Internet Engineering Task Force (IETF) J. Iyengar, Ed.
Request for Comments: 9000 Fastly
Category: Standards Track M. Thomson, Ed.
ISSN: 2070-1721 Mozilla
May 2021
QUIC: A UDP-Based Multiplexed and Secure Transport
Abstract
This document defines the core of the QUIC transport protocol. QUIC
provides applications with flow-controlled streams for structured
communication, low-latency connection establishment, and network path
migration. QUIC includes security measures that ensure
confidentiality, integrity, and availability in a range of deployment
circumstances. Accompanying documents describe the integration of
TLS for key negotiation, loss detection, and an exemplary congestion
control algorithm.
Status of This Memo
This is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
https://www.rfc-editor.org/info/rfc9000.
Copyright Notice
Copyright (c) 2021 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(https://trustee.ietf.org/license-info) in effect on the date of
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Overview
1.1. Document Structure
1.2. Terms and Definitions
1.3. Notational Conventions
2. Streams
2.1. Stream Types and Identifiers
2.2. Sending and Receiving Data
2.3. Stream Prioritization
2.4. Operations on Streams
3. Stream States
3.1. Sending Stream States
3.2. Receiving Stream States
3.3. Permitted Frame Types
3.4. Bidirectional Stream States
3.5. Solicited State Transitions
4. Flow Control
4.1. Data Flow Control
4.2. Increasing Flow Control Limits
4.3. Flow Control Performance
4.4. Handling Stream Cancellation
4.5. Stream Final Size
4.6. Controlling Concurrency
5. Connections
5.1. Connection ID
5.1.1. Issuing Connection IDs
5.1.2. Consuming and Retiring Connection IDs
5.2. Matching Packets to Connections
5.2.1. Client Packet Handling
5.2.2. Server Packet Handling
5.2.3. Considerations for Simple Load Balancers
5.3. Operations on Connections
6. Version Negotiation
6.1. Sending Version Negotiation Packets
6.2. Handling Version Negotiation Packets
6.3. Using Reserved Versions
7. Cryptographic and Transport Handshake
7.1. Example Handshake Flows
7.2. Negotiating Connection IDs
7.3. Authenticating Connection IDs
7.4. Transport Parameters
7.4.1. Values of Transport Parameters for 0-RTT
7.4.2. New Transport Parameters
7.5. Cryptographic Message Buffering
8. Address Validation
8.1. Address Validation during Connection Establishment
8.1.1. Token Construction
8.1.2. Address Validation Using Retry Packets
8.1.3. Address Validation for Future Connections
8.1.4. Address Validation Token Integrity
8.2. Path Validation
8.2.1. Initiating Path Validation
8.2.2. Path Validation Responses
8.2.3. Successful Path Validation
8.2.4. Failed Path Validation
9. Connection Migration
9.1. Probing a New Path
9.2. Initiating Connection Migration
9.3. Responding to Connection Migration
9.3.1. Peer Address Spoofing
9.3.2. On-Path Address Spoofing
9.3.3. Off-Path Packet Forwarding
9.4. Loss Detection and Congestion Control
9.5. Privacy Implications of Connection Migration
9.6. Server's Preferred Address
9.6.1. Communicating a Preferred Address
9.6.2. Migration to a Preferred Address
9.6.3. Interaction of Client Migration and Preferred Address
9.7. Use of IPv6 Flow Label and Migration
10. Connection Termination
10.1. Idle Timeout
10.1.1. Liveness Testing
10.1.2. Deferring Idle Timeout
10.2. Immediate Close
10.2.1. Closing Connection State
10.2.2. Draining Connection State
10.2.3. Immediate Close during the Handshake
10.3. Stateless Reset
10.3.1. Detecting a Stateless Reset
10.3.2. Calculating a Stateless Reset Token
10.3.3. Looping
11. Error Handling
11.1. Connection Errors
11.2. Stream Errors
12. Packets and Frames
12.1. Protected Packets
12.2. Coalescing Packets
12.3. Packet Numbers
12.4. Frames and Frame Types
12.5. Frames and Number Spaces
13. Packetization and Reliability
13.1. Packet Processing
13.2. Generating Acknowledgments
13.2.1. Sending ACK Frames
13.2.2. Acknowledgment Frequency
13.2.3. Managing ACK Ranges
13.2.4. Limiting Ranges by Tracking ACK Frames
13.2.5. Measuring and Reporting Host Delay
13.2.6. ACK Frames and Packet Protection
13.2.7. PADDING Frames Consume Congestion Window
13.3. Retransmission of Information
13.4. Explicit Congestion Notification
13.4.1. Reporting ECN Counts
13.4.2. ECN Validation
14. Datagram Size
14.1. Initial Datagram Size
14.2. Path Maximum Transmission Unit
14.2.1. Handling of ICMP Messages by PMTUD
14.3. Datagram Packetization Layer PMTU Discovery
14.3.1. DPLPMTUD and Initial Connectivity
14.3.2. Validating the Network Path with DPLPMTUD
14.3.3. Handling of ICMP Messages by DPLPMTUD
14.4. Sending QUIC PMTU Probes
14.4.1. PMTU Probes Containing Source Connection ID
15. Versions
16. Variable-Length Integer Encoding
17. Packet Formats
17.1. Packet Number Encoding and Decoding
17.2. Long Header Packets
17.2.1. Version Negotiation Packet
17.2.2. Initial Packet
17.2.3. 0-RTT
17.2.4. Handshake Packet
17.2.5. Retry Packet
17.3. Short Header Packets
17.3.1. 1-RTT Packet
17.4. Latency Spin Bit
18. Transport Parameter Encoding
18.1. Reserved Transport Parameters
18.2. Transport Parameter Definitions
19. Frame Types and Formats
19.1. PADDING Frames
19.2. PING Frames
19.3. ACK Frames
19.3.1. ACK Ranges
19.3.2. ECN Counts
19.4. RESET_STREAM Frames
19.5. STOP_SENDING Frames
19.6. CRYPTO Frames
19.7. NEW_TOKEN Frames
19.8. STREAM Frames
19.9. MAX_DATA Frames
19.10. MAX_STREAM_DATA Frames
19.11. MAX_STREAMS Frames
19.12. DATA_BLOCKED Frames
19.13. STREAM_DATA_BLOCKED Frames
19.14. STREAMS_BLOCKED Frames
19.15. NEW_CONNECTION_ID Frames
19.16. RETIRE_CONNECTION_ID Frames
19.17. PATH_CHALLENGE Frames
19.18. PATH_RESPONSE Frames
19.19. CONNECTION_CLOSE Frames
19.20. HANDSHAKE_DONE Frames
19.21. Extension Frames
20. Error Codes
20.1. Transport Error Codes
20.2. Application Protocol Error Codes
21. Security Considerations
21.1. Overview of Security Properties
21.1.1. Handshake
21.1.2. Protected Packets
21.1.3. Connection Migration
21.2. Handshake Denial of Service
21.3. Amplification Attack
21.4. Optimistic ACK Attack
21.5. Request Forgery Attacks
21.5.1. Control Options for Endpoints
21.5.2. Request Forgery with Client Initial Packets
21.5.3. Request Forgery with Preferred Addresses
21.5.4. Request Forgery with Spoofed Migration
21.5.5. Request Forgery with Version Negotiation
21.5.6. Generic Request Forgery Countermeasures
21.6. Slowloris Attacks
21.7. Stream Fragmentation and Reassembly Attacks
21.8. Stream Commitment Attack
21.9. Peer Denial of Service
21.10. Explicit Congestion Notification Attacks
21.11. Stateless Reset Oracle
21.12. Version Downgrade
21.13. Targeted Attacks by Routing
21.14. Traffic Analysis
22. IANA Considerations
22.1. Registration Policies for QUIC Registries
22.1.1. Provisional Registrations
22.1.2. Selecting Codepoints
22.1.3. Reclaiming Provisional Codepoints
22.1.4. Permanent Registrations
22.2. QUIC Versions Registry
22.3. QUIC Transport Parameters Registry
22.4. QUIC Frame Types Registry
22.5. QUIC Transport Error Codes Registry
23. References
23.1. Normative References
23.2. Informative References
Appendix A. Pseudocode
A.1. Sample Variable-Length Integer Decoding
A.2. Sample Packet Number Encoding Algorithm
A.3. Sample Packet Number Decoding Algorithm
A.4. Sample ECN Validation Algorithm
Contributors
Authors' Addresses
1. Overview
QUIC is a secure general-purpose transport protocol. This document
defines version 1 of QUIC, which conforms to the version-independent
properties of QUIC defined in [QUIC-INVARIANTS].
QUIC is a connection-oriented protocol that creates a stateful
interaction between a client and server.
The QUIC handshake combines negotiation of cryptographic and
transport parameters. QUIC integrates the TLS handshake [TLS13],
although using a customized framing for protecting packets. The
integration of TLS and QUIC is described in more detail in
[QUIC-TLS]. The handshake is structured to permit the exchange of
application data as soon as possible. This includes an option for
clients to send data immediately (0-RTT), which requires some form of
prior communication or configuration to enable.
Endpoints communicate in QUIC by exchanging QUIC packets. Most
packets contain frames, which carry control information and
application data between endpoints. QUIC authenticates the entirety
of each packet and encrypts as much of each packet as is practical.
QUIC packets are carried in UDP datagrams [UDP] to better facilitate
deployment in existing systems and networks.
Application protocols exchange information over a QUIC connection via
streams, which are ordered sequences of bytes. Two types of streams
can be created: bidirectional streams, which allow both endpoints to
send data; and unidirectional streams, which allow a single endpoint
to send data. A credit-based scheme is used to limit stream creation
and to bound the amount of data that can be sent.
QUIC provides the necessary feedback to implement reliable delivery
and congestion control. An algorithm for detecting and recovering
from loss of data is described in Section 6 of [QUIC-RECOVERY]. QUIC
depends on congestion control to avoid network congestion. An
exemplary congestion control algorithm is described in Section 7 of
[QUIC-RECOVERY].
QUIC connections are not strictly bound to a single network path.
Connection migration uses connection identifiers to allow connections
to transfer to a new network path. Only clients are able to migrate
in this version of QUIC. This design also allows connections to
continue after changes in network topology or address mappings, such
as might be caused by NAT rebinding.
Once established, multiple options are provided for connection
termination. Applications can manage a graceful shutdown, endpoints
can negotiate a timeout period, errors can cause immediate connection
teardown, and a stateless mechanism provides for termination of
connections after one endpoint has lost state.
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 describes address validation and critical denial-of-
service mitigations,
- 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 guidance for stream and connection 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 acknowledgment of data, and
- Section 14 specifies rules for managing the size of datagrams
carrying QUIC packets.
* Finally, encoding details of QUIC protocol elements are described
in:
- Section 15 (versions),
- Section 16 (integer encoding),
- 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 and other cryptographic
mechanisms [QUIC-TLS].
This document defines QUIC version 1, which conforms to the protocol
invariants in [QUIC-INVARIANTS].
To refer to QUIC version 1, cite this document. References to the
limited set of version-independent properties of QUIC can cite
[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 this document are described below.
QUIC: The transport protocol described by this document. QUIC is a
name, not an acronym.
Endpoint: An entity that can participate in a QUIC connection by
generating, receiving, and processing QUIC packets. There are
only two types of endpoints in QUIC: client and server.
Client: The endpoint that initiates a QUIC connection.
Server: The endpoint that accepts a QUIC connection.
QUIC packet: A complete processable unit of QUIC that can be
encapsulated in a UDP datagram. One or more 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.
Frame: A unit of structured protocol information. There are
multiple frame types, each of which carries different information.
Frames are contained in QUIC packets.
Address: When used without qualification, the tuple of IP version,
IP address, and UDP port number that represents one end of a
network path.
Connection ID: An identifier that is used to identify a QUIC
connection at an endpoint. Each endpoint selects one or more
connection IDs for its peer to include in packets sent towards the
endpoint. This value is opaque to the peer.
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.
This document uses the terms "QUIC packets", "UDP datagrams", and "IP
packets" to refer to the units of the respective protocols. That is,
one or more QUIC packets can be encapsulated in a UDP datagram, which
is in turn encapsulated in an IP packet.
1.3. Notational Conventions
Packet and frame diagrams in this document use a custom 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 holds an integer value using the variable-
length encoding described 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 a byte boundary
x (L) = C: Indicates that x has a fixed value of C; the length of x
is described by L, which can use any of the length forms above
x (L) = C..D: Indicates that x has a value in the range from C to D,
inclusive, with the length described by L, as above
[x (L)]: Indicates that x is optional and has a length of L
x (L) ...: Indicates that x is repeated zero or more times and that
each instance has a length of L
This document uses network byte order (that is, big endian) values.
Fields are placed starting from the high-order bits of each byte.
By convention, individual fields reference a complex field by using
the name of the complex field.
Figure 1 provides an example:
Example Structure {
One-bit Field (1),
7-bit Field with Fixed Value (7) = 61,
Field with Variable-Length Integer (i),
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
When a single-bit field is referenced in prose, the position of that
field can be clarified by using the value of the byte that carries
the field with the field's value set. For example, the value 0x80
could be used to refer to the single-bit field in the most
significant bit of the byte, such as One-bit Field in Figure 1.
2. Streams
Streams in QUIC provide a lightweight, ordered byte-stream
abstraction to an application. Streams can be unidirectional or
bidirectional.
Streams can be created by sending data. Other processes associated
with stream management -- ending, canceling, 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 canceled. 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.
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 (0x01) 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 (0x02) of the stream ID
distinguishes between bidirectional streams (with the bit set to 0)
and unidirectional streams (with the bit set to 1).
The two least significant bits from a stream ID therefore identify a
stream as one of four types, as summarized in Table 1.
+======+==================================+
| Bits | Stream Type |
+======+==================================+
| 0x00 | Client-Initiated, Bidirectional |
+------+----------------------------------+
| 0x01 | Server-Initiated, Bidirectional |
+------+----------------------------------+
| 0x02 | Client-Initiated, Unidirectional |
+------+----------------------------------+
| 0x03 | Server-Initiated, Unidirectional |
+------+----------------------------------+
Table 1: Stream ID Types
The stream space for each type begins at the minimum value (0x00
through 0x03, respectively); successive streams of each type 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.
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.
A QUIC implementation SHOULD provide ways in which an application can
indicate the relative priority of streams. An implementation uses
information provided by the application to determine how to allocate
resources to active streams.
2.4. Operations on Streams
This document does not define an API for QUIC; it instead defines a
set of functions on streams that application protocols can rely upon.
An application protocol can assume that a QUIC implementation
provides an interface that includes the operations described in this
section. An implementation designed for use with a specific
application protocol might provide only those operations that are
used by that protocol.
On the sending part of a stream, an application protocol can:
* 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, an application protocol can:
* read data; and
* abort reading of the stream and request closure, possibly
resulting in a STOP_SENDING frame (Section 19.5).
An application protocol can also request 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).
Unidirectional streams use either the sending or receiving state
machine, depending on the stream type and endpoint role.
Bidirectional streams use both state machines at both endpoints. 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 the send or receive side causes the
stream to open in both directions.
The state machines shown in this section 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 are not 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.
| Note: In some cases, a single event or action can cause a
| transition through multiple states. For instance, sending
| STREAM with a FIN bit set can cause two state transitions for a
| sending stream: from the "Ready" state to the "Send" state, and
| from the "Send" state to the "Data Sent" state.
3.1. Sending Stream States
Figure 2 shows the states for the part of a stream that sends data to
a peer.
o
| Create Stream (Sending)
| Peer Creates Bidirectional Stream
v
+-------+
| Ready | Send RESET_STREAM
| |-----------------------.
+-------+ |
| |
| Send STREAM / |
| STREAM_DATA_BLOCKED |
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 a 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.
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 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 state that is one of "Ready", "Send", or "Data Sent", 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.
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 Reset
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 frame is received for that
stream. For bidirectional streams initiated by a peer, receipt of a
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 a stream is "Recv".
For a bidirectional stream, the receiving part enters the "Recv"
state when the sending part 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 4.5. 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" state
causes the stream to enter the "Reset Recvd" state. This might cause
the delivery of stream data to the application to be interrupted.
It is possible that all stream data has already been received when a
RESET_STREAM is received (that is, in 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 a 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 the sender or the 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 a STREAM or
STREAM_DATA_BLOCKED frame for a stream in the "Reset Sent" state or
any terminal state -- that is, after sending a RESET_STREAM frame. 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 frames (Section 19.10)
and STOP_SENDING frames (Section 19.5).
The receiver only sends MAX_STREAM_DATA frames in the "Recv" state.
A receiver MAY send a STOP_SENDING frame 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, as all stream
data has been received. A sender could receive either of these two
types of frames in any state as a result of delayed delivery of
packets.
3.4. Bidirectional Stream States
A bidirectional stream is composed of sending and receiving parts.
Implementations can 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 defined 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 be acknowledged before the transition to a
"closed" or "half-closed" state.
