
22.1.3. Reclaiming Provisional Codepoints
A request might be made to remove an unused provisional registration
from the registry to reclaim space in a registry, or portion of the
registry (such as the 64-16383 range for codepoints that use
variable-length encodings). This SHOULD be done only for the
codepoints with the earliest recorded date and entries that have been
updated less than a year prior SHOULD NOT be reclaimed.
A request to remove a codepoint MUST be reviewed by the designated
expert(s). The expert(s) MUST attempt to determine whether the
codepoint is still in use. Experts are advised to contact the listed
contacts for the registration, plus as wide a set of protocol
implementers as possible in order to determine whether any use of the
codepoint is known. The expert(s) are advised to allow at least four
weeks for responses.
If any use of the codepoints is identified by this search or a
request to update the registration is made, the codepoint MUST NOT be
reclaimed. Instead, the date on the registration is updated. A note
might be added for the registration recording relevant information
that was learned.
If no use of the codepoint was identified and no request was made to
update the registration, the codepoint MAY be removed from the
registry.
This process also applies to requests to change a provisional
registration into a permanent registration, except that the goal is
not to determine whether there is no use of the codepoint, but to
determine that the registration is an accurate representation of any
deployed usage.
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22.1.4. Permanent Registrations
Permanent registrations in QUIC registries use the Specification
Required policy [RFC8126], unless otherwise specified. The
designated expert(s) verify that a specification exists and is
readily accessible. 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 creation of a registry MAY specify
additional constraints on permanent registrations.
The creation of a registries MAY identify a range of codepoints where
registrations are governed by a different registration policy. For
instance, the registries for 62-bit codepoints in this document have
stricter policies for codepoints in the range from 0 to 63.
Any stricter requirements for permanent registrations do not prevent
provisional registrations for affected codepoints. For instance, a
provisional registration for a frame type Section 22.3 of 61 could be
requested.
All registrations made by Standards Track publications MUST be
permanent.
All registrations in this document are assigned a permanent status
and list as contact both the IESG (ietf@ietf.org) and the QUIC
working group (quic@ietf.org (mailto:quic@ietf.org)).
22.2. QUIC Transport Parameter Registry
IANA [SHALL add/has added] a registry for "QUIC Transport Parameters"
under a "QUIC" heading.
The "QUIC Transport Parameters" registry governs a 62-bit space.
This registry follows the registration policy from Section 22.1.
Permanent registrations in this registry are assigned using the
Specification Required policy [RFC8126].
In addition to the fields in Section 22.1.1, permanent registrations
in this registry MUST include the following fields:
Parameter Name: A short mnemonic for the parameter.
The initial contents of this registry are shown in Table 6.
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+-------+-------------------------------------+---------------+
| Value | Parameter Name | Specification |
+=======+=====================================+===============+
| 0x00 | original_destination_connection_id | Section 18.2 |
+-------+-------------------------------------+---------------+
| 0x01 | max_idle_timeout | Section 18.2 |
+-------+-------------------------------------+---------------+
| 0x02 | stateless_reset_token | Section 18.2 |
+-------+-------------------------------------+---------------+
| 0x03 | max_udp_payload_size | Section 18.2 |
+-------+-------------------------------------+---------------+
| 0x04 | initial_max_data | Section 18.2 |
+-------+-------------------------------------+---------------+
| 0x05 | initial_max_stream_data_bidi_local | Section 18.2 |
+-------+-------------------------------------+---------------+
| 0x06 | initial_max_stream_data_bidi_remote | Section 18.2 |
+-------+-------------------------------------+---------------+
| 0x07 | initial_max_stream_data_uni | Section 18.2 |
+-------+-------------------------------------+---------------+
| 0x08 | initial_max_streams_bidi | Section 18.2 |
+-------+-------------------------------------+---------------+
| 0x09 | initial_max_streams_uni | Section 18.2 |
+-------+-------------------------------------+---------------+
| 0x0a | ack_delay_exponent | Section 18.2 |
+-------+-------------------------------------+---------------+
| 0x0b | max_ack_delay | Section 18.2 |
+-------+-------------------------------------+---------------+
| 0x0c | disable_active_migration | Section 18.2 |
+-------+-------------------------------------+---------------+
| 0x0d | preferred_address | Section 18.2 |
+-------+-------------------------------------+---------------+
| 0x0e | active_connection_id_limit | Section 18.2 |
+-------+-------------------------------------+---------------+
| 0x0f | initial_source_connection_id | Section 18.2 |
+-------+-------------------------------------+---------------+
| 0x10 | retry_source_connection_id | Section 18.2 |
+-------+-------------------------------------+---------------+
Table 6: Initial QUIC Transport Parameters Entries
Additionally, each value of the format "31 * N + 27" for integer
values of N (that is, 27, 58, 89, ...) are reserved and MUST NOT be
assigned by IANA.
22.3. QUIC Frame Type Registry
IANA [SHALL add/has added] a registry for "QUIC Frame Types" under a
"QUIC" heading.
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The "QUIC Frame Types" registry governs a 62-bit space. This
registry follows the registration policy from Section 22.1.
Permanent registrations in this registry are assigned using the
Specification Required policy [RFC8126], except for values between
0x00 and 0x3f (in hexadecimal; inclusive), which are assigned using
Standards Action or IESG Approval as defined in Section 4.9 and 4.10
of [RFC8126].
In addition to the fields in Section 22.1.1, permanent registrations
in this registry MUST include the following fields:
Frame Name: A short mnemonic for the frame type.
In addition to the advice in Section 22.1, specifications for new
permanent registrations SHOULD describe the means by which an
endpoint might determine that it can send the identified type of
frame. An accompanying transport parameter registration is expected
for most registrations; see Section 22.2. Specifications for
permanent registrations also needs to describe the format and
assigned semantics of any fields in the frame.
The initial contents of this registry are tabulated in Table 3.
22.4. QUIC Transport Error Codes Registry
IANA [SHALL add/has added] a registry for "QUIC Transport Error
Codes" under a "QUIC" heading.
The "QUIC Transport Error Codes" registry governs a 62-bit space.
This space is split into three spaces that are governed by different
policies. Permanent registrations in this registry are assigned
using the Specification Required policy [RFC8126], except for values
between 0x00 and 0x3f (in hexadecimal; inclusive), which are assigned
using Standards Action or IESG Approval as defined in Section 4.9 and
4.10 of [RFC8126].
In addition to the fields in Section 22.1.1, permanent registrations
in this registry MUST include the following fields:
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.
The initial contents of this registry are shown in Table 7.
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+------+---------------------------+----------------+---------------+
|Value | Error | Description | Specification |
+======+===========================+================+===============+
| 0x0 | NO_ERROR | No error | Section 20 |
+------+---------------------------+----------------+---------------+
| 0x1 | INTERNAL_ERROR | Implementation | Section 20 |
| | | error | |
+------+---------------------------+----------------+---------------+
| 0x2 | SERVER_BUSY |Server currently| Section 20 |
| | | busy | |
+------+---------------------------+----------------+---------------+
| 0x3 | FLOW_CONTROL_ERROR | Flow control | Section 20 |
| | | error | |
+------+---------------------------+----------------+---------------+
| 0x4 | STREAM_LIMIT_ERROR |Too many streams| Section 20 |
| | | opened | |
+------+---------------------------+----------------+---------------+
| 0x5 | STREAM_STATE_ERROR | Frame received | Section 20 |
| | | in invalid | |
| | | stream state | |
+------+---------------------------+----------------+---------------+
| 0x6 | FINAL_SIZE_ERROR |Change to final | Section 20 |
| | | size | |
+------+---------------------------+----------------+---------------+
| 0x7 | FRAME_ENCODING_ERROR | Frame encoding | Section 20 |
| | | error | |
+------+---------------------------+----------------+---------------+
| 0x8 | TRANSPORT_PARAMETER_ERROR | Error in | Section 20 |
| | | transport | |
| | | parameters | |
+------+---------------------------+----------------+---------------+
| 0x9 | CONNECTION_ID_LIMIT_ERROR | Too many | Section 20 |
| | | connection IDs | |
| | | received | |
+------+---------------------------+----------------+---------------+
| 0xA | PROTOCOL_VIOLATION |Generic protocol| Section 20 |
| | | violation | |
+------+---------------------------+----------------+---------------+
| 0xB | INVALID_TOKEN | Invalid Token | Section 20 |
| | | Received | |
+------+---------------------------+----------------+---------------+
| 0xC | APPLICATION_ERROR | Application | Section 20 |
| | | error | |
+------+---------------------------+----------------+---------------+
| 0xD | CRYPTO_BUFFER_EXCEEDED | CRYPTO data | Section 20 |
| | | buffer | |
| | | overflowed | |
+------+---------------------------+----------------+---------------+
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Table 7: Initial QUIC Transport Error Codes Entries
23. References
23.1. Normative References
[DPLPMTUD] Fairhurst, G., Jones, T., Tuexen, M., Ruengeler, I., and
T. Voelker, "Packetization Layer Path MTU Discovery for
Datagram Transports", Work in Progress, Internet-Draft,
draft-ietf-tsvwg-datagram-plpmtud-21, 12 May 2020,
<http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-
datagram-plpmtud-21.txt>.
[IPv4] Postel, J., "Internet Protocol", STD 5, RFC 791,
DOI 10.17487/RFC0791, September 1981,
<https://www.rfc-editor.org/info/rfc791>.
[QUIC-RECOVERY]
Iyengar, J., Ed. and I. Swett, Ed., "QUIC Loss Detection
and Congestion Control", Work in Progress, Internet-Draft,
draft-ietf-quic-recovery-28, 20 May 2020,
<https://tools.ietf.org/html/draft-ietf-quic-recovery-28>.
[QUIC-TLS] Thomson, M., Ed. and S. Turner, Ed., "Using Transport
Layer Security (TLS) to Secure QUIC", Work in Progress,
Internet-Draft, draft-ietf-quic-tls-28, 20 May 2020,
<https://tools.ietf.org/html/draft-ietf-quic-tls-28>.
[RFC1191] Mogul, J.C. and S.E. 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>.
[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP",
RFC 3168, DOI 10.17487/RFC3168, September 2001,
<https://www.rfc-editor.org/info/rfc3168>.
[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>.
[RFC5116] McGrew, D., "An Interface and Algorithms for Authenticated
Encryption", RFC 5116, DOI 10.17487/RFC5116, January 2008,
<https://www.rfc-editor.org/info/rfc5116>.
[RFC6437] Amante, S., Carpenter, B., Jiang, S., and J. Rajahalme,
"IPv6 Flow Label Specification", RFC 6437,
DOI 10.17487/RFC6437, November 2011,
<https://www.rfc-editor.org/info/rfc6437>.