+===================+=======================+=================+
| Sending Part | Receiving Part | Composite State |
+===================+=======================+=================+
| No Stream / Ready | No Stream / Recv (*1) | idle |
+-------------------+-----------------------+-----------------+
| Ready / Send / | Recv / Size Known | open |
| Data Sent | | |
+-------------------+-----------------------+-----------------+
| Ready / Send / | Data Recvd / Data | half-closed |
| Data Sent | Read | (remote) |
+-------------------+-----------------------+-----------------+
| Ready / Send / | Reset Recvd / Reset | half-closed |
| Data Sent | Read | (remote) |
+-------------------+-----------------------+-----------------+
| Data Recvd | Recv / Size Known | half-closed |
| | | (local) |
+-------------------+-----------------------+-----------------+
| Reset Sent / | Recv / Size Known | half-closed |
| Reset Recvd | | (local) |
+-------------------+-----------------------+-----------------+
| Reset Sent / | Data Recvd / Data | closed |
| Reset Recvd | Read | |
+-------------------+-----------------------+-----------------+
| Reset Sent / | Reset Recvd / Reset | closed |
| Reset Recvd | Read | |
+-------------------+-----------------------+-----------------+
| Data Recvd | Data Recvd / Data | closed |
| | Read | |
+-------------------+-----------------------+-----------------+
| 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.
If the stream is in the "Recv" or "Size Known" state, 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 a STOP_SENDING frame 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, the
endpoint MAY defer sending the RESET_STREAM frame until the packets
containing outstanding data are acknowledged or declared lost. If
any outstanding data is declared lost, the endpoint SHOULD send a
RESET_STREAM frame instead of retransmitting the data.
An endpoint SHOULD copy the error code from the STOP_SENDING frame to
the RESET_STREAM frame it sends, but it can use any application error
code. An endpoint that sends a STOP_SENDING frame MAY ignore the
error code in any RESET_STREAM frames subsequently received for that
stream.
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" state.
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 frame, and it can encourage prompt termination in the
opposite direction by sending a STOP_SENDING frame.
4. Flow Control
Receivers need to limit the amount of data that they are required to
buffer, in order to prevent a fast sender from overwhelming them or a
malicious sender from consuming a large amount of memory. To enable
a receiver to limit memory commitments for a connection, streams are
flow controlled both individually and across a connection as a whole.
A QUIC receiver controls the maximum amount of data the sender can
send on a stream as well as across all streams at any time, as
described in Sections 4.1 and 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 4.6.
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].
To avoid excessive buffering at multiple layers, QUIC implementations
SHOULD provide an interface for the cryptographic protocol
implementation to communicate its buffering limits.
4.1. Data Flow Control
QUIC employs a limit-based flow control scheme where a receiver
advertises the limit of total bytes it is prepared to receive on a
given stream or 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 each 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.
Senders MUST NOT send data in excess of either limit.
A receiver sets initial limits for all streams through transport
parameters during the handshake (Section 7.4). Subsequently, a
receiver sends MAX_STREAM_DATA frames (Section 19.10) or MAX_DATA
frames (Section 19.9) to the sender to advertise larger limits.
A receiver can advertise a larger limit for a stream by sending a
MAX_STREAM_DATA frame with the corresponding stream ID. A
MAX_STREAM_DATA frame indicates the maximum absolute byte offset of a
stream. A receiver could determine the flow control offset to be
advertised based on the current offset of data consumed on that
stream.
A receiver can advertise a larger limit 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 violations of the advertised connection or stream data
limits. A receiver could determine the maximum data limit to be
advertised based on the sum of bytes consumed on all streams.
Once a receiver advertises a limit for the connection or a stream, it
is not an error to advertise a smaller limit, but the smaller limit
has no effect.
A receiver MUST close the connection with an error of type
FLOW_CONTROL_ERROR if the sender violates the advertised connection
or stream data limits; see Section 11 for details on error handling.
A sender MUST ignore any MAX_STREAM_DATA or MAX_DATA frames that do
not increase flow control limits.
If a sender has sent data up to the limit, 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 to the receiver
that 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 10.1), the receiver might close the connection even when the
sender has data that 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. Increasing Flow Control Limits
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 MAY send a MAX_STREAM_DATA or
MAX_DATA frame multiple times within a round trip or send it early
enough to allow time for loss of the frame and subsequent recovery.
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, 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 7.7 of [QUIC-RECOVERY] for a discussion of
how a sender can avoid this congestion.
4.3. Flow Control Performance
If an endpoint cannot ensure that its peer always has available flow
control credit that is greater than the peer's bandwidth-delay
product on this connection, its receive throughput will be limited by
flow control.
Packet loss can cause gaps in the receive buffer, preventing the
application from consuming data and freeing up receive buffer space.
Sending timely updates of flow control limits can improve
performance. Sending packets only to provide flow control updates
can increase network load and adversely affect performance. Sending
flow control updates along with other frames, such as ACK frames,
reduces the cost of those updates.
4.4. Handling Stream Cancellation
Endpoints need to eventually agree on the amount of flow control
credit that has been consumed on every stream, to be able to account
for all bytes for connection-level flow control.
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.
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.
4.5. 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.
A sender always communicates the final size of a stream to the
receiver reliably, no matter how the stream is terminated. The final
size is the sum of the Offset and Length fields of a STREAM frame
with a FIN flag, noting that these fields might be implicit.
Alternatively, the Final Size field of a RESET_STREAM frame carries
this value. This guarantees that both endpoints agree on how much
flow control credit was consumed by the sender on that stream.
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). 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.
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 an error
of type FINAL_SIZE_ERROR; see Section 11 for details on error
handling. A receiver SHOULD treat receipt of data at or beyond the
final size as an error of type FINAL_SIZE_ERROR, even after a stream
is closed. Generating these errors is not mandatory, 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.
4.6. Controlling Concurrency
An endpoint limits the cumulative number of incoming streams a peer
can open. Only streams with a stream ID less than "(max_streams * 4
+ first_stream_id_of_type)" can be opened; see Table 1. Initial
limits are set in the transport parameters; see Section 18.2.
Subsequent limits are advertised using MAX_STREAMS frames; see
Section 19.11. Separate limits apply to unidirectional and
bidirectional streams.
If a max_streams transport parameter or a 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 TRANSPORT_PARAMETER_ERROR
if the offending value was received in a transport parameter or of
type FRAME_ENCODING_ERROR if it was received in a frame; see
Section 10.2.
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; see Section 11 for details on error handling.
Once a receiver advertises a stream limit using the MAX_STREAMS
frame, advertising a smaller limit has no effect. MAX_STREAMS frames
that do not increase the stream limit MUST be ignored.
As with stream and connection flow control, this document leaves
implementations to decide when and how many streams should be
advertised to a peer via MAX_STREAMS. Implementations might choose
to increase limits as streams are closed, 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 indefinitely if the peer chooses
not to send STREAMS_BLOCKED frames.
5. Connections
A QUIC connection is shared state between a client and a server.
Each connection starts with a handshake phase, during which the two
endpoints establish a shared secret using the cryptographic handshake
protocol [QUIC-TLS] and negotiate the application protocol. 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 use the connection during the handshake
phase with some limitations. 0-RTT allows application data to be
sent by a client before receiving a response from the server.
However, 0-RTT provides no protection against replay attacks; see
Section 9.2 of [QUIC-TLS]. 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 of trading 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, 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.
The primary function of a connection ID is to ensure that changes in
addressing at lower protocol layers (UDP, IP) do not 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 that will allow packets with that
connection ID to be routed back to the endpoint and to be identified
by the endpoint upon receipt.
Multiple connection IDs are used so that endpoints can send packets
that cannot be identified by an observer as being for the same
connection without cooperation from an endpoint; see Section 9.5.
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 demonstrate that the packet is in response to a packet
sent by the client.
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. An
endpoint MUST NOT use the same IP address and port for multiple
concurrent connections with zero-length connection IDs, unless it 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).
5.1.1. Issuing Connection IDs
Each connection ID has an associated sequence number to assist in
detecting when NEW_CONNECTION_ID or RETIRE_CONNECTION_ID frames refer
to the same value. 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 that a client selects for the first Destination Connection ID
field it sends and any connection ID provided by a Retry packet are
not assigned sequence numbers.
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
endpoint MAY limit the total number of connection IDs issued for each
connection to avoid the risk of running out of connection IDs; see
Section 10.3.2. 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 be unable to respond if the pool is
exhausted.
An endpoint that selects a zero-length connection ID during the
handshake cannot issue a new connection ID. A zero-length
Destination Connection ID field is used in all packets sent toward
such an endpoint over any network path.
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 details.
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
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 for which RETIRE_CONNECTION_ID frames 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 value of
the active_connection_id_limit transport parameter. 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 10.3) 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.
Invalid packets that lack strong integrity protection, such as
Initial, Retry, or Version Negotiation, MAY be discarded. An
endpoint MUST generate a connection error if processing the contents
of these packets prior to discovering an error, or fully revert any
changes made during that processing.
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 do not match an existing
connection -- based on Destination Connection ID or, if this value is
zero length, local IP address and port -- 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 it MAY buffer them in
anticipation of later packets that allow it to compute the key.
If a client receives a packet that uses a different version than it
initially selected, it MUST discard that packet.
5.2.2. Server Packet Handling
If a server receives a packet that indicates an unsupported version
and if the packet is large enough to initiate a new connection for
any supported version, the server SHOULD send a Version Negotiation
packet as described in Section 6.1. A server MAY limit the number of
packets to which it responds with a Version Negotiation packet.
Servers MUST drop smaller 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 or
properly interpret the result. Servers SHOULD respond with a Version
Negotiation packet, provided that the datagram 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. These packets
are processed using the selected connection; otherwise, the server
continues as described 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.
If a server refuses to accept a new connection, it SHOULD send an
Initial packet containing a CONNECTION_CLOSE frame with error code
CONNECTION_REFUSED.
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 when the client address changes SHOULD
indicate that migration is not supported by using the
disable_active_migration transport parameter. The
disable_active_migration transport parameter does not prohibit
connection migration after a client has acted on a preferred_address
transport parameter.
Server deployments that use this simple form of load balancing MUST
avoid the creation of a stateless reset oracle; see Section 21.11.
5.3. Operations on Connections
This document does not define an API for QUIC; it instead defines a
set of functions for QUIC connections that application protocols can
rely upon. An application protocol can assume that an implementation
of QUIC provides an interface that includes the operations described
in this section. An implementation designed for use with a specific
application protocol might provide only those operations that are
used by that protocol.
When implementing the client role, an application protocol can:
* open a connection, which begins the exchange described in
Section 7;
* enable Early Data when available; and
* be informed when Early Data has been accepted or rejected by a
server.
When implementing the server role, an application protocol can:
* listen for incoming connections, which prepares for the exchange
described in Section 7;
* 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 accept or reject Early
Data based on that information.
In either role, an application protocol can:
* 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 for receive buffers by setting flow
control limits both for streams and for the connection;
* identify whether the handshake has completed successfully or is
still ongoing;
* keep a connection from silently closing, by either generating PING
frames (Section 19.2) or requesting that the transport send
additional frames before the idle timeout expires (Section 10.1);
and
* immediately close (Section 10.2) the connection.
6. Version Negotiation
Version negotiation allows a server to indicate that it does not
support the version the client used. 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 ensure that the first UDP datagram they
send is sized to the largest of the minimum datagram sizes from all
versions they support, using PADDING frames (Section 19.1) as
necessary. This ensures that the server responds if there is a
mutually supported version. A server might not send a Version
Negotiation packet if the datagram it receives is smaller than the
minimum size specified in a different version; see Section 14.1.
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.
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.
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 for functionality
to be defined in the future that allows QUIC to negotiate the version
of QUIC to use for a connection. Future Standards Track
specifications might change how implementations that support multiple
versions of QUIC react to Version Negotiation packets received in
response to an attempt 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 Standards Track specifications. In particular, that future
work will ensure robustness against version downgrade attacks; see
Section 21.12.
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. The
version of QUIC defined in this document is identified as 0x00000001
and uses TLS as described in [QUIC-TLS]; a different QUIC version
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, and
- keying material is usable for packet protection for both 0-RTT
and 1-RTT packets.
* authenticated exchange of 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
Application-Layer Protocol Negotiation (ALPN) [ALPN] for this
purpose).
The CRYPTO frame can be sent in different packet number spaces
(Section 12.3). The offsets used by CRYPTO frames to ensure ordered
delivery of cryptographic handshake data start from zero in each
packet number space.
Figure 4 shows a simplified handshake and the exchange of packets and
frames that are used to advance the handshake. Exchange of
application data during the handshake is enabled where possible,
shown with an asterisk ("*"). Once the handshake is complete,
endpoints are able to exchange application data freely.
Client Server
Initial (CRYPTO)
0-RTT (*) ---------->
Initial (CRYPTO)
Handshake (CRYPTO)
<---------- 1-RTT (*)
Handshake (CRYPTO)
1-RTT (*) ---------->
<---------- 1-RTT (HANDSHAKE_DONE)
1-RTT <=========> 1-RTT
Figure 4: Simplified QUIC Handshake
Endpoints can use packets sent during the handshake to test for
Explicit Congestion Notification (ECN) support; see Section 13.4. An
endpoint validates support for ECN by observing whether the ACK
frames acknowledging the first packets it sends carry ECN counts, as
described in Section 13.4.2.
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 5 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. For instance, the first packet is of type Initial, with
packet number 0, and contains a CRYPTO frame carrying the
ClientHello.
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 could consist of as few as four UDP datagrams, or any
number more (subject to limits inherent to the protocol, such as
congestion control and anti-amplification). For instance, the
server's first flight contains Initial packets, Handshake packets,
and "0.5-RTT data" in 1-RTT packets.
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]: HANDSHAKE_DONE, STREAM[3, "..."], ACK[0]
Figure 5: Example 1-RTT Handshake
Figure 6 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]: HANDSHAKE_DONE, STREAM[3, "..."], ACK[1]
Figure 6: 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.
During the handshake, packets with the long header (Section 17.2) are
used to establish the connection IDs used by both endpoints. 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. After processing the first Initial
packet, each endpoint sets the Destination Connection ID field in
subsequent packets it sends to the value of the Source Connection ID
field that it received.
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.
The Destination Connection ID field from the first Initial packet
sent by a client is used to determine packet protection keys for
Initial packets. These keys change after receiving a Retry packet;
see Section 5.2 of [QUIC-TLS].
The client populates the Source Connection ID field with a value of
its choosing and sets the Source Connection ID Length field to
indicate the length.
0-RTT packets in the first flight 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 packet
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 on that connection 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 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.
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 (and received,
for servers). Endpoints MUST validate that received transport
parameters match received connection ID values. Including connection
ID values in transport parameters and verifying them ensures 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 the absence of the
initial_source_connection_id transport parameter from either endpoint
or the absence of the original_destination_connection_id transport
parameter from the server as a connection error of type
TRANSPORT_PARAMETER_ERROR.
An endpoint MUST treat the following as a connection error of type
TRANSPORT_PARAMETER_ERROR or PROTOCOL_VIOLATION:
* 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.
If a zero-length connection ID is selected, the corresponding
transport parameter is included with a zero-length value.
Figure 7 shows the connection IDs (with DCID=Destination Connection
ID, SCID=Source Connection ID) 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 7: Use of Connection IDs in a Handshake
Figure 8 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 8: Use of Connection IDs in a Handshake with Retry
In both cases (Figures 7 and 8), the client sets the value of the
initial_source_connection_id transport parameter to "C1".
When the handshake does not include a Retry (Figure 7), the server
sets original_destination_connection_id to "S1" (note that this value
is chosen by the client) and initial_source_connection_id to "S3".
In this case, the server does not include a
retry_source_connection_id transport parameter.
When the handshake includes a Retry (Figure 8), the server sets
original_destination_connection_id to "S1",
retry_source_connection_id to "S2", and initial_source_connection_id
to "S3".
7.4. Transport Parameters
During connection establishment, both endpoints make authenticated
declarations of their transport parameters. Endpoints are required
to comply with the restrictions that each parameter defines; 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
values 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.
ALPN (see [ALPN]) allows clients to offer multiple application
protocols during connection establishment. The transport parameters
that a client includes during the handshake apply to all application
protocols that the client offers. Application protocols can
recommend values for transport parameters, such as the initial flow
control limits. However, application protocols that set constraints
on values for transport parameters could make it impossible for a
client to offer multiple application protocols if these constraints
conflict.
7.4.1. Values of Transport Parameters for 0-RTT
Using 0-RTT depends on both client and server using protocol
parameters that were negotiated from a previous connection. To
enable 0-RTT, endpoints store the values of the server transport
parameters with any session tickets it receives on the connection.
Endpoints also store any information required by the application
protocol or cryptographic handshake; see Section 4.6 of [QUIC-TLS].
The values of stored transport parameters are used when attempting
0-RTT using the session tickets.
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. Not all transport
parameters are remembered, as some do not apply to future connections
or they have no effect on the use of 0-RTT.
The definition of a new transport parameter (Section 7.4.2) MUST
specify whether storing the transport parameter for 0-RTT is
mandatory, optional, or prohibited. A client need not store a
transport parameter it cannot process.