[RFC8085] Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage
Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085,
March 2017, <https://www.rfc-editor.org/info/rfc8085>.
[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>.
[RFC8201] 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>.
[RFC8311] Black, D., "Relaxing Restrictions on Explicit Congestion
Notification (ECN) Experimentation", RFC 8311,
DOI 10.17487/RFC8311, January 2018,
<https://www.rfc-editor.org/info/rfc8311>.
[TLS13] Rescorla, E., "The Transport Layer Security (TLS) Protocol
Version 1.3", RFC 8446, DOI 10.17487/RFC8446, August 2018,
<https://www.rfc-editor.org/info/rfc8446>.
23.2. Informative References
[ALTSVC] Nottingham, M., McManus, P., and J. Reschke, "HTTP
Alternative Services", RFC 7838, DOI 10.17487/RFC7838,
April 2016, <https://www.rfc-editor.org/info/rfc7838>.
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[EARLY-DESIGN]
Roskind, J., "QUIC: Multiplexed Transport Over UDP", 2
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>.
[QUIC-INVARIANTS]
Thomson, M., "Version-Independent Properties of QUIC",
Work in Progress, Internet-Draft, draft-ietf-quic-
invariants-08, 20 May 2020, <https://tools.ietf.org/html/
draft-ietf-quic-invariants-08>.
[QUIC-MANAGEABILITY]
Kuehlewind, M. and B. Trammell, "Manageability of the QUIC
Transport Protocol", Work in Progress, Internet-Draft,
draft-ietf-quic-manageability-06, 6 January 2020,
<http://www.ietf.org/internet-drafts/draft-ietf-quic-
manageability-06.txt>.
[RFC1812] Baker, F., Ed., "Requirements for IP Version 4 Routers",
RFC 1812, DOI 10.17487/RFC1812, June 1995,
<https://www.rfc-editor.org/info/rfc1812>.
[RFC2018] Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP
Selective Acknowledgment Options", RFC 2018,
DOI 10.17487/RFC2018, October 1996,
<https://www.rfc-editor.org/info/rfc2018>.
[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>.
[RFC4303] Kent, S., "IP Encapsulating Security Payload (ESP)",
RFC 4303, DOI 10.17487/RFC4303, December 2005,
<https://www.rfc-editor.org/info/rfc4303>.
[RFC4443] Conta, A., Deering, S., and M. Gupta, Ed., "Internet
Control Message Protocol (ICMPv6) for the Internet
Protocol Version 6 (IPv6) Specification", STD 89,
RFC 4443, DOI 10.17487/RFC4443, March 2006,
<https://www.rfc-editor.org/info/rfc4443>.
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[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>.
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
<https://www.rfc-editor.org/info/rfc5681>.
[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>.
[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>.
[RFC8200] Deering, S. and R. Hinden, "Internet Protocol, Version 6
(IPv6) Specification", STD 86, RFC 8200,
DOI 10.17487/RFC8200, July 2017,
<https://www.rfc-editor.org/info/rfc8200>.
[SEC-CONS] Rescorla, E. and B. Korver, "Guidelines for Writing RFC
Text on Security Considerations", BCP 72, RFC 3552,
DOI 10.17487/RFC3552, July 2003,
<https://www.rfc-editor.org/info/rfc3552>.
[SLOWLORIS]
RSnake Hansen, R., "Welcome to Slowloris...", June 2009,
<https://web.archive.org/web/20150315054838/
http://ha.ckers.org/slowloris/>.
[STD] Bradner, S., "The Internet Standards Process -- Revision
3", BCP 9, RFC 2026, DOI 10.17487/RFC2026, October 1996,
<https://www.rfc-editor.org/info/rfc2026>.
Appendix A. Sample Packet Number Decoding Algorithm
The pseudo-code in Figure 41 shows how an implementation can decode
packet numbers after header protection has been removed.
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DecodePacketNumber(largest_pn, truncated_pn, pn_nbits):
expected_pn = largest_pn + 1
pn_win = 1 << pn_nbits
pn_hwin = pn_win / 2
pn_mask = pn_win - 1
// The incoming packet number should be greater than
// expected_pn - pn_hwin and less than or equal to
// expected_pn + pn_hwin
//
// This means we can't just strip the trailing bits from
// expected_pn and add the truncated_pn because that might
// yield a value outside the window.
//
// The following code calculates a candidate value and
// makes sure it's within the packet number window.
// Note the extra checks to prevent overflow and underflow.
candidate_pn = (expected_pn & ~pn_mask) | truncated_pn
if candidate_pn <= expected_pn - pn_hwin and
candidate_pn < (1 << 62) - pn_win:
return candidate_pn + pn_win
if candidate_pn > expected_pn + pn_hwin and
candidate_pn >= pn_win:
return candidate_pn - pn_win
return candidate_pn
Figure 41: Sample Packet Number Decoding Algorithm
Appendix B. Sample ECN Validation Algorithm
Each time an endpoint commences sending on a new network path, it
determines whether the path supports ECN; see Section 13.4. If the
path supports ECN, the goal is to use ECN. Endpoints might also
periodically reassess a path that was determined to not support ECN.
This section describes one method for testing new paths. This
algorithm is intended to show how a path might be tested for ECN
support. Endpoints can implement different methods.
The path is assigned an ECN state that is one of "testing",
"unknown", "failed", or "capable". On paths with a "testing" or
"capable" state the endpoint sends packets with an ECT marking, by
default ECT(0); otherwise, the endpoint sends unmarked packets.
To start testing a path, the ECN state is set to "testing" and
existing ECN counts are remembered as a baseline.
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The testing period runs for a number of packets or round-trip times,
as determined by the endpoint. The goal is not to limit the duration
of the testing period, but to ensure that enough marked packets are
sent for received ECN counts to provide a clear indication of how the
path treats marked packets. Section 13.4.2.2 suggests limiting this
to 10 packets or 3 round-trip times.
After the testing period ends, the ECN state for the path becomes
"unknown". From the "unknown" state, successful validation of the
ECN counts an ACK frame (see Section 13.4.2.2) causes the ECN state
for the path to become "capable", unless no marked packet has been
acknowledged.
If validation of ECN counts fails at any time, the ECN state for the
affected path becomes "failed". An endpoint can also mark the ECN
state for a path as "failed" if marked packets are all declared lost
or if they are all CE marked.
Following this algorithm ensures that ECN is rarely disabled for
paths that properly support ECN. Any path that incorrectly modifies
markings will cause ECN to be disabled. For those rare cases where
marked packets are discarded by the path, the short duration of the
testing period limits the number of losses incurred.
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-27
* Allowed CONNECTION_CLOSE in any packet number space, with a
requirement to use a new transport-level error for application-
specific errors in Initial and Handshake packets (#3430, #3435,
#3440)
* Clearer requirements for address validation (#2125, #3327)
* Security analysis of handshake and migration (#2143, #2387, #2925)
* The entire payload of a datagram is used when counting bytes for
mitigating amplification attacks (#3333, #3470)
* Connection IDs can be used at any time, including in the handshake
(#3348, #3560, #3438, #3565)
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* Only one ACK should be sent for each instance of reordering
(#3357, #3361)
* Remove text allowing a server to proceed with a bad Retry token
(#3396, #3398)
* Ignore active_connection_id_limit with a zero-length connection ID
(#3427, #3426)
* Require active_connection_id_limit be remembered for 0-RTT (#3423,
#3425)
* Require ack_delay not be remembered for 0-RTT (#3433, #3545)
* Redefined max_packet_size to max_udp_datagram_size (#3471, #3473)
* Guidance on limiting outstanding attempts to retire connection IDs
(#3489, #3509, #3557, #3547)
* Restored text on dropping bogus Version Negotiation packets
(#3532, #3533)
* Clarified that largest acknowledged needs to be saved, but not
necessarily signaled in all cases (#3541, #3581)
* Addressed linkability risk with the use of preferred_address
(#3559, #3563)
C.2. Since draft-ietf-quic-transport-26
* Change format of transport parameters to use varints (#3294,
#3169)
C.3. Since draft-ietf-quic-transport-25
* Define the use of CONNECTION_CLOSE prior to establishing
connection state (#3269, #3297, #3292)
* Allow use of address validation tokens after client address
changes (#3307, #3308)
* Define the timer for address validation (#2910, #3339)
C.4. Since draft-ietf-quic-transport-24
* Added HANDSHAKE_DONE to signal handshake confirmation (#2863,
#3142, #3145)
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* Add integrity check to Retry packets (#3014, #3274, #3120)
* Specify handling of reordered NEW_CONNECTION_ID frames (#3194,
#3202)
* Require checking of sequence numbers in RETIRE_CONNECTION_ID
(#3037, #3036)
* active_connection_id_limit is enforced (#3193, #3197, #3200,
#3201)
* Correct overflow in packet number decode algorithm (#3187, #3188)
* Allow use of CRYPTO_BUFFER_EXCEEDED for CRYPTO frame errors
(#3258, #3186)
* Define applicability and scope of NEW_TOKEN (#3150, #3152, #3155,
#3156)
* Tokens from Retry and NEW_TOKEN must be differentiated (#3127,
#3128)
* Allow CONNECTION_CLOSE in response to invalid token (#3168, #3107)
* Treat an invalid CONNECTION_CLOSE as an invalid frame (#2475,
#3230, #3231)
* Throttle when sending CONNECTION_CLOSE after discarding state
(#3095, #3157)
* Application-variant of CONNECTION_CLOSE can only be sent in 0-RTT
or 1-RTT packets (#3158, #3164)
* Advise sending while blocked to avoid idle timeout (#2744, #3266)
* Define error codes for invalid frames (#3027, #3042)
* Idle timeout is symmetric (#2602, #3099)
* Prohibit IP fragmentation (#3243, #3280)
* Define the use of provisional registration for all registries
(#3109, #3020, #3102, #3170)
* Packets on one path must not adjust values for a different path
(#2909, #3139)
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C.5. Since draft-ietf-quic-transport-23
* Allow ClientHello to span multiple packets (#2928, #3045)
* Client Initial size constraints apply to UDP datagram payload
(#3053, #3051)
* Stateless reset changes (#2152, #2993)
- tokens need to be compared in constant time
- detection uses UDP datagrams, not packets
- tokens cannot be reused (#2785, #2968)
* Clearer rules for sharing of UDP ports and use of connection IDs
when doing so (#2844, #2851)
* A new connection ID is necessary when responding to migration
(#2778, #2969)
* Stronger requirements for connection ID retirement (#3046, #3096)
* NEW_TOKEN cannot be empty (#2978, #2977)
* PING can be sent at any encryption level (#3034, #3035)
* CONNECTION_CLOSE is not ack-eliciting (#3097, #3098)
* Frame encoding error conditions updated (#3027, #3042)
* Non-ack-eliciting packets cannot be sent in response to non-ack-
eliciting packets (#3100, #3104)
* Servers have to change connection IDs in Retry (#2837, #3147)
C.6. Since draft-ietf-quic-transport-22
* Rules for preventing correlation by connection ID tightened
(#2084, #2929)
* Clarified use of CONNECTION_CLOSE in Handshake packets (#2151,
#2541, #2688)
* Discourage regressions of largest acknowledged in ACK (#2205,
#2752)
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* Improved robustness of validation process for ECN counts (#2534,
#2752)
* Require endpoints to ignore spurious migration attempts (#2342,
#2893)
* Transport parameter for disabling migration clarified to allow NAT
rebinding (#2389, #2893)
* Document principles for defining new error codes (#2388, #2880)
* Reserve transport parameters for greasing (#2550, #2873)
* A maximum ACK delay of 0 is used for handshake packet number
spaces (#2646, #2638)
* Improved rules for use of congestion control state on new paths
(#2685, #2918)
* Removed recommendation to coordinate spin for multiple connections
that share a path (#2763, #2882)
* Allow smaller stateless resets and recommend a smaller minimum on
packets that might trigger a stateless reset (#2770, #2869, #2927,
#3007).