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; if the
server does not provide new values, the default values are used.
A client that attempts to send 0-RTT data MUST remember all other
transport parameters used by the server that it is able to process.
The server can remember these transport parameters or can 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 values 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 the sending of application
data SHOULD be set to non-zero values for 0-RTT. This includes
initial_max_data and either (1) initial_max_streams_bidi and
initial_max_stream_data_bidi_remote or (2) initial_max_streams_uni
and initial_max_stream_data_uni.
A server might provide larger initial stream flow control limits for
streams than the remembered values that a client applies when sending
0-RTT. Once the handshake completes, the client updates the flow
control limits on all sending streams using the updated values of
initial_max_stream_data_bidi_remote and initial_max_stream_data_uni.
A server MAY store and recover the previously sent values of the
max_idle_timeout, max_udp_payload_size, and disable_active_migration
parameters and reject 0-RTT if it selects smaller values. Lowering
the values of these parameters while also accepting 0-RTT data could
degrade the performance of the connection. Specifically, lowering
the max_udp_payload_size could result in dropped packets, leading to
worse performance compared to rejecting 0-RTT data outright.
A server MUST reject 0-RTT data if the restored 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
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 the 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. The 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.
A client that does not understand a transport parameter can discard
it and attempt 0-RTT on subsequent connections. However, if the
client adds support for a discarded transport parameter, it risks
violating the constraints that the transport parameter establishes if
it attempts 0-RTT. New transport parameters can avoid this problem
by setting a default of the most conservative value. Clients can
avoid this problem by remembering all parameters, even those not
currently supported.
New transport parameters can be registered according to the rules in
Section 22.3.
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 out-of-order CRYPTO frames. 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.
8. Address Validation
Address validation ensures that an endpoint cannot be 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 attacks is verifying that a
peer is able to receive packets at the transport address that it
claims. Therefore, after receiving packets from an address that is
not yet validated, an endpoint MUST limit the amount of data it sends
to the unvalidated address to three times the amount of data received
from that address. This limit on the size of responses is known as
the anti-amplification limit.
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 peer successfully processed an
Initial packet. Once an endpoint has successfully processed a
Handshake packet from the peer, it can consider the peer address to
have been validated.
Additionally, an endpoint MAY consider the peer address validated if
the peer uses a connection ID chosen by the endpoint and the
connection ID contains at least 64 bits of entropy.
For the client, the value of the Destination Connection ID field in
its first Initial packet allows it to validate the server address as
a part of successfully processing any packet. Initial packets from
the server are protected with keys that are derived from this value
(see Section 5.2 of [QUIC-TLS]). Alternatively, the value is echoed
by the server in Version Negotiation packets (Section 6) or included
in the Integrity Tag in Retry packets (Section 5.8 of [QUIC-TLS]).
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 frames as
necessary. A client that sends padded datagrams allows the server to
send more data prior to completing address validation.
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 acknowledgments 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
(PTO); see Section 6.2 of [QUIC-RECOVERY]. Specifically, the client
MUST send an Initial packet in a UDP datagram that contains 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 frame 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 packet containing a token that
was provided in a Retry packet, a server cannot send another Retry
packet; it can only refuse 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.
If a server receives a client Initial that contains an invalid Retry
token but is otherwise valid, 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 10.2) 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 9.
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 9: 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. In a future connection, the client includes this
token in Initial packets to provide address validation. 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.
Unlike the token that is created for a Retry packet, which is used
immediately, the token sent in the NEW_TOKEN frame can 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. For example, it cannot include the previous
connection ID or addressing information, unless the values are
encrypted. 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 can
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 token from a NEW_TOKEN frame for
different connection attempts. Reusing a token allows connections to
be linked by entities on the network path; see Section 9.5.
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 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 have no negative consequences,
clients can regard older tokens as being less likely to 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 packet. Tokens
provided with NEW_TOKEN frames and Retry packets can be distinguished
by servers (see Section 8.1.1), and the latter can be validated more
strictly. 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.
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.
8.1.4. Address Validation Token Integrity
An address validation token MUST be difficult to guess. Including a
random value with at least 128 bits of entropy 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 remain 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
frames 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 limit; see Section 8.
Note that in the presence of NAT, this requirement might be
insufficient to protect other hosts that share the NAT from
amplification attacks.
Attackers could replay tokens to use servers as amplifiers in DDoS
attacks. To protect against such attacks, servers MUST ensure that
replay of tokens is prevented or limited. Servers SHOULD ensure that
tokens sent in Retry packets are only accepted for a short time, as
they are returned immediately by clients. Tokens that are provided
in NEW_TOKEN frames (Section 19.7) need to be valid for longer but
SHOULD NOT be accepted multiple times. Servers are encouraged to
allow tokens to be used only once, if possible; tokens MAY include
additional information about clients to further narrow applicability
or reuse.
8.2. Path Validation
Path validation is used by both peers during connection migration
(see Section 9) to verify reachability after a change of address. In
path validation, endpoints test reachability between a specific local
address and a specific peer address, where an address is the 2-tuple
of IP address and port.
Path validation tests that packets sent on a path to a peer are
received by that peer. Path validation is used to ensure that
packets received from a migrating peer do not carry a spoofed source
address.
Path validation does not validate that a peer can send in the return
direction. Acknowledgments cannot be used for return path validation
because they contain insufficient entropy and might be spoofed.
Endpoints independently determine reachability on each direction of a
path, and therefore return reachability can only be established by
the peer.
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
endpoint 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 include other frames with the PATH_CHALLENGE and
PATH_RESPONSE frames used for path validation. In particular, an
endpoint can include PADDING frames with a PATH_CHALLENGE frame for
Path Maximum Transmission Unit Discovery (PMTUD); see Section 14.2.1.
An endpoint can also include its own PATH_CHALLENGE frame when
sending a PATH_RESPONSE frame.
An endpoint uses a new connection ID for probes sent from a new local
address; see Section 9.5. When probing a new path, an endpoint can
ensure that its peer has an unused connection ID available for
responses. Sending NEW_CONNECTION_ID and PATH_CHALLENGE frames in
the same packet, if the peer's active_connection_id_limit permits,
ensures that an unused connection ID will be available to the peer
when sending a response.
An endpoint can choose to simultaneously probe multiple paths. The
number of simultaneous paths used for probes is limited by the number
of extra connection IDs its peer has previously supplied, since each
new local address used for a probe requires a previously unused
connection ID.
8.2.1. Initiating Path Validation
To initiate path validation, an endpoint sends a PATH_CHALLENGE frame
containing an unpredictable 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 probe a new path with packets containing a
PATH_CHALLENGE frame more frequently than it would send an Initial
packet. This ensures 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.
An endpoint MUST expand datagrams that contain a PATH_CHALLENGE frame
to at least the smallest allowed maximum datagram size of 1200 bytes,
unless the anti-amplification limit for the path does not permit
sending a datagram of this size. Sending UDP datagrams of this size
ensures that the network path from the endpoint to the peer can be
used for QUIC; see Section 14.
When an endpoint is unable to expand the datagram size to 1200 bytes
due to the anti-amplification limit, the path MTU will not be
validated. To ensure that the path MTU is large enough, the endpoint
MUST perform a second path validation by sending a PATH_CHALLENGE
frame in a datagram of at least 1200 bytes. This additional
validation can be performed after a PATH_RESPONSE is successfully
received or when enough bytes have been received on the path that
sending the larger datagram will not result in exceeding the anti-
amplification limit.
Unlike other cases where datagrams are expanded, endpoints MUST NOT
discard datagrams that appear to be too small when they contain
PATH_CHALLENGE or PATH_RESPONSE.
8.2.2. Path Validation Responses
On receiving a PATH_CHALLENGE frame, an endpoint MUST respond by
echoing the data contained in the PATH_CHALLENGE frame in a
PATH_RESPONSE frame. An endpoint MUST NOT delay transmission of a
packet containing a PATH_RESPONSE frame unless constrained by
congestion control.
A PATH_RESPONSE frame MUST be sent on the network path where the
PATH_CHALLENGE frame was received. This ensures that path validation
by a peer only succeeds if the path is functional in both directions.
This requirement MUST NOT be enforced by the endpoint that initiates
path validation, as that would enable an attack on migration; see
Section 9.3.3.
An endpoint MUST expand datagrams that contain a PATH_RESPONSE frame
to at least the smallest allowed maximum datagram size of 1200 bytes.
This verifies that the path is able to carry datagrams of this size
in both directions. However, an endpoint MUST NOT expand the
datagram containing the PATH_RESPONSE if the resulting data exceeds
the anti-amplification limit. This is expected to only occur if the
received PATH_CHALLENGE was not sent in an expanded datagram.
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.
8.2.3. Successful Path Validation
Path validation succeeds when a PATH_RESPONSE frame is received that
contains the data that was sent in a previous PATH_CHALLENGE frame.
A PATH_RESPONSE frame received on any network path validates the path
on which the PATH_CHALLENGE was sent.
If an endpoint sends a PATH_CHALLENGE frame in a datagram that is not
expanded to at least 1200 bytes and if the response to it validates
the peer address, the path is validated but not the path MTU. As a
result, the endpoint can now send more than three times the amount of
data that has been received. However, the endpoint MUST initiate
another path validation with an expanded datagram to verify that the
path supports the required MTU.
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.
8.2.4. 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 PTO or the PTO for the new path
(using kInitialRtt, as defined in [QUIC-RECOVERY]) is RECOMMENDED.
This timeout allows for multiple PTOs to expire prior to failing path
validation, so that loss of a single PATH_CHALLENGE or PATH_RESPONSE
frame does not cause path validation failure.
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. An endpoint
that has no valid network path to its peer MAY signal this using the
NO_VIABLE_PATH connection error, noting that this is only possible if
the network path exists but does not support the required MTU
(Section 14).
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.
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].
If the peer sent the disable_active_migration transport parameter, an
endpoint also MUST NOT send packets (including probing packets; see
Section 9.1) from a different local address to the address the peer
used during the handshake, unless the endpoint has acted on a
preferred_address transport parameter from the peer. If the peer
violates this requirement, the endpoint 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.
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.
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.
To establish reachability on the new path, an endpoint initiates path
validation (Section 8.2) on the new path. An endpoint MAY defer path
validation until after a peer sends the next non-probing frame to its
new address.
When migrating, the new path might not support the endpoint's current
sending rate. Therefore, the endpoint resets its congestion
controller and RTT estimate, as described in Section 9.4.
The new path might not have the same ECN capability. Therefore, the
endpoint validates ECN capability as described in Section 13.4.
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.
If the recipient permits the migration, it MUST send subsequent
packets to the new peer address and MUST initiate path validation
(Section 8.2) to verify the peer's ownership of the address if
validation is not already underway. If the recipient has no unused
connection IDs from the peer, it will not be able to send anything on
the new path until the peer provides one; see Section 9.5.
An endpoint only changes the address to which it sends packets 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.
An endpoint MAY send data to an unvalidated peer address, but it MUST
protect against potential attacks as described in Sections 9.3.1 and
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.
After changing the address to which it sends non-probing packets, an
endpoint can abandon any path validation for other addresses.
Receiving a packet from a new peer address could 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.
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 limits
the amount of data it sends to that address; see Section 8. In the
absence of this limit, an endpoint risks being used for a denial-of-
service attack against an unsuspecting victim.
If an endpoint skips validation of a peer address as described above,
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. Additionally,
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, thus containing migrations initiated by the attacker
injecting a single packet.
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.
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.
This style of attack relies on the attacker using a path that has
approximately the same characteristics 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 an attack and an improvement
in routing.
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 Appendices A.3 and
B.3 of [QUIC-RECOVERY]) unless the only change in the peer's address
is its port number. Because port-only changes are commonly the
result of NAT rebinding or other middlebox activity, the endpoint MAY
instead retain its congestion control state and round-trip estimate
in those cases instead of reverting to initial values. In cases
where congestion control state retained from an old path is used on a
new path with substantially different characteristics, a sender could
transmit too aggressively until the congestion controller and the RTT
estimator have adapted. Generally, implementations are advised to be
cautious when using previous values on a new path.
There could 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 could 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]) could 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 described 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 canceled 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 6.2.1 of
[QUIC-RECOVERY] and MUST NOT be more aggressive.
9.5. Privacy Implications of Connection Migration
Using a stable connection ID on multiple network paths would allow 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
transmit with 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 field value,
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 associate 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
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 switching to a new
connection ID, source UDP port, or IP address (see [RFC8981]) when
sending traffic after a period of inactivity. Changing the address
from which it sends packets at the same time might cause the server
to detect a connection migration. This ensures that the mechanisms
that support migration are exercised even for clients that do not
experience NAT rebindings or genuine migrations. Changing address
can cause a peer to reset its congestion control state (see
Section 9.4), so addresses 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 not
supported by the version of QUIC specified in this document. If a
client receives packets from a new server address when the client has
not initiated a migration to that address, 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.
Once the handshake is confirmed, the client SHOULD select one of the
two addresses provided by the server and initiate path validation
(see Section 8.2). A client constructs packets using any previously
unused active connection ID, taken from either the preferred_address
transport parameter or a NEW_CONNECTION_ID frame.
As soon as path validation succeeds, the client SHOULD 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. Migration to a Preferred Address
A client that migrates to a preferred address MUST validate the
address it chooses before migrating; see Section 21.5.3.
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 packet containing a
PATH_RESPONSE frame as per Section 8.2. The server MUST send non-
probing packets 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. The server SHOULD drop
newer packets for this connection that are received on the old IP
address. The server MAY continue to process delayed packets that are
received on the old IP address.
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
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 packets received at the server's preferred address have a
different source address than observed from the client during the
handshake, the server MUST protect against potential attacks as
described in Sections 9.3.1 and 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; see
Section 8.
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 flow label generation MUST be designed to minimize the chances of
linkability with a previously used flow label, as a stable flow label
would enable correlating activity on multiple paths; see Section 9.5.
[RFC6437] suggests deriving values using a pseudorandom function to
generate flow labels. Including the Destination Connection ID field
in addition to source and destination addresses when generating flow
labels ensures that changes are synchronized with changes in other
observable identifiers. A cryptographic hash function that combines
these inputs with a local secret is one way this might be
implemented.
10. Connection Termination
An established QUIC connection can be terminated in one of three
ways:
* idle timeout (Section 10.1)
* immediate close (Section 10.2)
* stateless reset (Section 10.3)
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. Idle Timeout
If a max_idle_timeout is specified by either endpoint 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 the max_idle_timeout value advertised by both
endpoints.
Each endpoint advertises a max_idle_timeout, but the effective value
at an endpoint is computed as the minimum of the two advertised
values (or the sole advertised value, if only one endpoint advertises
a non-zero value). By announcing a max_idle_timeout, an endpoint
commits to initiating an immediate close (Section 10.2) 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.
To avoid excessively small idle timeout periods, endpoints MUST
increase the idle timeout period to be at least three times the
current Probe Timeout (PTO). This allows for multiple PTOs to
expire, and therefore multiple probes to be sent and lost, prior to
idle timeout.
10.1.1. Liveness Testing
An endpoint that sends packets close to the effective timeout risks
having them be discarded at the peer, since the idle timeout period
might have expired at the peer 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 6.2 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.
10.1.2. Deferring Idle Timeout
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.
An implementation of QUIC might provide applications with an option
to defer an idle timeout. This facility could be used when the
application wishes to avoid losing state that has been associated
with an open connection but does not expect to exchange application
data for some time. With this option, an endpoint could send a PING
frame (Section 19.2) periodically, which will cause the peer to
restart its idle timeout period. Sending a packet containing a PING
frame restarts the idle timeout for this endpoint also if this is the
first ack-eliciting packet sent since receiving a packet. Sending a
PING frame causes the peer to respond with an acknowledgment, which
also restarts the idle timeout for the endpoint.
Application protocols that use QUIC SHOULD provide guidance on when
deferring an idle timeout is appropriate. Unnecessary sending of
PING frames could have a detrimental effect on 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 30 seconds is necessary to prevent the majority of
middleboxes from losing state for UDP flows [GATEWAY].
10.2. 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; see Section 10.2.1. After receiving a
CONNECTION_CLOSE frame, endpoints enter the draining state; see
Section 10.2.2.
Violations of the protocol lead to an immediate close.
An immediate close can be used after an application protocol has
arranged to close a connection. This might be after the application
protocol negotiates a graceful shutdown. The application protocol
can exchange messages that are needed for both application endpoints
to agree that the connection can be closed, after which the
application requests that QUIC close the connection. When QUIC
consequently closes the connection, a CONNECTION_CLOSE frame with an
application-supplied error code will be used to signal closure to the
peer.
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 PTO interval as defined in [QUIC-RECOVERY].
Disposing of connection state prior to exiting the closing or
draining state could result in an endpoint generating a Stateless
Reset unnecessarily when it receives a late-arriving packet.
Endpoints that have some alternative means to ensure that late-
arriving packets do not induce a response, such as those that are
able to close the UDP socket, MAY end these states earlier to allow
for faster resource recovery. Servers that retain an open socket for
accepting new connections SHOULD NOT end the closing or draining
state early.