* Provide guidance around the interface to QUIC as used by
application protocols (#2805, #2857)
* Frames other than STREAM can cause STREAM_LIMIT_ERROR (#2825,
#2826)
* Tighter rules about processing of rejected 0-RTT packets (#2829,
#2840, #2841)
* Explanation of the effect of Retry on 0-RTT packets (#2842, #2852)
* Cryptographic handshake needs to provide server transport
parameter encryption (#2920, #2921)
* Moved ACK generation guidance from recovery draft to transport
draft (#1860, #2916).
C.7. Since draft-ietf-quic-transport-21
* Connection ID lengths are now one octet, but limited in version 1
to 20 octets of length (#2736, #2749)
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C.8. Since draft-ietf-quic-transport-20
* Error codes are encoded as variable-length integers (#2672, #2680)
* NEW_CONNECTION_ID includes a request to retire old connection IDs
(#2645, #2769)
* Tighter rules for generating and explicitly eliciting ACK frames
(#2546, #2794)
* Recommend having only one packet per encryption level in a
datagram (#2308, #2747)
* More normative language about use of stateless reset (#2471,
#2574)
* Allow reuse of stateless reset tokens (#2732, #2733)
* Allow, but not require, enforcing non-duplicate transport
parameters (#2689, #2691)
* Added an active_connection_id_limit transport parameter (#1994,
#1998)
* max_ack_delay transport parameter defaults to 0 (#2638, #2646)
* When sending 0-RTT, only remembered transport parameters apply
(#2458, #2360, #2466, #2461)
* Define handshake completion and confirmation; define clearer rules
when it encryption keys should be discarded (#2214, #2267, #2673)
* Prohibit path migration prior to handshake confirmation (#2309,
#2370)
* PATH_RESPONSE no longer needs to be received on the validated path
(#2582, #2580, #2579, #2637)
* PATH_RESPONSE frames are not stored and retransmitted (#2724,
#2729)
* Document hack for enabling routing of ICMP when doing PMTU probing
(#1243, #2402)
C.9. Since draft-ietf-quic-transport-19
* Refine discussion of 0-RTT transport parameters (#2467, #2464)
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* Fewer transport parameters need to be remembered for 0-RTT (#2624,
#2467)
* Spin bit text incorporated (#2564)
* Close the connection when maximum stream ID in MAX_STREAMS exceeds
2^62 - 1 (#2499, #2487)
* New connection ID required for intentional migration (#2414,
#2413)
* Connection ID issuance can be rate-limited (#2436, #2428)
* The "QUIC bit" is ignored in Version Negotiation (#2400, #2561)
* Initial packets from clients need to be padded to 1200 unless a
Handshake packet is sent as well (#2522, #2523)
* CRYPTO frames can be discarded if too much data is buffered
(#1834, #2524)
* Stateless reset uses a short header packet (#2599, #2600)
C.10. Since draft-ietf-quic-transport-18
* Removed version negotiation; version negotiation, including
authentication of the result, will be addressed in the next
version of QUIC (#1773, #2313)
* Added discussion of the use of IPv6 flow labels (#2348, #2399)
* A connection ID can't be retired in a packet that uses that
connection ID (#2101, #2420)
* Idle timeout transport parameter is in milliseconds (from seconds)
(#2453, #2454)
* Endpoints are required to use new connection IDs when they use new
network paths (#2413, #2414)
* Increased the set of permissible frames in 0-RTT (#2344, #2355)
C.11. Since draft-ietf-quic-transport-17
* Stream-related errors now use STREAM_STATE_ERROR (#2305)
* Endpoints discard initial keys as soon as handshake keys are
available (#1951, #2045)
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* Expanded conditions for ignoring ICMP packet too big messages
(#2108, #2161)
* Remove rate control from PATH_CHALLENGE/PATH_RESPONSE (#2129,
#2241)
* Endpoints are permitted to discard malformed initial packets
(#2141)
* Clarified ECN implementation and usage requirements (#2156, #2201)
* Disable ECN count verification for packets that arrive out of
order (#2198, #2215)
* Use Probe Timeout (PTO) instead of RTO (#2206, #2238)
* Loosen constraints on retransmission of ACK ranges (#2199, #2245)
* Limit Retry and Version Negotiation to once per datagram (#2259,
#2303)
* Set a maximum value for max_ack_delay transport parameter (#2282,
#2301)
* Allow server preferred address for both IPv4 and IPv6 (#2122,
#2296)
* Corrected requirements for migration to a preferred address
(#2146, #2349)
* ACK of non-existent packet is illegal (#2298, #2302)
C.12. Since draft-ietf-quic-transport-16
* Stream limits are defined as counts, not maximums (#1850, #1906)
* Require amplification attack defense after closing (#1905, #1911)
* Remove reservation of application error code 0 for STOPPING
(#1804, #1922)
* Renumbered frames (#1945)
* Renumbered transport parameters (#1946)
* Numeric transport parameters are expressed as varints (#1608,
#1947, #1955)
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* Reorder the NEW_CONNECTION_ID frame (#1952, #1963)
* Rework the first byte (#2006)
- Fix the 0x40 bit
- Change type values for long header
- Add spin bit to short header (#631, #1988)
- Encrypt the remainder of the first byte (#1322)
- Move packet number length to first byte
- Move ODCIL to first byte of retry packets
- Simplify packet number protection (#1575)
* Allow STOP_SENDING to open a remote bidirectional stream (#1797,
#2013)
* Added mitigation for off-path migration attacks (#1278, #1749,
#2033)
* Don't let the PMTU to drop below 1280 (#2063, #2069)
* Require peers to replace retired connection IDs (#2085)
* Servers are required to ignore Version Negotiation packets (#2088)
* Tokens are repeated in all Initial packets (#2089)
* Clarified how PING frames are sent after loss (#2094)
* Initial keys are discarded once Handshake are available (#1951,
#2045)
* ICMP PTB validation clarifications (#2161, #2109, #2108)
C.13. Since draft-ietf-quic-transport-15
Substantial editorial reorganization; no technical changes.
C.14. Since draft-ietf-quic-transport-14
* Merge ACK and ACK_ECN (#1778, #1801)
* Explicitly communicate max_ack_delay (#981, #1781)
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* Validate original connection ID after Retry packets (#1710, #1486,
#1793)
* Idle timeout is optional and has no specified maximum (#1765)
* Update connection ID handling; add RETIRE_CONNECTION_ID type
(#1464, #1468, #1483, #1484, #1486, #1495, #1729, #1742, #1799,
#1821)
* Include a Token in all Initial packets (#1649, #1794)
* Prevent handshake deadlock (#1764, #1824)
C.15. Since draft-ietf-quic-transport-13
* Streams open when higher-numbered streams of the same type open
(#1342, #1549)
* Split initial stream flow control limit into 3 transport
parameters (#1016, #1542)
* All flow control transport parameters are optional (#1610)
* Removed UNSOLICITED_PATH_RESPONSE error code (#1265, #1539)
* Permit stateless reset in response to any packet (#1348, #1553)
* Recommended defense against stateless reset spoofing (#1386,
#1554)
* Prevent infinite stateless reset exchanges (#1443, #1627)
* Forbid processing of the same packet number twice (#1405, #1624)
* Added a packet number decoding example (#1493)
* More precisely define idle timeout (#1429, #1614, #1652)
* Corrected format of Retry packet and prevented looping (#1492,
#1451, #1448, #1498)
* Permit 0-RTT after receiving Version Negotiation or Retry (#1507,
#1514, #1621)
* Permit Retry in response to 0-RTT (#1547, #1552)
* Looser verification of ECN counters to account for ACK loss
(#1555, #1481, #1565)
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* Remove frame type field from APPLICATION_CLOSE (#1508, #1528)
C.16. Since draft-ietf-quic-transport-12
* Changes to integration of the TLS handshake (#829, #1018, #1094,
#1165, #1190, #1233, #1242, #1252, #1450, #1458)
- The cryptographic handshake uses CRYPTO frames, not stream 0
- QUIC packet protection is used in place of TLS record
protection
- Separate QUIC packet number spaces are used for the handshake
- Changed Retry to be independent of the cryptographic handshake
- Added NEW_TOKEN frame and Token fields to Initial packet
- Limit the use of HelloRetryRequest to address TLS needs (like
key shares)
* Enable server to transition connections to a preferred address
(#560, #1251, #1373)
* Added ECN feedback mechanisms and handling; new ACK_ECN frame
(#804, #805, #1372)
* Changed rules and recommendations for use of new connection IDs
(#1258, #1264, #1276, #1280, #1419, #1452, #1453, #1465)
* Added a transport parameter to disable intentional connection
migration (#1271, #1447)
* Packets from different connection ID can't be coalesced (#1287,
#1423)
* Fixed sampling method for packet number encryption; the length
field in long headers includes the packet number field in addition
to the packet payload (#1387, #1389)
* Stateless Reset is now symmetric and subject to size constraints
(#466, #1346)
* Added frame type extension mechanism (#58, #1473)
C.17. Since draft-ietf-quic-transport-11
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* Enable server to transition connections to a preferred address
(#560, #1251)
* Packet numbers are encrypted (#1174, #1043, #1048, #1034, #850,
#990, #734, #1317, #1267, #1079)
* Packet numbers use a variable-length encoding (#989, #1334)
* STREAM frames can now be empty (#1350)
C.18. Since draft-ietf-quic-transport-10
* Swap payload length and packed number fields in long header
(#1294)
* Clarified that CONNECTION_CLOSE is allowed in Handshake packet
(#1274)
* Spin bit reserved (#1283)
* Coalescing multiple QUIC packets in a UDP datagram (#1262, #1285)
* A more complete connection migration (#1249)
* Refine opportunistic ACK defense text (#305, #1030, #1185)
* A Stateless Reset Token isn't mandatory (#818, #1191)
* Removed implicit stream opening (#896, #1193)
* An empty STREAM frame can be used to open a stream without sending
data (#901, #1194)
* Define stream counts in transport parameters rather than a maximum
stream ID (#1023, #1065)
* STOP_SENDING is now prohibited before streams are used (#1050)
* Recommend including ACK in Retry packets and allow PADDING (#1067,
#882)
* Endpoints now become closing after an idle timeout (#1178, #1179)
* Remove implication that Version Negotiation is sent when a packet
of the wrong version is received (#1197)
C.19. Since draft-ietf-quic-transport-09
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* Added PATH_CHALLENGE and PATH_RESPONSE frames to replace PING with
Data and PONG frame. Changed ACK frame type from 0x0e to 0x0d.