Once its closing or draining state ends, an endpoint SHOULD discard
all connection state. The endpoint MAY send a Stateless Reset in
response to any further incoming packets belonging to this
connection.
10.2.1. Closing Connection State
An endpoint enters the closing state after initiating an immediate
close.
In the closing state, an endpoint retains only enough information to
generate a packet containing a CONNECTION_CLOSE frame and to identify
packets as belonging to the connection. An endpoint in the closing
state sends a packet containing a CONNECTION_CLOSE frame in response
to any incoming packet that it attributes to the connection.
An endpoint SHOULD limit the rate at which it generates packets in
the closing state. For instance, an endpoint could wait for a
progressively increasing number of received packets or amount of time
before responding to received packets.
An endpoint's selected connection ID and the QUIC version are
sufficient information to identify packets for a closing connection;
the endpoint MAY discard all other connection state. An endpoint
that is closing is not required to process any received frame. An
endpoint MAY retain packet protection keys for incoming packets to
allow it to read and process a CONNECTION_CLOSE frame.
An endpoint MAY drop packet protection keys when entering the closing
state and send a packet containing a CONNECTION_CLOSE frame in
response to any UDP datagram that is received. However, an endpoint
that discards packet protection keys cannot identify and discard
invalid packets. To avoid being used for an amplification attack,
such endpoints MUST limit the cumulative size of packets it sends to
three times the cumulative size of the packets that are received and
attributed to the connection. To minimize the state that an endpoint
maintains for a closing connection, endpoints MAY send the exact same
packet in response to any received packet.
| Note: Allowing retransmission of a closing packet is an
| exception to the requirement that a new packet number be used
| for each packet; see Section 12.3. 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.
While in the closing state, an endpoint could receive packets from a
new source address, possibly indicating a connection migration; see
Section 9. An endpoint in the closing state MUST either discard
packets received from an unvalidated address or limit the cumulative
size of packets it sends to an unvalidated address to three times the
size of packets it receives from that address.
An endpoint is not expected to handle key updates when it is closing
(Section 6 of [QUIC-TLS]). A key update might prevent the endpoint
from moving from the closing state to the draining state, as the
endpoint will not be able to process subsequently received packets,
but it otherwise has no impact.
10.2.2. Draining Connection State
The draining state is entered once an endpoint receives a
CONNECTION_CLOSE frame, which indicates 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 that receives a CONNECTION_CLOSE frame MAY send a single
packet containing a CONNECTION_CLOSE frame before entering the
draining state, using a NO_ERROR code if appropriate. An endpoint
MUST NOT send further packets. Doing so could result in a constant
exchange of CONNECTION_CLOSE frames until one of the endpoints exits
the closing state.
An endpoint MAY enter the draining state from the closing state if it
receives a CONNECTION_CLOSE frame, which indicates that the peer is
also closing or draining. In this case, the draining state ends when
the closing state would have ended. In other words, the endpoint
uses the same end time but ceases transmission of any packets on this
connection.
10.2.3. Immediate Close during the Handshake
When sending a CONNECTION_CLOSE frame, 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.
* A client that sends a CONNECTION_CLOSE frame in a 0-RTT packet
cannot be assured that the server has accepted 0-RTT. 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 a
CONNECTION_CLOSE frame in both Handshake and 1-RTT packets. A
server SHOULD also send a CONNECTION_CLOSE frame 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 can send a CONNECTION_CLOSE frame in an Initial packet.
This might be 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 21.2. 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.
10.3. 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 indicating errors in active
connections. An endpoint that wishes to communicate a fatal
connection error MUST use a CONNECTION_CLOSE frame if it is able.
To support this process, an endpoint issues a stateless reset token,
which is a 16-byte value that is hard to guess. If the peer
subsequently receives a Stateless Reset, which is a UDP datagram that
ends in that stateless reset token, the peer will immediately end the
connection.
A stateless reset token is specific to a connection ID. An endpoint
issues a stateless reset token by including the value in the
Stateless Reset Token field of a NEW_CONNECTION_ID frame. Servers
can also issue a stateless_reset_token transport parameter during the
handshake that applies to the connection ID that it selected during
the handshake. These exchanges are protected by encryption, so only
client and server know their value. Note that clients cannot use the
stateless_reset_token transport parameter because their transport
parameters do not have confidentiality protection.
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 (see Section 1.3):
Stateless Reset {
Fixed Bits (2) = 1,
Unpredictable Bits (38..),
Stateless Reset Token (128),
}
Figure 10: Stateless Reset
This design ensures that a Stateless Reset 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 are set to values that
SHOULD be indistinguishable from random. The last 16 bytes of the
datagram contain a stateless reset token.
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).
The resulting 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 achieve that end,
the endpoint SHOULD ensure that all packets it sends are at least 22
bytes longer than the minimum connection ID length that it requests
the peer to include in its packets, adding PADDING frames as
necessary. This ensures that any Stateless Reset sent by the peer is
indistinguishable from a valid packet sent to the endpoint. An
endpoint that sends a Stateless Reset in response to a packet that is
43 bytes or shorter 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 length
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 10.3.3 describes additional limits on
Stateless Reset size.
Endpoints MUST discard packets that are too small to be valid QUIC
packets. To give an example, with the set of AEAD functions defined
in [QUIC-TLS], short header packets that are smaller than 21 bytes
are never valid.
Endpoints MUST send Stateless Resets 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. The Destination 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; see 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 10.3.3. 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 (1) reuse this design or (2) use a portion
of the packet other than the last 16 bytes for carrying data.
10.3.1. 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 Token
field values 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.
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 pseudorandom
function using a secret key (e.g., block cipher, Hashed Message
Authentication Code (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.
10.3.2. 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
[RANDOM] 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 pseudorandom 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 the HMAC-based Key
Derivation Function (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
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 21.11. 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 Resets do not have any cryptographic protection.
10.3.3. 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 that 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 Resets that it has
sent and stop generating new Stateless Resets once a limit is
reached. Using separate limits for different remote addresses will
ensure that Stateless Resets can be used to close connections when
other peers or connections have exhausted limits.
A Stateless Reset that is smaller than 41 bytes might be identifiable
as a Stateless Reset by an observer, depending upon the length of the
peer's connection IDs. Conversely, not sending a Stateless Reset in
response to a small packet might result in Stateless Resets 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 10.3) 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).
Application-specific protocol errors are signaled using the
CONNECTION_CLOSE frame with a frame type of 0x1d. Errors that are
specific to the transport, including all those described in this
document, are carried in the CONNECTION_CLOSE frame with a frame type
of 0x1c.
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.
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 attempt the stateless reset
process (Section 10.3).
As the AEAD for Initial packets does not provide strong
authentication, an endpoint MAY discard an invalid Initial packet.
Discarding an Initial packet is permitted even where this
specification otherwise mandates a connection error. An endpoint can
only discard a packet if it does not process the frames in the packet
or reverts the effects of any processing. Discarding invalid Initial
packets might be used to reduce exposure to denial of service; see
Section 21.2.
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 canceled 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.
12.1. Protected Packets
QUIC packets have different levels of cryptographic protection based
on the type of packet. Details of packet protection are found in
[QUIC-TLS]; this section includes an overview of the protections that
are provided.
Version Negotiation packets have no cryptographic protection; see
[QUIC-INVARIANTS].
Retry packets use an AEAD function [AEAD] to protect against
accidental modification.
Initial packets use an AEAD function, the keys for which are derived
using a value that is visible on the wire. Initial packets therefore
do not have effective confidentiality protection. Initial protection
exists to ensure that the sender of the packet is on the network
path. Any entity that receives an Initial packet from a client can
recover the keys that will allow them to both read the contents of
the packet and generate Initial packets that will be successfully
authenticated at either endpoint. The AEAD also protects Initial
packets against accidental modification.
All other packets are protected with keys derived from the
cryptographic handshake. The cryptographic handshake ensures that
only the communicating endpoints receive the corresponding keys for
Handshake, 0-RTT, and 1-RTT packets. Packets protected with 0-RTT
and 1-RTT keys have strong confidentiality and integrity protection.
The Packet Number field that appears in some packet types has
alternative confidentiality protection that is applied as part of
header protection; see Section 5.4 of [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 that 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 Path Maximum Transmission
Unit (PMTU) probes; see Section 14.4.1. 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 that the receiver will be able to process all the packets
in a 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 include multiple frames in a single packet if they
are to be sent at the same encryption level, instead of coalescing
multiple packets at the same encryption level.
Receivers MAY route based on the information in the first packet
contained in a UDP datagram. Senders MUST NOT coalesce QUIC packets
with different connection IDs 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 for 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 three spaces in QUIC:
Initial space: All Initial packets (Section 17.2.2) are in this
space.
Handshake space: All Handshake packets (Section 17.2.4) are in this
space.
Application data space: All 0-RTT (Section 17.2.3) and 1-RTT
(Section 17.3.1) packets 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 that
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 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 10.3) 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 9.5 of [QUIC-TLS].
Endpoints that track all individual packets for the purposes of
detecting duplicates are at risk of accumulating excessive state.
The data required for detecting duplicates can be limited by
maintaining a minimum packet number below which all packets are
immediately dropped. Any minimum needs to account for large
variations in round-trip time, which includes the possibility that a
peer might probe network paths with much larger round-trip times; see
Section 9.
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 11.
Version Negotiation, Stateless Reset, and Retry packets do not
contain frames.
Packet Payload {
Frame (8..) ...,
}
Figure 11: 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.
An endpoint MUST treat receipt of a packet containing no frames as a
connection error of type PROTOCOL_VIOLATION. 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 12: Generic Frame Layout
Table 3 lists and summarizes information about each frame type that
is defined in this specification. A description of this summary is
included after the table.
+============+======================+===============+======+======+
| Type Value | Frame Type Name | Definition | Pkts | Spec |
+============+======================+===============+======+======+
| 0x00 | PADDING | Section 19.1 | IH01 | NP |
+------------+----------------------+---------------+------+------+
| 0x01 | PING | Section 19.2 | IH01 | |
+------------+----------------------+---------------+------+------+
| 0x02-0x03 | ACK | Section 19.3 | IH_1 | NC |
+------------+----------------------+---------------+------+------+
| 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 | F |
+------------+----------------------+---------------+------+------+
| 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 | P |
+------------+----------------------+---------------+------+------+
| 0x19 | RETIRE_CONNECTION_ID | Section 19.16 | __01 | |
+------------+----------------------+---------------+------+------+
| 0x1a | PATH_CHALLENGE | Section 19.17 | __01 | P |
+------------+----------------------+---------------+------+------+
| 0x1b | PATH_RESPONSE | Section 19.18 | ___1 | P |
+------------+----------------------+---------------+------+------+
| 0x1c-0x1d | CONNECTION_CLOSE | Section 19.19 | ih01 | N |
+------------+----------------------+---------------+------+------+
| 0x1e | HANDSHAKE_DONE | Section 19.20 | ___1 | |
+------------+----------------------+---------------+------+------+
Table 3: Frame Types
The format and semantics of each frame type are explained in more
detail in Section 19. The remainder of this section provides a
summary of important and general information.
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.
The "Pkts" column in Table 3 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 17.3.1)
ih: Only a CONNECTION_CLOSE frame of type 0x1c can appear in Initial
or Handshake packets.
For more details about these restrictions, see Section 12.5. Note
that all frames can appear in 1-RTT packets. An endpoint MUST treat
receipt of a frame in a packet type that is not permitted as a
connection error of type PROTOCOL_VIOLATION.
The "Spec" column in Table 3 summarizes any special rules governing
the processing or generation of the frame type, as indicated by the
following characters:
N: Packets containing only frames with this marking are not ack-
eliciting; see Section 13.2.
C: Packets containing only frames with this marking do not count
toward bytes in flight for congestion control purposes; see
[QUIC-RECOVERY].
P: Packets containing only frames with this marking can be used to
probe new network paths during connection migration; see
Section 9.1.
F: The contents of frames with this marking are flow controlled;
see Section 4.
The "Pkts" and "Spec" columns in Table 3 do not form part of the IANA
registry; see Section 22.4.
An endpoint MUST treat the receipt of a frame of unknown type as a
connection error of type FRAME_ENCODING_ERROR.
All 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.
12.5. Frames and Number Spaces
Some frames are prohibited in different packet number spaces. The
rules here generalize those of TLS, in that frames associated with
establishing the connection can usually appear in packets in any
packet number space, whereas those associated with transferring data
can only appear in the application data packet number space:
* PADDING, PING, and CRYPTO frames MAY appear in any packet number
space.
* CONNECTION_CLOSE frames signaling errors at the QUIC layer (type
0x1c) MAY appear in any packet number space. CONNECTION_CLOSE
frames signaling application errors (type 0x1d) MUST only appear
in the application data packet number space.
* ACK frames MAY appear in any packet number space but can only
acknowledge packets that appeared in that packet number space.
However, as noted below, 0-RTT packets cannot contain ACK frames.
* All other frame types MUST only be sent in the application data
packet number space.
Note that it is not possible to send the following frames in 0-RTT
packets for various reasons: ACK, CRYPTO, HANDSHAKE_DONE, NEW_TOKEN,
PATH_RESPONSE, and RETIRE_CONNECTION_ID. A server MAY treat receipt
of these frames in 0-RTT packets as a connection error of type
PROTOCOL_VIOLATION.
13. Packetization and Reliability
A sender sends one or more frames in a QUIC packet; see Section 12.4.
A sender can minimize per-packet bandwidth and computational costs by
including as many frames as possible in each QUIC packet. A sender
MAY wait for a short period of time to collect multiple frames before
sending a packet that is not maximally packed, to avoid sending out
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 included in a single QUIC
packet, loss of that packet blocks all those streams from making
progress. Implementations are advised to include 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 be 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.
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. For further discussion of how this
might be achieved, see Section 21.4.
13.2. Generating Acknowledgments
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
include an ACK frame if one has not been sent recently. Doing so
helps with timely loss detection at the peer.
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 delay
an endpoint communicated 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. A sender uses the
receiver's max_ack_delay value in determining timeouts for timer-
based retransmission, as detailed in Section 6.2 of [QUIC-RECOVERY].
An endpoint MUST acknowledge all ack-eliciting Initial and Handshake
packets immediately and all ack-eliciting 0-RTT and 1-RTT packets
within its advertised max_ack_delay, with the following exception.
Prior to handshake confirmation, an endpoint might not have packet
protection keys for decrypting Handshake, 0-RTT, or 1-RTT packets
when they are received. It might therefore buffer them and
acknowledge them when the requisite keys become available.
Since packets containing only ACK frames are not congestion
controlled, an endpoint MUST NOT send more than one such 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 that precede
the received packet. This avoids an infinite feedback loop of
acknowledgments, which could prevent the connection from ever
becoming idle. Non-ack-eliciting packets are eventually acknowledged
when the endpoint sends an ACK frame in response to other events.
An endpoint that is only sending ACK frames will not receive
acknowledgments from its peer unless those acknowledgments are
included in packets with ack-eliciting frames. An endpoint SHOULD
send an ACK frame 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 choose not to send an ACK frame with
outgoing frames until an ack-eliciting packet has been received.
An endpoint that is only sending non-ack-eliciting packets might
choose to occasionally add an ack-eliciting frame to those packets to
ensure that it receives an acknowledgment; see Section 13.2.4. In
that case, an endpoint MUST NOT send an ack-eliciting frame in all
packets that would otherwise be non-ack-eliciting, to avoid an
infinite feedback loop of acknowledgments.
In order to assist loss detection at the sender, an endpoint SHOULD
generate and send an ACK frame without delay when it receives an ack-
eliciting packet either:
* when the received packet has a packet number less than another
ack-eliciting packet that has been received, or
* when the packet has a packet number larger than the highest-
numbered ack-eliciting packet that has been received and there are
missing packets between that packet and this 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.
The algorithms in [QUIC-RECOVERY] are expected to be resilient to
receivers that do not follow the guidance offered above. However, an
implementation should only deviate from these requirements after
careful consideration of the performance implications of a change,
for connections made by the endpoint and for other users of the
network.
13.2.2. Acknowledgment Frequency
A receiver determines how frequently to send acknowledgments in
response to ack-eliciting packets. This determination involves a
trade-off.
Endpoints rely on timely acknowledgment to detect loss; see Section 6
of [QUIC-RECOVERY]. Window-based congestion controllers, such as the
one described in Section 7 of [QUIC-RECOVERY], rely on
acknowledgments to manage their congestion window. In both cases,
delaying acknowledgments can adversely affect performance.
On the other hand, reducing the frequency of packets that carry only
acknowledgments reduces packet transmission and processing cost at
both endpoints. It can improve connection throughput on severely
asymmetric links and reduce the volume of acknowledgment traffic
using return path capacity; see Section 3 of [RFC3449].
A receiver SHOULD send an ACK frame after receiving at least two ack-
eliciting packets. This recommendation is general in nature and
consistent with recommendations for TCP endpoint behavior [RFC5681].
Knowledge of network conditions, knowledge of the peer's congestion
controller, or further research and experimentation might suggest
alternative acknowledgment strategies with better performance
characteristics.
A receiver MAY process multiple available packets before determining
whether to send an ACK frame in response.
13.2.3. Managing ACK Ranges
When an ACK frame is sent, one or more ranges of acknowledged packets
are included. Including acknowledgments for older packets reduces
the chance of spurious retransmissions 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.
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. Senders can expect acknowledgments
for most packets, but QUIC does not guarantee receipt of an
acknowledgment for every packet that the receiver processes.
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. Receivers MAY also limit ACK
frame size further to preserve space for other frames or to limit the
capacity that acknowledgments consume.
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.
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.
Section 13.2.4 describes an exemplary approach for determining what
packets to acknowledge in each ACK frame. Though the goal of this
algorithm is to generate an acknowledgment for every packet that is
processed, it is still possible for acknowledgments to be lost.
13.2.4. Limiting Ranges by Tracking ACK Frames
When a packet containing an ACK frame is sent, the Largest
Acknowledged field in that frame can 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
field in the sent ACK frame.
A receiver that sends only non-ack-eliciting packets, such as ACK
frames, might not receive an acknowledgment 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
send a PING or other small ack-eliciting frame occasionally, such as
once per round trip, to elicit an ACK from the peer.
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 acknowledgment 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 retransmissions, but the sender will
continue making forward progress.
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
acknowledgment delay in the ACK Delay field of an ACK frame; see
Section 19.3. This allows the receiver of the ACK frame 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.
However, endpoints SHOULD include buffering delays caused by
unavailability of decryption keys, since these delays can be large
and are likely to be non-repeating.
When the measured acknowledgment delay is larger than its
max_ack_delay, an endpoint SHOULD report the measured delay. This
information is especially useful during the handshake when delays
might be large; see Section 13.2.1.
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 acknowledged; 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.
13.2.7. PADDING Frames Consume Congestion Window
Packets containing PADDING frames are considered to be in flight for
congestion control purposes [QUIC-RECOVERY]. Packets containing only
PADDING frames therefore consume congestion window but do not
generate acknowledgments that will open the congestion window. To
avoid a deadlock, a sender SHOULD ensure that other frames are sent
periodically in addition to PADDING frames to elicit acknowledgments
from the receiver.
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 acknowledgments and the
acknowledgment delay from the largest acknowledged packet, as
described in Section 13.2.1. Delaying the transmission of packets
containing ACK frames or resending 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. Resending these signals is 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
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; see Section 4.2.
* 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" or "Reset Recvd" 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. A
new frame is sent if a packet 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.
* 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 changes 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 value than one found in an older packet.
A sender SHOULD avoid retransmitting information from packets once
they are acknowledged. This includes packets that are acknowledged
after being declared lost, which can happen in the presence of
network reordering. Doing so requires senders to retain information
about packets after they are declared lost. A sender can discard
this information after a period of time elapses that adequately
allows for reordering, such as a PTO (Section 6.2 of
[QUIC-RECOVERY]), or based on other events, such as reaching a memory
limit.
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 ECN [RFC3168] to detect and respond to network
congestion. ECN allows an endpoint to set an ECN-Capable Transport
(ECT) codepoint in the ECN field of an IP packet. A network node can
then indicate congestion by setting the ECN-CE codepoint in the ECN
field instead of dropping the packet [RFC8087]. Endpoints react to
reported congestion by reducing their sending rate in response, as
described in [QUIC-RECOVERY].
To enable ECN, a sending QUIC endpoint first determines whether a
path supports ECN marking and whether the peer reports the ECN values
in received IP headers; see Section 13.4.2.
13.4.1. Reporting ECN Counts
The use of ECN requires the receiving endpoint to read the ECN field
from an IP packet, which is not possible on all platforms. If an
endpoint does not implement ECN support or does not have access to
received ECN fields, it does not report ECN counts for packets it
receives.
Even if an endpoint does not set an ECT field in packets it sends,
the endpoint MUST provide feedback about ECN markings it receives, if
these are accessible. Failing to report the ECN counts will cause
the sender to disable the use of ECN for this connection.
On receiving an IP packet with an ECT(0), ECT(1), or ECN-CE
codepoint, an ECN-enabled endpoint accesses the ECN field and
increases the corresponding ECT(0), ECT(1), or ECN-CE count. These
ECN counts are included in subsequent ACK frames; see Sections 13.2
and 19.3.
Each packet number space maintains separate acknowledgment state and
separate ECN counts. Coalesced QUIC packets (see Section 12.2) share
the same IP header so the ECN counts are incremented once for each
coalesced QUIC packet.
For example, if one each of an Initial, Handshake, and 1-RTT QUIC
packet are coalesced into a single UDP datagram, the ECN counts for
all three packet number spaces will be incremented by one each, based
on the ECN field of the single IP header.
ECN counts are only incremented when QUIC packets from the received
IP packet are processed. As such, duplicate QUIC packets are not
processed and do not increase ECN counts; see Section 21.10 for
relevant security concerns.
13.4.2. ECN Validation
It is possible for faulty network devices to corrupt or erroneously
drop packets that carry a non-zero ECN codepoint. To ensure
connectivity in the presence of such devices, an endpoint validates
the ECN counts for each network path and disables the use of ECN on
that path if errors are detected.
To perform ECN validation for a new path:
* The endpoint sets an ECT(0) codepoint in the IP header of early
outgoing packets sent on a new path to the peer [RFC8311].
* The endpoint monitors whether all packets sent with an ECT
codepoint are eventually deemed lost (Section 6 of
[QUIC-RECOVERY]), indicating that ECN validation has failed.
If an endpoint has cause to expect that IP packets with an ECT
codepoint might be dropped by a faulty network element, the endpoint
could set an ECT codepoint for only the first ten outgoing packets on
a path, or for a period of three PTOs (see Section 6.2 of
[QUIC-RECOVERY]). If all packets marked with non-zero ECN codepoints
are subsequently lost, it can disable marking on the assumption that
the marking caused the loss.
An endpoint thus attempts to use ECN and validates this for each new
connection, when switching to a server's preferred address, and on
active connection migration to a new path. Appendix A.4 describes
one possible algorithm.
Other methods of probing paths for ECN support are possible, as are
different marking strategies. Implementations MAY use other methods
defined in RFCs; see [RFC8311]. Implementations that use the ECT(1)
codepoint need to perform ECN validation using the reported ECT(1)
counts.
13.4.2.1. Receiving ACK Frames with ECN Counts
Erroneous application of ECN-CE markings by the network can result in
degraded connection performance. An endpoint that receives an ACK
frame with ECN counts therefore validates the counts before using
them. It performs this validation by comparing newly received counts
against those from the last successfully processed ACK frame. Any
increase in the ECN counts is validated based on the ECN markings
that were applied to packets that are newly acknowledged in the ACK
frame.
If an ACK frame newly acknowledges a packet that the endpoint sent
with either the ECT(0) or ECT(1) codepoint set, ECN validation fails
if the corresponding ECN counts are not present in the ACK frame.
This check detects a network element that zeroes the ECN field or a
peer that does not report ECN markings.
ECN validation also fails if the sum of the increase in ECT(0) and
ECN-CE counts is less than the number of newly acknowledged packets
that were originally sent with an ECT(0) marking. Similarly, ECN
validation fails if the sum of the increases to ECT(1) and ECN-CE
counts is less than the number of newly acknowledged packets sent
with an ECT(1) marking. These checks can detect remarking of ECN-CE
markings by the network.
An endpoint could miss acknowledgments for a packet when ACK frames
are lost. It is therefore possible for the total increase in ECT(0),
ECT(1), and ECN-CE counts to be greater than the number of packets
that are newly acknowledged by an ACK frame. This is why ECN counts
are permitted to be larger than the total number of packets that are
acknowledged.
Validating ECN counts from reordered ACK frames can result in
failure. An endpoint MUST NOT fail ECN validation as a result of
processing an ACK frame that does not increase the largest
acknowledged packet number.
ECN validation can fail if the received total count for either ECT(0)
or ECT(1) exceeds the total number of packets sent with each
corresponding ECT codepoint. In particular, validation will fail
when an endpoint receives a non-zero ECN count corresponding to an
ECT codepoint that it never applied. This check detects when packets
are remarked to ECT(0) or ECT(1) in the network.
13.4.2.2. ECN Validation Outcomes
If validation fails, then the endpoint MUST disable ECN. It stops
setting the ECT codepoint in IP packets that it sends, assuming that
either the network path or the peer does not support ECN.
Even if validation fails, an endpoint MAY revalidate ECN for the same
path at any later time in the connection. An endpoint could continue
to periodically attempt validation.
Upon successful validation, an endpoint MAY continue to set an ECT
codepoint in subsequent packets it sends, with the expectation that
the path is ECN capable. Network routing and path elements can
change mid-connection; an endpoint MUST disable ECN if validation
later fails.
14. Datagram Size
A UDP datagram can include one or more QUIC packets. The datagram
size refers to the total UDP payload size of a single UDP datagram
carrying QUIC packets. The datagram size includes one or more QUIC
packet headers and protected payloads, but not the UDP or IP headers.
The maximum datagram size is defined as the largest size of UDP
payload that can be sent across a network path using a single UDP
datagram. QUIC MUST NOT be used if the network path cannot support a
maximum datagram size of at least 1200 bytes.
QUIC assumes a minimum IP packet size of at least 1280 bytes. This
is the IPv6 minimum size [IPv6] and is also supported by most modern
IPv4 networks. Assuming the minimum IP header size of 40 bytes for
IPv6 and 20 bytes for IPv4 and a UDP header size of 8 bytes, this
results in a maximum datagram size of 1232 bytes for IPv6 and 1252
bytes for IPv4. Thus, modern IPv4 and all IPv6 network paths are
expected to be able to support QUIC.
| Note: This requirement to support a UDP payload of 1200 bytes
| limits the space available for IPv6 extension headers to 32
| bytes or IPv4 options to 52 bytes if the path only supports the
| IPv6 minimum MTU of 1280 bytes. This affects Initial packets
| and path validation.
Any maximum datagram size larger than 1200 bytes can be discovered
using Path Maximum Transmission Unit Discovery (PMTUD) (see
Section 14.2.1) or Datagram Packetization Layer PMTU Discovery
(DPLPMTUD) (see Section 14.3).
Enforcement of the max_udp_payload_size transport parameter
(Section 18.2) might act as an additional limit on the maximum
datagram size. A sender can avoid exceeding this limit, once the
value is known. However, prior to learning the value of the
transport parameter, endpoints risk datagrams being lost if they send
datagrams larger than the smallest allowed maximum datagram size of
1200 bytes.
UDP datagrams MUST NOT be fragmented at the IP layer. In IPv4
[IPv4], the Don't Fragment (DF) bit MUST be set if possible, to
prevent fragmentation on the path.
QUIC sometimes requires datagrams to be no smaller than a certain
size; see Section 8.1 as an example. However, the size of a datagram
is not authenticated. That is, if an endpoint receives a datagram of
a certain size, it cannot know that the sender sent the datagram at
the same size. Therefore, an endpoint MUST NOT close a connection
when it receives a datagram that does not meet size constraints; the
endpoint MAY discard such datagrams.
14.1. Initial Datagram Size
A client MUST expand the payload of all UDP datagrams carrying
Initial packets to at least the smallest allowed maximum datagram
size of 1200 bytes by adding PADDING frames to the Initial packet or
by coalescing the Initial packet; see Section 12.2. Initial packets
can even be coalesced with invalid packets, which a receiver will
discard. Similarly, a server MUST expand the payload of all UDP
datagrams carrying ack-eliciting Initial packets to at least the
smallest allowed maximum datagram size of 1200 bytes.
Sending UDP datagrams of this size ensures that the network path
supports a reasonable Path Maximum Transmission Unit (PMTU), in both
directions. Additionally, a client that expands Initial packets
helps reduce the amplitude of amplification attacks caused by server
responses toward an unverified client address; see Section 8.
Datagrams containing Initial packets MAY exceed 1200 bytes if the
sender believes that the network path and peer both support the size
that it chooses.
A server MUST discard an Initial packet that is carried in a UDP
datagram with a payload that is smaller than the smallest allowed
maximum datagram size of 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 10.2.3.
The server MUST also limit the number of bytes it sends before
validating the address of the client; see Section 8.
14.2. Path Maximum Transmission Unit
The PMTU is the maximum size of the entire IP packet, including the
IP header, UDP header, and UDP payload. The UDP payload includes one
or more QUIC packet headers and protected payloads. 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 datagram size.
An endpoint SHOULD use DPLPMTUD (Section 14.3) or PMTUD
(Section 14.2.1) to determine whether the path to a destination will
support a desired maximum datagram size without fragmentation. In
the absence of these mechanisms, QUIC endpoints SHOULD NOT send
datagrams larger than the smallest allowed maximum datagram size.
Both DPLPMTUD and PMTUD send datagrams that are larger than the
current maximum datagram size, referred to as PMTU probes. All QUIC
packets that are not sent in a PMTU probe SHOULD be sized to fit
within the maximum datagram size to avoid the datagram being
fragmented or dropped [RFC8085].
If a QUIC endpoint determines that the PMTU between any pair of local
and remote IP addresses cannot support the smallest allowed maximum
datagram size of 1200 bytes, it MUST immediately cease sending QUIC
packets, except for those in PMTU probes or those containing
CONNECTION_CLOSE frames, on the affected path. An endpoint MAY
terminate the connection if an alternative path cannot be found.
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 datagram size for each
combination of local and remote IP addresses.
A QUIC implementation MAY be more conservative in computing the
maximum datagram size to allow for unknown tunnel overheads or IP
header options/extensions.
14.2.1. Handling of ICMP Messages by PMTUD
PMTUD [RFC1191] [RFC8201] relies on reception of ICMP messages (that
is, IPv6 Packet Too Big (PTB) messages) that indicate when an IP
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 attacks by entities that cannot
observe packets but might successfully guess the addresses used on
the path. These attacks could reduce the PMTU to a bandwidth-
inefficient value.
An endpoint MUST ignore an ICMP message that claims the PMTU has
decreased below QUIC's smallest allowed maximum datagram size.
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 using PMTUD SHOULD validate ICMP messages to protect
from packet 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 (see Section 4.6.1 of [DPLPMTUD]).
ICMP message validation MUST include matching IP addresses and UDP
ports [RFC8085] and, when possible, connection IDs to an active QUIC
session. The endpoint SHOULD ignore all ICMP messages that fail
validation.
An endpoint MUST NOT increase the PMTU based on ICMP messages; see
Item 6 in Section 3 of [DPLPMTUD]. Any reduction in QUIC's maximum
datagram 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
DPLPMTUD [DPLPMTUD] relies on tracking loss or acknowledgment of QUIC
packets that are carried in PMTU probes. PMTU probes for DPLPMTUD
that use the PADDING frame implement "Probing using padding data", as
defined in Section 4.1 of [DPLPMTUD].
Endpoints SHOULD set the initial value of BASE_PLPMTU (Section 5.1 of
[DPLPMTUD]) to be consistent with QUIC's smallest allowed maximum
datagram size. The MIN_PLPMTU is the same as the BASE_PLPMTU.
QUIC endpoints implementing DPLPMTUD maintain a DPLPMTUD Maximum
Packet Size (MPS) (Section 4.4 of [DPLPMTUD]) for each combination of
local and remote IP addresses. This corresponds to the maximum
datagram size.
14.3.1. DPLPMTUD and Initial Connectivity
From the perspective of DPLPMTUD, QUIC is an acknowledged
Packetization Layer (PL). A QUIC sender can therefore enter the
DPLPMTUD BASE state (Section 5.2 of [DPLPMTUD]) when the QUIC
connection handshake has been completed.
14.3.2. Validating the Network Path with DPLPMTUD
QUIC is an acknowledged PL; therefore, a QUIC sender does not
implement a DPLPMTUD CONFIRMATION_TIMER while in the SEARCH_COMPLETE
state; see Section 5.2 of [DPLPMTUD].
14.3.3. Handling of ICMP Messages by DPLPMTUD
An endpoint using DPLPMTUD requires the validation of any received
ICMP PTB message before using the PTB information, as defined in
Section 4.6 of [DPLPMTUD]. In addition to UDP port validation, QUIC
validates an ICMP message by using other PL information (e.g.,
validation of connection IDs in the quoted packet of any received
ICMP message).
The considerations for processing ICMP messages described in
Section 14.2.1 also apply if these messages are used by DPLPMTUD.
14.4. Sending QUIC PMTU Probes
PMTU probes are ack-eliciting packets.
Endpoints could limit the content of PMTU probes to PING and PADDING
frames, since packets that are larger than the current maximum
datagram size are more likely to be dropped by the network. Loss of
a QUIC packet that is carried in a PMTU probe is therefore not a
reliable indication of congestion and SHOULD NOT trigger a congestion
control reaction; see Item 7 in Section 3 of [DPLPMTUD]. However,
PMTU probes consume congestion window, which could delay subsequent
transmission by an application.