(#1091, #725, #1086)
* A server can now only send 3 packets without validating the client
address (#38, #1090)
* Delivery order of stream data is no longer strongly specified
(#252, #1070)
* Rework of packet handling and version negotiation (#1038)
* Stream 0 is now exempt from flow control until the handshake
completes (#1074, #725, #825, #1082)
* Improved retransmission rules for all frame types: information is
retransmitted, not packets or frames (#463, #765, #1095, #1053)
* Added an error code for server busy signals (#1137)
* Endpoints now set the connection ID that their peer uses.
Connection IDs are variable length. Removed the
omit_connection_id transport parameter and the corresponding short
header flag. (#1089, #1052, #1146, #821, #745, #821, #1166, #1151)
C.20. Since draft-ietf-quic-transport-08
* Clarified requirements for BLOCKED usage (#65, #924)
* BLOCKED frame now includes reason for blocking (#452, #924, #927,
#928)
* GAP limitation in ACK Frame (#613)
* Improved PMTUD description (#614, #1036)
* Clarified stream state machine (#634, #662, #743, #894)
* Reserved versions don't need to be generated deterministically
(#831, #931)
* You don't always need the draining period (#871)
* Stateless reset clarified as version-specific (#930, #986)
* initial_max_stream_id_x transport parameters are optional (#970,
#971)
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* Ack Delay assumes a default value during the handshake (#1007,
#1009)
* Removed transport parameters from NewSessionTicket (#1015)
C.21. Since draft-ietf-quic-transport-07
* The long header now has version before packet number (#926, #939)
* Rename and consolidate packet types (#846, #822, #847)
* Packet types are assigned new codepoints and the Connection ID
Flag is inverted (#426, #956)
* Removed type for Version Negotiation and use Version 0 (#963,
#968)
* Streams are split into unidirectional and bidirectional (#643,
#656, #720, #872, #175, #885)
- Stream limits now have separate uni- and bi-directional
transport parameters (#909, #958)
- Stream limit transport parameters are now optional and default
to 0 (#970, #971)
* The stream state machine has been split into read and write (#634,
#894)
* Employ variable-length integer encodings throughout (#595)
* Improvements to connection close
- Added distinct closing and draining states (#899, #871)
- Draining period can terminate early (#869, #870)
- Clarifications about stateless reset (#889, #890)
* Address validation for connection migration (#161, #732, #878)
* Clearly defined retransmission rules for BLOCKED (#452, #65, #924)
* negotiated_version is sent in server transport parameters (#710,
#959)
* Increased the range over which packet numbers are randomized
(#864, #850, #964)
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C.22. Since draft-ietf-quic-transport-06
* Replaced FNV-1a with AES-GCM for all "Cleartext" packets (#554)
* Split error code space between application and transport (#485)
* Stateless reset token moved to end (#820)
* 1-RTT-protected long header types removed (#848)
* No acknowledgments during draining period (#852)
* Remove "application close" as a separate close type (#854)
* Remove timestamps from the ACK frame (#841)
* Require transport parameters to only appear once (#792)
C.23. Since draft-ietf-quic-transport-05
* Stateless token is server-only (#726)
* Refactor section on connection termination (#733, #748, #328,
#177)
* Limit size of Version Negotiation packet (#585)
* Clarify when and what to ack (#736)
* Renamed STREAM_ID_NEEDED to STREAM_ID_BLOCKED
* Clarify Keep-alive requirements (#729)
C.24. Since draft-ietf-quic-transport-04
* Introduce STOP_SENDING frame, RESET_STREAM only resets in one
direction (#165)
* Removed GOAWAY; application protocols are responsible for graceful
shutdown (#696)
* Reduced the number of error codes (#96, #177, #184, #211)
* Version validation fields can't move or change (#121)
* Removed versions from the transport parameters in a
NewSessionTicket message (#547)
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* Clarify the meaning of "bytes in flight" (#550)
* Public reset is now stateless reset and not visible to the path
(#215)
* Reordered bits and fields in STREAM frame (#620)
* Clarifications to the stream state machine (#572, #571)
* Increased the maximum length of the Largest Acknowledged field in
ACK frames to 64 bits (#629)
* truncate_connection_id is renamed to omit_connection_id (#659)
* CONNECTION_CLOSE terminates the connection like TCP RST (#330,
#328)
* Update labels used in HKDF-Expand-Label to match TLS 1.3 (#642)
C.25. Since draft-ietf-quic-transport-03
* Change STREAM and RESET_STREAM layout
* Add MAX_STREAM_ID settings
C.26. Since draft-ietf-quic-transport-02
* The size of the initial packet payload has a fixed minimum (#267,
#472)
* Define when Version Negotiation packets are ignored (#284, #294,
#241, #143, #474)
* The 64-bit FNV-1a algorithm is used for integrity protection of
unprotected packets (#167, #480, #481, #517)
* Rework initial packet types to change how the connection ID is