14.4.1. PMTU Probes Containing Source Connection ID
Endpoints that rely on the Destination Connection ID field for
routing incoming QUIC packets are likely to require that the
connection ID be included in PMTU probes to route any resulting ICMP
messages (Section 14.2.1) back to the correct endpoint. However,
only long header packets (Section 17.2) contain the Source Connection
ID field, and long header packets are not decrypted or acknowledged
by the peer once the handshake is complete.
One way to construct a PMTU probe is to coalesce (see Section 12.2) a
packet with a long header, such as a Handshake or 0-RTT packet
(Section 17.2), with a short header packet in a single UDP datagram.
If the resulting PMTU probe reaches the endpoint, the packet with the
long header will be ignored, but the short header packet will be
acknowledged. If the PMTU probe causes an ICMP message to be sent,
the first part of the probe will be quoted in that message. If the
Source Connection ID field is within the quoted portion of the probe,
that could be used for routing or validation of the ICMP message.
| Note: The purpose of using a packet with a long header is only
| to ensure that the quoted packet contained in the ICMP message
| contains a Source Connection ID field. This packet does not
| need to be a valid packet, and it can be sent even if there is
| no current use for packets of that type.
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 from 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.
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.
+======+========+=============+=======================+
| 2MSB | 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
An example of a decoding algorithm and sample encodings are shown in
Appendix A.1.
Values do not need to be encoded on the minimum number of bytes
necessary, with the sole exception of the Frame Type field; see
Section 12.4.
Versions (Section 15), packet numbers sent in the header
(Section 17.1), and the length of connection IDs in long header
packets (Section 17.2) 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.
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 only the least significant bits of the
packet number.
The encoded packet number is protected as described in Section 5.4 of
[QUIC-TLS].
Prior to receiving an acknowledgment for a packet number space, the
full packet number MUST be included; it is not to be truncated, as
described below.
After an acknowledgment is received for a packet number space, the
sender MUST use a packet number size able to represent more than
twice as large a range as the difference between the largest
acknowledged packet number and the 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. Pseudocode
and an example for packet number encoding can be found in
Appendix A.2.
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 in a
successfully authenticated packet. Recovering the full packet number
is necessary to successfully complete the removal of 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. Pseudocode and an example for packet number
decoding can be found in Appendix A.3.
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),
Destination Connection ID Length (8),
Destination Connection ID (0..160),
Source Connection ID Length (8),
Source Connection ID (0..160),
Type-Specific Payload (..),
}
Figure 13: 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, unless the
packet is a Version Negotiation packet. Packets containing a zero
value for this bit are not valid packets in this version and MUST
be discarded. A value of 1 for this bit allows QUIC to coexist
with other protocols; see [RFC7983].
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 semantics of the lower four bits (those with
a mask of 0x0f) of byte 0 are determined by the packet type.
Version: The QUIC Version is a 32-bit field that follows the first
byte. This field indicates the version of QUIC that is in use and
determines how the rest of the protocol fields are interpreted.
Destination Connection ID 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 bytes. Endpoints that receive a version 1 long header with a
value larger than 20 MUST drop the packet. In order to properly
form a Version Negotiation packet, servers SHOULD be able to read
longer connection IDs from other QUIC versions.
Destination Connection ID: The Destination Connection ID field
follows the Destination Connection ID Length field, which
indicates the length of this field. Section 7.2 describes the use
of this field in more detail.
Source Connection ID 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 an
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. In order to
properly form a Version Negotiation packet, servers SHOULD be able
to read longer connection IDs from other QUIC versions.
Source Connection ID: The Source Connection ID field follows the
Source Connection ID Length field, which indicates the length of
this field. Section 7.2 describes the use of this field in more
detail.
Type-Specific Payload: The remainder of the packet, if any, is type
specific.
In this version of QUIC, the following packet types with the long
header are defined:
+======+===========+================+
| Type | Name | Section |
+======+===========+================+
| 0x00 | Initial | Section 17.2.2 |
+------+-----------+----------------+
| 0x01 | 0-RTT | Section 17.2.3 |
+------+-----------+----------------+
| 0x02 | Handshake | Section 17.2.4 |
+------+-----------+----------------+
| 0x03 | Retry | Section 17.2.5 |
+------+-----------+----------------+
Table 5: Long Header Packet Types
The header form bit, Destination and Source Connection ID lengths,
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 9.5 of [QUIC-TLS].
Packet Number Length: In packet types that 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 field, 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: This is 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: This field is 1 to 4 bytes long. The packet number
is protected using header protection; see 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.
Packet Payload: This is the payload of the packet -- containing a
sequence of frames -- that is protected using packet protection.
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. It is only
sent by servers.
The layout of a Version Negotiation packet is:
Version Negotiation Packet {
Header Form (1) = 1,
Unused (7),
Version (32) = 0,
Destination Connection ID Length (8),
Destination Connection ID (0..2040),
Source Connection ID Length (8),
Source Connection ID (0..2040),
Supported Version (32) ...,
}
Figure 14: Version Negotiation Packet
The value in the Unused field is set to an arbitrary value by the
server. Clients MUST ignore the value of this field. Where QUIC
might be multiplexed with other protocols (see [RFC7983]), 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. Note
that other versions of QUIC might not make a similar recommendation.
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 entity that
did not observe the Initial packet.
Future versions of QUIC could have different requirements for the
lengths of connection IDs. In particular, connection IDs might have
a smaller minimum length or a greater maximum length. Version-
specific rules for the connection ID therefore MUST NOT influence a
decision about whether to send a Version Negotiation packet.
The remainder of the Version Negotiation packet is a list of 32-bit
versions that the server supports.
A Version Negotiation packet is not acknowledged. It is only sent in
response to a packet that indicates an unsupported version; see
Section 5.2.2.
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.
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 0x00. It
carries the first CRYPTO frames sent by the client and server to
perform key exchange, and it carries ACK frames 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),
Destination Connection ID Length (8),
Destination Connection ID (0..160),
Source Connection ID Length (8),
Source Connection ID (0..160),
Token Length (i),
Token (..),
Length (i),
Packet Number (8..32),
Packet Payload (8..),
}
Figure 15: Initial Packet
The Initial packet contains a long header as well as the Length and
Packet Number fields; see Section 17.2. The first byte contains the
Reserved and Packet Number Length bits; see also Section 17.2.
Between the Source Connection ID 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 0 if no token is present.
Initial packets sent by the server MUST set the Token Length field
to 0; 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; see Section 8.1.
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 attackers that can
observe packets, but it does prevent attackers that cannot observe
packets from spoofing Initial packets.
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; see 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 of type 0x1c 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 TLS HelloRetryRequest (see
Section 4.7 of [QUIC-TLS]), 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 4.9.1 of [QUIC-TLS]) along with any loss recovery and
congestion control state; see Section 6.4 of [QUIC-RECOVERY].
Any data in CRYPTO frames is discarded -- and no longer retransmitted
-- when Initial keys are discarded.
17.2.3. 0-RTT
A 0-RTT packet uses long headers with a type value of 0x01, followed
by the Length and Packet Number fields; see Section 17.2. The first
byte contains the Reserved and Packet Number Length bits; see
Section 17.2. A 0-RTT packet 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),
Destination Connection ID Length (8),
Destination Connection ID (0..160),
Source Connection ID Length (8),
Source Connection ID (0..160),
Length (i),
Packet Number (8..32),
Packet Payload (8..),
}
Figure 16: 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.
New packet numbers MUST be used for any new packets that are sent; as
described in Section 17.2.5.3, reusing packet numbers could
compromise packet protection.
A client only receives acknowledgments for its 0-RTT packets once the
handshake is complete, as defined in Section 4.1.1 of [QUIC-TLS].
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
(Section 7.4.1) 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 0x02,
followed by the Length and Packet Number fields; see Section 17.2.
The first byte contains the Reserved and Packet Number Length bits;
see Section 17.2. It is used to carry cryptographic handshake
messages and acknowledgments 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),
Destination Connection ID Length (8),
Destination Connection ID (0..160),
Source Connection ID Length (8),
Source Connection ID (0..160),
Length (i),
Packet Number (8..32),
Packet Payload (8..),
}
Figure 17: 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.
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 have 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 of type 0x1c. Endpoints MUST treat receipt
of Handshake packets with other frames as a connection error of type
PROTOCOL_VIOLATION.
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
As shown in Figure 18, a Retry packet uses a long packet header with
a type value of 0x03. 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),
Destination Connection ID Length (8),
Destination Connection ID (0..160),
Source Connection ID Length (8),
Source Connection ID (0..160),
Retry Token (..),
Retry Integrity Tag (128),
}
Figure 18: Retry Packet
A Retry packet does not contain any protected fields. The value in
the Unused field is set to an arbitrary value by the server; a client
MUST ignore these bits. 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.
Retry Integrity Tag: Defined in Section 5.8 ("Retry Packet
Integrity") 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 Section 5.8 of [QUIC-TLS]. This
diminishes an 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.
A client sets the Destination Connection ID field of this Initial
packet to the value from the Source Connection ID field in the Retry
packet. Changing the Destination Connection ID field 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 packet. 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
Subsequent Initial packets from the client include the connection ID
and token values from the Retry packet. The client copies the Source
Connection ID field from the Retry packet to the Destination
Connection ID field and uses this value until an Initial packet with
an updated value is received; see Section 7.2. The value of the
Token field is copied to all subsequent Initial packets; see
Section 8.1.2.
Other than updating the Destination Connection ID and Token fields,
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 included in this packet. A server
MAY treat a packet that contains a different cryptographic handshake
message as a connection error or discard it. Note that including a
Token field reduces the available space for the cryptographic
handshake message, which might result in the client needing to send
multiple Initial packets.
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 reset the packet number for any packet number space
after processing a Retry packet. In particular, 0-RTT packets
contain confidential information that will most likely be
retransmitted on receiving a Retry packet. The keys used to protect
these new 0-RTT packets will not change as a result of responding to
a Retry packet. However, the data sent in these packets could be
different than what was sent earlier. Sending these new packets with
the same packet number is likely to compromise the packet protection
for those packets because the same key and nonce could be used to
protect different content. A server MAY abort the connection if it
detects that the client reset the packet number.
The connection IDs used in Initial and Retry packets exchanged
between client and server are copied to the transport parameters and
validated as described in Section 7.3.
17.3. Short Header Packets
This version of QUIC defines a single packet type that uses the short
packet header.
17.3.1. 1-RTT Packet
A 1-RTT packet uses a short packet header. It is used after the
version and 1-RTT keys are negotiated.
1-RTT 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 (8..),
}
Figure 19: 1-RTT Packet
1-RTT packets 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. A value of 1 for this bit allows
QUIC to coexist with other protocols; see [RFC7983].
Spin Bit: The third most significant bit (0x20) of byte 0 is the
latency spin bit, set as described in Section 17.4.
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 9.5 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
field, 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].
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 is protected using header protection; see
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: 1-RTT packets always include a 1-RTT protected
payload.
The header form bit and the Destination 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.
17.4. Latency Spin Bit
The latency spin bit, which is defined for 1-RTT packets
(Section 17.3.1), enables passive latency monitoring from observation
points on the network path throughout the duration of a connection.
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.
The spin bit is only present in 1-RTT packets, 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 are further discussed in
[QUIC-MANAGEABILITY].
The spin bit is an OPTIONAL feature of this version of QUIC. An
endpoint that does not support this feature MUST disable it, as
defined below.
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, in order to ensure that QUIC
connections that disable the spin bit are commonly observed on the
network. As each endpoint disables the spin bit independently, this
ensures that the spin bit signal is disabled on approximately one in
eight network paths.
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 for each network path and sets the spin bit in the
packet header to the currently stored value when a 1-RTT packet is
sent on that path. The spin value is initialized to 0 in the
endpoint for each network path. Each endpoint also remembers the
highest packet number seen from its peer on each path.
When a server receives a 1-RTT packet that increases the highest
packet number seen by the server from the client on a given network
path, it sets the spin value for that path to be equal to the spin
bit in the received packet.
When a client receives a 1-RTT packet that increases the highest
packet number seen by the client from the server on a given network
path, it sets the spin value for that path to the inverse of the spin
bit in the received packet.
An endpoint resets the spin value for a network path to 0 when
changing the connection ID being used on that network path.
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 20:
Transport Parameters {
Transport Parameter (..) ...,
}
Figure 20: Sequence of Transport Parameters
Each transport parameter is encoded as an (identifier, length, value)
tuple, as shown in Figure 21:
Transport Parameter {
Transport Parameter ID (i),
Transport Parameter Length (i),
Transport Parameter Value (..),
}
Figure 21: Transport Parameter Encoding
The Transport Parameter Length field contains the length of the
Transport Parameter Value field in bytes.
QUIC encodes transport parameters into a sequence of bytes, which is
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 can 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): This parameter is 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 maximum idle timeout is a value in
milliseconds that is encoded as an integer; see (Section 10.1).
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 10.3. 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
send this transport parameter cannot use stateless reset
(Section 10.3) 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 datagrams 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; see Section 14. 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 0x00; in server transport parameters,
this applies to streams with the least significant two bits set to
0x01.
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 0x01; in server transport parameters, this applies
to streams with the least significant two bits set to 0x00.
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 0x03; in server transport parameters, this applies to streams
with the least significant two bits set to 0x02.
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 endpoint that receives
this transport parameter is permitted to 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 endpoint that
receives this transport parameter is permitted to 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 acknowledgment 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 acknowledgment 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) on the address being used
during the handshake. An endpoint that receives this transport
parameter MUST NOT use a new local address when sending to the
address that the peer used during the handshake. This transport
parameter does not prohibit connection migration after a client
has acted on a preferred_address transport parameter. This
parameter is a zero-length value.
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. 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 preferred_address transport parameter contains an address and
port for both IPv4 and IPv6. The four-byte IPv4 Address field is
followed by the associated two-byte IPv4 Port field. This is
followed by a 16-byte IPv6 Address field and two-byte IPv6 Port
field. After address and port pairs, a Connection ID Length field
describes the length of the following Connection ID field.
Finally, a 16-byte Stateless Reset Token field includes the
stateless reset token associated with the connection ID. The
format of this transport parameter is shown in Figure 22 below.
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 sent alongside 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 a 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),
Connection ID Length (8),
Connection ID (..),
Stateless Reset Token (128),
}
Figure 22: Preferred Address Format
active_connection_id_limit (0x0e): This 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): This is 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): This is the value that 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 per-stream 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 Frames
A 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 packet to the minimum required size or to provide
protection against traffic analysis for protected packets.
PADDING frames are formatted as shown in Figure 23, which shows that
PADDING frames have no content. That is, a PADDING frame consists of
the single byte that identifies the frame as a PADDING frame.
PADDING Frame {
Type (i) = 0x00,
}
Figure 23: PADDING Frame Format
19.2. PING Frames
Endpoints can use PING frames (type=0x01) to verify that their peers
are still alive or to check reachability to the peer.
PING frames are formatted as shown in Figure 24, which shows that
PING frames have no content.
PING Frame {
Type (i) = 0x01,
}
Figure 24: PING Frame Format
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; see Section 10.1.2.
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 cumulative count
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 acknowledgments are irrevocable. Once acknowledged, a packet
remains acknowledged, even if it does not appear in a future ACK
frame. This is unlike reneging for TCP Selective Acknowledgments
(SACKs) [RFC2018].
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.
ACK frames are formatted as shown in Figure 25.
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 25: 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 encoding the acknowledgment
delay in microseconds; see Section 13.2.5. It is decoded by
multiplying the value in the field by 2 to the power of the
ack_delay_exponent transport parameter sent by the sender of the
ACK frame; see Section 18.2. Compared to simply expressing the
delay as an integer, this encoding allows for a larger range of
values within the same number of bytes, at the cost of lower
resolution.
ACK Range Count: A variable-length integer specifying the number of
ACK Range fields in the frame.
First ACK Range: A variable-length integer indicating the number of
contiguous packets preceding the Largest Acknowledged that are
being acknowledged. That is, the smallest packet acknowledged in
the range is determined by subtracting the First ACK Range value
from the Largest Acknowledged field.
ACK Ranges: Contains additional ranges of packets that are
alternately not acknowledged (Gap) and acknowledged (ACK Range);
see Section 19.3.1.
ECN Counts: The three ECN Counts; counts; see Section 19.3.2.
19.3.1. ACK Ranges
Each ACK Range consists of alternating Gap and ACK Range Length
values in descending packet number order. ACK Ranges can be
repeated. The number of Gap and ACK Range Length 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 26.
ACK Range {
Gap (i),
ACK Range Length (i),
}
Figure 26: 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 Length values 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.
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 0 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
following 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 Range Lengths 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 of the type value (that
is, type 0x03) to indicate ECN feedback and report receipt of QUIC
packets with associated ECN codepoints of ECT(0), ECT(1), or ECN-CE
in the packet's IP header. ECN Counts counts are only present when the ACK
frame type is 0x03.
When present, there are three ECN counts, as shown in Figure 27.
ECN Counts {
ECT0 Count (i),
ECT1 Count (i),
ECN-CE Count (i),
}
Figure 27: ECN Count Format
The three ECN Counts count fields are:
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.
ECN-CE Count: A variable-length integer representing the total
number of packets received with the ECN-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 Frames
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.
RESET_STREAM frames are formatted as shown in Figure 28.
RESET_STREAM Frame {
Type (i) = 0x04,
Stream ID (i),
Application Protocol Error Code (i),
Final Size (i),
}
Figure 28: 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 20.2)
that indicates why the stream is being closed.
Final Size: A variable-length integer indicating the final size of
the stream by the RESET_STREAM sender, in units of bytes; see
Section 4.5.
19.5. STOP_SENDING Frames
An endpoint uses a STOP_SENDING frame (type=0x05) to communicate that
incoming data is being discarded on receipt per 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 3.2. 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.
STOP_SENDING frames are formatted as shown in Figure 29.
STOP_SENDING Frame {
Type (i) = 0x05,
Stream ID (i),
Application Protocol Error Code (i),
}
Figure 29: 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 20.2.
19.6. CRYPTO Frames
A 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.
CRYPTO frames are formatted as shown in Figure 30.
CRYPTO Frame {
Type (i) = 0x06,
Offset (i),
Length (i),
Crypto Data (..),
}
Figure 30: 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 Frames
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.
NEW_TOKEN frames are formatted as shown in Figure 31.
NEW_TOKEN Frame {
Type (i) = 0x07,
Token Length (i),
Token (..),
}
Figure 31: 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 can use with a future Initial
packet. The token MUST NOT be empty. A client MUST treat receipt
of a NEW_TOKEN frame with an empty Token field as a connection
error of type FRAME_ENCODING_ERROR.
A client might receive multiple NEW_TOKEN frames that contain the
same token value if packets containing the frame are incorrectly
determined to be lost. Clients 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. A server 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
Type field in the STREAM frame takes the form 0b00001XXX (or the set
of values from 0x08 to 0x0f). The three low-order bits of the frame
type determine 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) indicates that the frame marks the end of the
stream. The final size of the stream is the sum of the offset and
the length of this frame.
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.
STREAM frames are formatted as shown in Figure 32.
STREAM Frame {
Type (i) = 0x08..0x0f,
Stream ID (i),
[Offset (i)],
[Length (i)],
Stream Data (..),
}
Figure 32: 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.
19.9. MAX_DATA Frames
A 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.
MAX_DATA frames are formatted as shown in Figure 33.
MAX_DATA Frame {
Type (i) = 0x10,
Maximum Data (i),
}
Figure 33: MAX_DATA Frame Format
MAX_DATA frames contain the following field:
Maximum Data: A variable-length integer indicating the maximum
amount of data that can be sent on the entire connection, in units
of bytes.
All data sent in STREAM frames counts toward this limit. The sum of
the final sizes on all streams -- including streams in terminal
states -- MUST NOT exceed the value advertised by a receiver. An
endpoint MUST terminate a connection with an error of type
FLOW_CONTROL_ERROR if it receives more data than the maximum data
value that it has sent. This includes violations of remembered
limits in Early Data; see Section 7.4.1.
19.10. MAX_STREAM_DATA Frames
A 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 3.2. 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.
MAX_STREAM_DATA frames are formatted as shown in Figure 34.
MAX_STREAM_DATA Frame {
Type (i) = 0x11,
Stream ID (i),
Maximum Stream Data (i),
}
Figure 34: MAX_STREAM_DATA Frame Format
MAX_STREAM_DATA frames contain the following fields:
Stream ID: The stream ID of the affected stream, encoded as a
variable-length integer.
Maximum Stream Data: A variable-length integer indicating the
maximum amount of data that can be sent on the identified stream,
in units of bytes.
When counting data toward this limit, an endpoint accounts for the
largest 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 an error of type FLOW_CONTROL_ERROR if it receives
more data than the largest maximum stream data that it has sent for
the affected stream. This includes violations of remembered limits
in Early Data; see Section 7.4.1.
19.11. MAX_STREAMS Frames
A MAX_STREAMS frame (type=0x12 or 0x13) informs 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.
MAX_STREAMS frames are formatted as shown in Figure 35.
MAX_STREAMS Frame {
Type (i) = 0x12..0x13,
Maximum Streams (i),
}
Figure 35: MAX_STREAMS Frame Format
MAX_STREAMS frames contain the following field:
Maximum Streams: A count of the cumulative number of streams of the
corresponding type that can be opened over the lifetime of the
connection. This value cannot exceed 2^60, as it is not possible
to encode stream IDs larger than 2^62-1. Receipt of a frame that
permits opening of a stream larger than this limit MUST be treated
as a connection error of type FRAME_ENCODING_ERROR.
Loss or reordering can cause an endpoint to receive a MAX_STREAMS
frame with a lower stream limit than was previously received.
MAX_STREAMS frames that 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 streams 3, 7,
and 11, but not stream 15. An endpoint MUST terminate a connection
with an error of type STREAM_LIMIT_ERROR if a peer opens more streams
than was permitted. This includes violations of remembered limits in
Early Data; see Section 7.4.1.
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 Frames
A sender SHOULD send a DATA_BLOCKED frame (type=0x14) when it wishes
to send data but is unable to do so 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.
DATA_BLOCKED frames are formatted as shown in Figure 36.
DATA_BLOCKED Frame {
Type (i) = 0x14,
Maximum Data (i),
}
Figure 36: DATA_BLOCKED Frame Format
DATA_BLOCKED frames contain the following field:
Maximum Data: A variable-length integer indicating the connection-
level limit at which blocking occurred.
19.13. STREAM_DATA_BLOCKED Frames
A sender SHOULD send a STREAM_DATA_BLOCKED frame (type=0x15) when it
wishes to send data but is unable to do so due to stream-level flow
control. This frame is analogous to 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.
STREAM_DATA_BLOCKED frames are formatted as shown in Figure 37.
STREAM_DATA_BLOCKED Frame {
Type (i) = 0x15,
Stream ID (i),
Maximum Stream Data (i),
}
Figure 37: STREAM_DATA_BLOCKED Frame Format
STREAM_DATA_BLOCKED frames contain the following fields:
Stream ID: A variable-length integer indicating the stream that is
blocked due to flow control.
Maximum Stream Data: A variable-length integer indicating the offset
of the stream at which the blocking occurred.
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 do so 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 is used to
indicate 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.
STREAMS_BLOCKED frames are formatted as shown in Figure 38.
STREAMS_BLOCKED Frame {
Type (i) = 0x16..0x17,
Maximum Streams (i),
}
Figure 38: STREAMS_BLOCKED Frame Format
STREAMS_BLOCKED frames contain the following field:
Maximum Streams: A variable-length integer indicating the maximum
number of streams allowed at the time the frame was sent. This
value cannot exceed 2^60, as it is not possible to encode stream
IDs larger than 2^62-1. Receipt of a frame that encodes a larger
stream ID MUST be treated as a connection error of type
STREAM_LIMIT_ERROR or FRAME_ENCODING_ERROR.
19.15. NEW_CONNECTION_ID Frames
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.
NEW_CONNECTION_ID frames are formatted as shown in Figure 39.
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 39: 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, encoded as a variable-length integer; 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 10.3.
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 handle receiving the same
NEW_CONNECTION_ID frame multiple times.
If an endpoint receives a NEW_CONNECTION_ID frame that repeats a
previously issued connection ID with a different Stateless Reset
Token field value or a different Sequence Number field value, 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 applies to connection IDs established
during connection setup and the preferred_address transport
parameter; see Section 5.1.2. The value in the Retire Prior To field
MUST be less than or equal to the value in the Sequence Number field.
Receiving a value in the Retire Prior To field that is greater than
that in the Sequence Number field 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 Frames
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 includes 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).
Retiring a connection ID invalidates the stateless reset token
associated with that connection ID.
RETIRE_CONNECTION_ID frames are formatted as shown in Figure 40.
RETIRE_CONNECTION_ID Frame {
Type (i) = 0x19,
Sequence Number (i),
}
Figure 40: RETIRE_CONNECTION_ID Frame Format
RETIRE_CONNECTION_ID frames contain the following field:
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 PROTOCOL_VIOLATION.
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 Frames
Endpoints can use PATH_CHALLENGE frames (type=0x1a) to check
reachability to the peer and for path validation during connection
migration.
PATH_CHALLENGE frames are formatted as shown in Figure 41.
PATH_CHALLENGE Frame {
Type (i) = 0x1a,
Data (64),
}
Figure 41: PATH_CHALLENGE Frame Format
PATH_CHALLENGE frames contain the following field:
Data: This 8-byte field contains arbitrary data.
Including 64 bits of entropy in a PATH_CHALLENGE frame ensures 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 value.
19.18. PATH_RESPONSE Frames
A PATH_RESPONSE frame (type=0x1b) is sent in response to a
PATH_CHALLENGE frame.
PATH_RESPONSE frames are formatted as shown in Figure 42. The format
of a PATH_RESPONSE frame is identical to that of the PATH_CHALLENGE
frame; see Section 19.17.
PATH_RESPONSE Frame {
Type (i) = 0x1b,
Data (64),
}
Figure 42: 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 frame with a 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 have not been explicitly closed, they
are implicitly closed when the connection is closed.
CONNECTION_CLOSE frames are formatted as shown in Figure 43.
CONNECTION_CLOSE Frame {
Type (i) = 0x1c..0x1d,
Error Code (i),
[Frame Type (i)],
Reason Phrase Length (i),
Reason Phrase (..),
}
Figure 43: CONNECTION_CLOSE Frame Format
CONNECTION_CLOSE frames contain the following fields:
Error Code: A variable-length integer that indicates the reason for
closing this connection. A CONNECTION_CLOSE frame of type 0x1c
uses codes from the space defined in Section 20.1. A
CONNECTION_CLOSE frame of type 0x1d uses codes defined by the
application protocol; see Section 20.2.
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: Additional diagnostic information for the closure.
This can be zero length if the sender chooses not to give details
beyond the Error Code value. This SHOULD be a UTF-8 encoded
string [RFC3629], though the frame does not carry information,
such as language tags, that would aid comprehension by any entity
other than the one that created the text.
The application-specific variant of CONNECTION_CLOSE (type 0x1d) can
only be sent using 0-RTT or 1-RTT packets; see Section 12.5. 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 Handshake packet.
19.20. HANDSHAKE_DONE Frames
The server uses a HANDSHAKE_DONE frame (type=0x1e) to signal
confirmation of the handshake to the client.
HANDSHAKE_DONE frames are formatted as shown in Figure 44, which
shows that HANDSHAKE_DONE frames have no content.
HANDSHAKE_DONE Frame {
Type (i) = 0x1e,
}
Figure 44: HANDSHAKE_DONE Frame Format
A HANDSHAKE_DONE 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.
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 extension frame types. One transport parameter can indicate
support for one or more extension frame types.
Extensions that modify or replace core protocol functionality
(including frame types) will be difficult to combine with other
extensions that 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 22.4.
20. Error Codes
QUIC transport error codes and application error codes are 62-bit
unsigned integers.
20.1. Transport Error Codes
This section lists the defined QUIC transport error codes that can be
used in a CONNECTION_CLOSE frame with a type of 0x1c. These errors
apply to the entire connection.
NO_ERROR (0x00): An endpoint uses this with CONNECTION_CLOSE to
signal that the connection is being closed abruptly in the absence
of any error.
INTERNAL_ERROR (0x01): The endpoint encountered an internal error
and cannot continue with the connection.
CONNECTION_REFUSED (0x02): The server refused to accept a new
connection.
FLOW_CONTROL_ERROR (0x03): An endpoint received more data than it
permitted in its advertised data limits; see Section 4.
STREAM_LIMIT_ERROR (0x04): An endpoint received a frame for a stream
identifier that exceeded its advertised stream limit for the
corresponding stream type.
STREAM_STATE_ERROR (0x05): An endpoint received a frame for a stream
that was not in a state that permitted that frame; see Section 3.
FINAL_SIZE_ERROR (0x06): (1) An endpoint received a STREAM frame
containing data that exceeded the previously established final
size, (2) 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 (3) an endpoint received
a STREAM frame or a RESET_STREAM frame containing a different
final size to the one already established.
FRAME_ENCODING_ERROR (0x07): 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 (0x08): An endpoint received transport
parameters that were badly formatted, included an invalid value,
omitted a mandatory transport parameter, included a forbidden
transport parameter, or were otherwise in error.
CONNECTION_ID_LIMIT_ERROR (0x09): The number of connection IDs
provided by the peer exceeds the advertised
active_connection_id_limit.
PROTOCOL_VIOLATION (0x0a): An endpoint detected an error with
protocol compliance that was not covered by more specific error
codes.
INVALID_TOKEN (0x0b): A server received a client Initial that
contained an invalid Token field.
APPLICATION_ERROR (0x0c): The application or application protocol
caused the connection to be closed.
CRYPTO_BUFFER_EXCEEDED (0x0d): An endpoint has received more data in
CRYPTO frames than it can buffer.
KEY_UPDATE_ERROR (0x0e): An endpoint detected errors in performing
key updates; see Section 6 of [QUIC-TLS].
AEAD_LIMIT_REACHED (0x0f): An endpoint has reached the
confidentiality or integrity limit for the AEAD algorithm used by
the given connection.
NO_VIABLE_PATH (0x10): An endpoint has determined that the network
path is incapable of supporting QUIC. An endpoint is unlikely to
receive a CONNECTION_CLOSE frame carrying this code except when
the path does not support a large enough MTU.
CRYPTO_ERROR (0x0100-0x01ff): 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 cryptographic handshake are
described in Section 4.8 of [QUIC-TLS].
See Section 22.5 for details on 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.
20.2. Application Protocol Error Codes
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
The goal of QUIC is to provide a secure transport connection.
Section 21.1 provides an overview of those properties; subsequent
sections discuss constraints and caveats regarding these properties,
including descriptions of known attacks and countermeasures.
21.1. 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 implementers 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 ability to read packets from the
network, while active attackers also have the ability to write
packets into the network. However, a passive attack could 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. An on-path attacker can read, modify, or remove
any packet it observes such that the packet no longer reaches its
destination, while an off-path attacker observes the packets but
cannot prevent the original packet from reaching its intended
destination. Both types of attackers can also transmit arbitrary
packets. This definition differs from that of Section 3.5 of
[SEC-CONS] in that an off-path attacker is able to observe packets.
Properties of the handshake, protected packets, and connection
migration are considered separately.
21.1.1. 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, or the authentication of the
participating peers, affects other security guarantees provided by
QUIC that depend on those 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.
21.1.1.1. 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 can observe packets.
Prior to address validation, endpoints are limited in what they are
able to send. Endpoints cannot send data toward an unvalidated
address in excess of three times the data received from that address.
| Note: The anti-amplification limit only applies when an
| endpoint responds to packets received from an unvalidated
| address. The anti-amplification limit does not apply to
| clients when establishing a new connection or when initiating
| connection migration.
21.1.1.2. 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 that 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.
21.1.1.3. 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.
21.1.1.4. 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 salt)
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.
21.1.2. Protected Packets
Packet protection (Section 12.1) applies authenticated encryption to
all packets except Version Negotiation packets, though Initial and
Retry packets have limited protection due to the use of version-
specific keying material; 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.
An attacker that 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
that possess the key material established during the handshake; see
Sections 7 and 21.1.1. 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, other than Initial packets.
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.
21.1.3. 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 under various types of DoS attacks.
21.1.3.1. 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
* 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.
QUIC aims to constrain the capabilities of an on-path attacker as
follows:
1. An on-path attacker can prevent the use of a path for a
connection, causing the connection 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.
21.1.3.2. 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 packets sent by endpoints
* Delay packets
* Drop packets
* Reorder original packets
An off-path attacker can create modified copies of packets that it
has observed and inject those copies 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 inject a modified copy of 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. In particular, an attacker can cause an endpoint
to lose its NAT binding and then obtain the same port for use with
its own traffic.
QUIC aims to constrain the capabilities of an off-path attacker as
follows:
1. An off-path attacker can race packets and attempt to become a
"limited" on-path attacker.
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.
21.1.3.3. 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:
* 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.
QUIC aims to constrain the capabilities of a limited off-path
attacker as follows:
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.
21.2. 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 14.2.1), 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 10.3),
which relies on the token being kept secret until it is used.
During the creation of a connection, QUIC only provides protection
against attacks 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.
The Source and Destination Connection ID fields are the primary means
of protection against an off-path attack during the handshake; see
Section 8.1. These are required to match those set by a peer.
Except for Initial and Stateless Resets, 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 a 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 can be
identified and discarded using other methods, but no specific method
is mandated in this document.
21.3. 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 can 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.
21.4. 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 10.2.
21.5. Request Forgery Attacks
A request forgery attack occurs where an endpoint causes its peer to
issue a request towards a victim, with the request controlled by the
endpoint. Request forgery attacks aim to provide an attacker with
access to capabilities of its peer that might otherwise be
unavailable to the attacker. For a networking protocol, a request
forgery attack is often used to exploit any implicit authorization
conferred on the peer by the victim due to the peer's location in the
network.