chosen (#482, #442, #493)
* No timestamps are forbidden in unprotected packets (#542, #429)
* Cryptographic handshake is now on stream 0 (#456)
* Remove congestion control exemption for cryptographic handshake
(#248, #476)
* Version 1 of QUIC uses TLS; a new version is needed to use a
different handshake protocol (#516)
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* STREAM frames have a reduced number of offset lengths (#543, #430)
* 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)
* A NEW_CONNECTION_ID frame supports connection migration without
linkability (#232, #491, #496)
* 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)
* Expanded security considerations (#440, #444, #445, #448)
C.27. Since draft-ietf-quic-transport-01
* Defined short and long packet headers (#40, #148, #361)
* Defined a versioning scheme and stable fields (#51, #361)
* Define reserved version values for "greasing" negotiation (#112,
#278)
* The initial packet number is randomized (#35, #283)
* Narrow the packet number encoding range requirement (#67, #286,
#299, #323, #356)
* Defined client address validation (#52, #118, #120, #275)
* Define transport parameters as a TLS extension (#49, #122)
* SCUP and COPT parameters are no longer valid (#116, #117)
* Transport parameters for 0-RTT are either remembered from before,
or assume default values (#126)
* The server chooses connection IDs in its final flight (#119, #349,
#361)
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* The server echoes the Connection ID and packet number fields when
sending a Version Negotiation packet (#133, #295, #244)
* Defined a minimum packet size for the initial handshake packet
from the client (#69, #136, #139, #164)
* Path MTU Discovery (#64, #106)
* The initial handshake packet from the client needs to fit in a
single packet (#338)
* Forbid acknowledgment of packets containing only ACK and PADDING
(#291)
* Require that frames are processed when packets are acknowledged
(#381, #341)
* Removed the STOP_WAITING frame (#66)
* Don't require retransmission of old timestamps for lost ACK frames
(#308)
* Clarified that frames are not retransmitted, but the information
in them can be (#157, #298)
* Error handling definitions (#335)
* Split error codes into four sections (#74)
* Forbid the use of Public Reset where CONNECTION_CLOSE is possible
(#289)
* Define packet protection rules (#336)
* Require that stream be entirely delivered or reset, including
acknowledgment of all STREAM frames or the RESET_STREAM, before it
closes (#381)
* Remove stream reservation from state machine (#174, #280)
* Only stream 1 does not contribute to connection-level flow control
(#204)
* Stream 1 counts towards the maximum concurrent stream limit (#201,
#282)
* Remove connection-level flow control exclusion for some streams
(except 1) (#246)
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* RESET_STREAM affects connection-level flow control (#162, #163)
* Flow control accounting uses the maximum data offset on each
stream, rather than bytes received (#378)
* Moved length-determining fields to the start of STREAM and ACK
(#168, #277)
* Added the ability to pad between frames (#158, #276)
* Remove error code and reason phrase from GOAWAY (#352, #355)
* GOAWAY includes a final stream number for both directions (#347)
* Error codes for RESET_STREAM and CONNECTION_CLOSE are now at a
consistent offset (#249)
* Defined priority as the responsibility of the application protocol
(#104, #303)
C.28. Since draft-ietf-quic-transport-00
* Replaced DIVERSIFICATION_NONCE flag with KEY_PHASE flag
* Defined versioning
* Reworked description of packet and frame layout
* Error code space is divided into regions for each component
* Use big endian for all numeric values
C.29. Since draft-hamilton-quic-transport-protocol-01
* Adopted as base for draft-ietf-quic-tls
* Updated authors/editors list
* Added IANA Considerations section
* Moved Contributors and Acknowledgments to appendices
Contributors
The original design and rationale behind this protocol draw
significantly from work by Jim Roskind [EARLY-DESIGN].
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The IETF QUIC Working Group received an enormous amount of support
from many people. The following people provided substantive
contributions to this document:
* Alessandro Ghedini
* Alyssa Wilk
* Antoine Delignat-Lavaud
* Brian Trammell
* Christian Huitema
* Colin Perkins
* David Schinazi
* Dmitri Tikhonov
* Eric Kinnear
* Eric Rescorla
* Gorry Fairhurst
* Ian Swett
* Igor Lubashev
* 奥 一穂 (Kazuho Oku)
* Lucas Pardue
* Magnus Westerlund
* Marten Seemann
* Martin Duke
* Mike Bishop
* Mikkel Fahnøe Jørgensen
* Mirja Kühlewind
* Nick Banks
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* Nick Harper
* Patrick McManus
* Roberto Peon
* Ryan Hamilton
* Subodh Iyengar
* Tatsuhiro Tsujikawa
* Ted Hardie
* Tom Jones
* Victor Vasiliev
Authors' Addresses
Jana Iyengar (editor)
Fastly
Email: jri.ietf@gmail.com
Martin Thomson (editor)
Mozilla
Email: mt@lowentropy.net
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