For request forgery to be effective, an attacker needs to be able to
influence what packets the peer sends and where these packets are
sent. If an attacker can target a vulnerable service with a
controlled payload, that service might perform actions that are
attributed to the attacker's peer but are decided by the attacker.
For example, cross-site request forgery [CSRF] exploits on the Web
cause a client to issue requests that include authorization cookies
[COOKIE], allowing one site access to information and actions that
are intended to be restricted to a different site.
As QUIC runs over UDP, the primary attack modality of concern is one
where an attacker can select the address to which its peer sends UDP
datagrams and can control some of the unprotected content of those
packets. As much of the data sent by QUIC endpoints is protected,
this includes control over ciphertext. An attack is successful if an
attacker can cause a peer to send a UDP datagram to a host that will
perform some action based on content in the datagram.
This section discusses ways in which QUIC might be used for request
forgery attacks.
This section also describes limited countermeasures that can be
implemented by QUIC endpoints. These mitigations can be employed
unilaterally by a QUIC implementation or deployment, without
potential targets for request forgery attacks taking action.
However, these countermeasures could be insufficient if UDP-based
services do not properly authorize requests.
Because the migration attack described in Section 21.5.4 is quite
powerful and does not have adequate countermeasures, QUIC server
implementations should assume that attackers can cause them to
generate arbitrary UDP payloads to arbitrary destinations. QUIC
servers SHOULD NOT be deployed in networks that do not deploy ingress
filtering [BCP38] and also have inadequately secured UDP endpoints.
Although it is not generally possible to ensure that clients are not
co-located with vulnerable endpoints, this version of QUIC does not
allow servers to migrate, thus preventing spoofed migration attacks
on clients. Any future extension that allows server migration MUST
also define countermeasures for forgery attacks.
21.5.1. Control Options for Endpoints
QUIC offers some opportunities for an attacker to influence or
control where its peer sends UDP datagrams:
* initial connection establishment (Section 7), where a server is
able to choose where a client sends datagrams -- for example, by
populating DNS records;
* preferred addresses (Section 9.6), where a server is able to
choose where a client sends datagrams;
* spoofed connection migrations (Section 9.3.1), where a client is
able to use source address spoofing to select where a server sends
subsequent datagrams; and
* spoofed packets that cause a server to send a Version Negotiation
packet (Section 21.5.5).
In all cases, the attacker can cause its peer to send datagrams to a
victim that might not understand QUIC. That is, these packets are
sent by the peer prior to address validation; see Section 8.
Outside of the encrypted portion of packets, QUIC offers an endpoint
several options for controlling the content of UDP datagrams that its
peer sends. The Destination Connection ID field offers direct
control over bytes that appear early in packets sent by the peer; see
Section 5.1. The Token field in Initial packets offers a server
control over other bytes of Initial packets; see Section 17.2.2.
There are no measures in this version of QUIC to prevent indirect
control over the encrypted portions of packets. It is necessary to
assume that endpoints are able to control the contents of frames that
a peer sends, especially those frames that convey application data,
such as STREAM frames. Though this depends to some degree on details
of the application protocol, some control is possible in many
protocol usage contexts. As the attacker has access to packet
protection keys, they are likely to be capable of predicting how a
peer will encrypt future packets. Successful control over datagram
content then only requires that the attacker be able to predict the
packet number and placement of frames in packets with some amount of
reliability.
This section assumes that limiting control over datagram content is
not feasible. The focus of the mitigations in subsequent sections is
on limiting the ways in which datagrams that are sent prior to
address validation can be used for request forgery.
21.5.2. Request Forgery with Client Initial Packets
An attacker acting as a server can choose the IP address and port on
which it advertises its availability, so Initial packets from clients
are assumed to be available for use in this sort of attack. The
address validation implicit in the handshake ensures that -- for a
new connection -- a client will not send other types of packets to a
destination that does not understand QUIC or is not willing to accept
a QUIC connection.
Initial packet protection (Section 5.2 of [QUIC-TLS]) makes it
difficult for servers to control the content of Initial packets sent
by clients. A client choosing an unpredictable Destination
Connection ID ensures that servers are unable to control any of the
encrypted portion of Initial packets from clients.
However, the Token field is open to server control and does allow a
server to use clients to mount request forgery attacks. The use of
tokens provided with the NEW_TOKEN frame (Section 8.1.3) offers the
only option for request forgery during connection establishment.
Clients, however, are not obligated to use the NEW_TOKEN frame.
Request forgery attacks that rely on the Token field can be avoided
if clients send an empty Token field when the server address has
changed from when the NEW_TOKEN frame was received.
Clients could avoid using NEW_TOKEN if the server address changes.
However, not including a Token field could adversely affect
performance. Servers could rely on NEW_TOKEN to enable the sending
of data in excess of the three-times limit on sending data; see
Section 8.1. In particular, this affects cases where clients use
0-RTT to request data from servers.
Sending a Retry packet (Section 17.2.5) offers a server the option to
change the Token field. After sending a Retry, the server can also
control the Destination Connection ID field of subsequent Initial
packets from the client. This also might allow indirect control over
the encrypted content of Initial packets. However, the exchange of a
Retry packet validates the server's address, thereby preventing the
use of subsequent Initial packets for request forgery.
21.5.3. Request Forgery with Preferred Addresses
Servers can specify a preferred address, which clients then migrate
to after confirming the handshake; see Section 9.6. The Destination
Connection ID field of packets that the client sends to a preferred
address can be used for request forgery.
A client MUST NOT send non-probing frames to a preferred address
prior to validating that address; see Section 8. This greatly
reduces the options that a server has to control the encrypted
portion of datagrams.
This document does not offer any additional countermeasures that are
specific to the use of preferred addresses and can be implemented by
endpoints. The generic measures described in Section 21.5.6 could be
used as further mitigation.
21.5.4. Request Forgery with Spoofed Migration
Clients are able to present a spoofed source address as part of an
apparent connection migration to cause a server to send datagrams to
that address.
The Destination Connection ID field in any packets that a server
subsequently sends to this spoofed address can be used for request
forgery. A client might also be able to influence the ciphertext.
A server that only sends probing packets (Section 9.1) to an address
prior to address validation provides an attacker with only limited
control over the encrypted portion of datagrams. However,
particularly for NAT rebinding, this can adversely affect
performance. If the server sends frames carrying application data,
an attacker might be able to control most of the content of
datagrams.
This document does not offer specific countermeasures that can be
implemented by endpoints, aside from the generic measures described
in Section 21.5.6. However, countermeasures for address spoofing at
the network level -- in particular, ingress filtering [BCP38] -- are
especially effective against attacks that use spoofing and originate
from an external network.
21.5.5. Request Forgery with Version Negotiation
Clients that are able to present a spoofed source address on a packet
can cause a server to send a Version Negotiation packet
(Section 17.2.1) to that address.
The absence of size restrictions on the connection ID fields for
packets of an unknown version increases the amount of data that the
client controls from the resulting datagram. The first byte of this
packet is not under client control and the next four bytes are zero,
but the client is able to control up to 512 bytes starting from the
fifth byte.
No specific countermeasures are provided for this attack, though
generic protections (Section 21.5.6) could apply. In this case,
ingress filtering [BCP38] is also effective.
21.5.6. Generic Request Forgery Countermeasures
The most effective defense against request forgery attacks is to
modify vulnerable services to use strong authentication. However,
this is not always something that is within the control of a QUIC
deployment. This section outlines some other steps that QUIC
endpoints could take unilaterally. These additional steps are all
discretionary because, depending on circumstances, they could
interfere with or prevent legitimate uses.
Services offered over loopback interfaces often lack proper
authentication. Endpoints MAY prevent connection attempts or
migration to a loopback address. Endpoints SHOULD NOT allow
connections or migration to a loopback address if the same service
was previously available at a different interface or if the address
was provided by a service at a non-loopback address. Endpoints that
depend on these capabilities could offer an option to disable these
protections.
Similarly, endpoints could regard a change in address to a link-local
address [RFC4291] or an address in a private-use range [RFC1918] from
a global, unique-local [RFC4193], or non-private address as a
potential attempt at request forgery. Endpoints could refuse to use
these addresses entirely, but that carries a significant risk of
interfering with legitimate uses. Endpoints SHOULD NOT refuse to use
an address unless they have specific knowledge about the network
indicating that sending datagrams to unvalidated addresses in a given
range is not safe.
Endpoints MAY choose to reduce the risk of request forgery by not
including values from NEW_TOKEN frames in Initial packets or by only
sending probing frames in packets prior to completing address
validation. Note that this does not prevent an attacker from using
the Destination Connection ID field for an attack.
Endpoints are not expected to have specific information about the
location of servers that could be vulnerable targets of a request
forgery attack. However, it might be possible over time to identify
specific UDP ports that are common targets of attacks or particular
patterns in datagrams that are used for attacks. Endpoints MAY
choose to avoid sending datagrams to these ports or not send
datagrams that match these patterns prior to validating the
destination address. Endpoints MAY retire connection IDs containing
patterns known to be problematic without using them.
| Note: Modifying endpoints to apply these protections is more
| efficient than deploying network-based protections, as
| endpoints do not need to perform any additional processing when
| sending to an address that has been validated.
21.6. 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.
21.7. Stream Fragmentation and Reassembly Attacks
An adversarial sender might intentionally not send portions of the
stream data, causing the receiver to commit resources for the unsent
data. This could cause a disproportionate receive buffer memory
commitment and/or the creation of a large and inefficient data
structure at the receiver.
An adversarial receiver might intentionally not acknowledge packets
containing stream data in an attempt 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
overcommit memory and advertise flow control offsets in the aggregate
that exceed actual available memory. The overcommitment strategy can
lead to better performance when endpoints are well behaved, but
renders endpoints vulnerable to the stream fragmentation attack.
QUIC deployments SHOULD provide mitigations for stream fragmentation
attacks. Mitigations could consist of avoiding overcommitting
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 of these.
21.8. Stream Commitment Attack
An adversarial endpoint can open a large number 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 can 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 updated by any received MAX_STREAMS frames, as
explained in Section 4.6. 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 a large number of streams.
21.9. Peer Denial of Service
QUIC and TLS both contain frames or messages that have legitimate
uses in some contexts, but these frames or messages 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.
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.
21.10. Explicit Congestion Notification Attacks
An on-path attacker could manipulate the value of ECN fields in the
IP header to influence the sender's rate. [RFC3168] discusses
manipulations and their effects in more detail.
A limited on-path attacker can duplicate and send packets with
modified ECN fields to affect the sender's rate. If duplicate
packets are discarded by a receiver, an attacker will need to race
the duplicate packet against the original to be successful in this
attack. Therefore, QUIC endpoints ignore the ECN field in an IP
packet unless at least one QUIC packet in that IP packet is
successfully processed; see Section 13.4.
21.11. 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 of service, endpoints that share a
static key for stateless resets (see Section 10.3.2) 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.
More generally, servers MUST NOT generate a stateless reset if a
connection with the corresponding connection ID could be active on
any endpoint using the same static key.
In the case of a cluster that uses dynamic load balancing, it is
possible that a change in load-balancer configuration could occur
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 of the packet being routed to the
correct instance, it is better to send a stateless reset than wait
for the connection to time out. However, this is acceptable only if
the routing cannot be influenced by an attacker.
21.12. Version Downgrade
This document defines QUIC Version Negotiation packets (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.
21.13. Targeted Attacks by Routing
Deployments should limit the ability of an attacker to target a new
connection to a particular server instance. Ideally, routing
decisions are made independently of client-selected values, including
addresses. Once an instance is selected, a connection ID can be
selected so that later packets are routed to the same instance.
21.14. Traffic Analysis
The length of QUIC packets can reveal information about the length of
the content of those packets. The PADDING frame is provided so that
endpoints have some ability to obscure the length of packet content;
see Section 19.1.
Defeating traffic analysis is challenging and the subject of active
research. Length is not the only way that information might leak.
Endpoints might also reveal sensitive information through other side
channels, such as the timing of packets.
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 registrations 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.
Provisional registrations require Expert Review, as defined in
Section 4.5 of [RFC8126]. The designated expert or experts 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 the last update to the registration.
Change Controller: The entity that is responsible for the definition
of 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 requests for 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.
The use of the first unassigned codepoint is reserved for allocation
using the Standards Action policy; see Section 4.9 of [RFC8126]. The
early codepoint assignment process [EARLY-ASSIGN] can be used for
these values.
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
requested codepoint as part of the registration. IANA MUST allocate
the selected codepoint if the codepoint is unassigned and the
requirements of the registration policy are met.
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 a 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
experts. The experts 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 experts are also 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 review and consultation 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.
22.1.4. Permanent Registrations
Permanent registrations in QUIC registries use the Specification
Required policy (Section 4.6 of [RFC8126]), unless otherwise
specified. The designated expert or experts verify that a
specification exists and is readily accessible. Experts 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 registry MAY identify a range of codepoints where
registrations are governed by a different registration policy. For
instance, the "QUIC Frame Types" registry (Section 22.4) has a
stricter policy 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 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 a change controller of the IETF and a contact of the QUIC
Working Group (quic@ietf.org).
22.2. QUIC Versions Registry
IANA has added a registry for "QUIC Versions" under a "QUIC" heading.
The "QUIC Versions" registry governs a 32-bit space; see Section 15.
This registry follows the registration policy from Section 22.1.
Permanent registrations in this registry are assigned using the
Specification Required policy (Section 4.6 of [RFC8126]).
The codepoint of 0x00000001 for the protocol is assigned with
permanent status to the protocol defined in this document. The
codepoint of 0x00000000 is permanently reserved; the note for this
codepoint indicates that this version is reserved for version
negotiation.
All codepoints that follow the pattern 0x?a?a?a?a are reserved, MUST
NOT be assigned by IANA, and MUST NOT appear in the listing of
assigned values.
22.3. QUIC Transport Parameters Registry
IANA 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 (Section 4.6 of [RFC8126]), except for
values between 0x00 and 0x3f (in hexadecimal), inclusive, which are
assigned using Standards Action or IESG Approval as defined in
Sections 4.9 and 4.10 of [RFC8126].
In addition to the fields listed in Section 22.1.1, permanent
registrations in this registry MUST include the following field:
Parameter Name: A short mnemonic for the parameter.
The initial contents of this registry are shown in Table 6.
+=======+=====================================+===============+
| 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 Registry Entries
Each value of the form "31 * N + 27" for integer values of N (that
is, 27, 58, 89, ...) are reserved; these values MUST NOT be assigned
by IANA and MUST NOT appear in the listing of assigned values.
22.4. QUIC Frame Types Registry
IANA has added a registry for "QUIC Frame Types" under a "QUIC"
heading.
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 (Section 4.6 of [RFC8126]), except for
values between 0x00 and 0x3f (in hexadecimal), inclusive, which are
assigned using Standards Action or IESG Approval as defined in
Sections 4.9 and 4.10 of [RFC8126].
In addition to the fields listed in Section 22.1.1, permanent
registrations in this registry MUST include the following field:
Frame Type 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 22.3. Specifications for
permanent registrations also need to describe the format and assigned
semantics of any fields in the frame.
The initial contents of this registry are tabulated in Table 3. Note
that the registry does not include the "Pkts" and "Spec" columns from
Table 3.
22.5. QUIC Transport Error Codes Registry
IANA 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 ranges that are governed by different
policies. Permanent registrations in this registry are assigned
using the Specification Required policy (Section 4.6 of [RFC8126]),
except for values between 0x00 and 0x3f (in hexadecimal), inclusive,
which are assigned using Standards Action or IESG Approval as defined
in Sections 4.9 and 4.10 of [RFC8126].
In addition to the fields listed 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.
+=======+===========================+================+==============+
|Value | Code |Description |Specification |
+=======+===========================+================+==============+
|0x00 | NO_ERROR |No error |Section 20 |
+-------+---------------------------+----------------+--------------+
|0x01 | INTERNAL_ERROR |Implementation |Section 20 |
| | |error | |
+-------+---------------------------+----------------+--------------+
|0x02 | CONNECTION_REFUSED |Server refuses a|Section 20 |
| | |connection | |
+-------+---------------------------+----------------+--------------+
|0x03 | FLOW_CONTROL_ERROR |Flow control |Section 20 |
| | |error | |
+-------+---------------------------+----------------+--------------+
|0x04 | STREAM_LIMIT_ERROR |Too many streams|Section 20 |
| | |opened | |
+-------+---------------------------+----------------+--------------+
|0x05 | STREAM_STATE_ERROR |Frame received |Section 20 |
| | |in invalid | |
| | |stream state | |
+-------+---------------------------+----------------+--------------+
|0x06 | FINAL_SIZE_ERROR |Change to final |Section 20 |
| | |size | |
+-------+---------------------------+----------------+--------------+
|0x07 | FRAME_ENCODING_ERROR |Frame encoding |Section 20 |
| | |error | |
+-------+---------------------------+----------------+--------------+
|0x08 | TRANSPORT_PARAMETER_ERROR |Error in |Section 20 |
| | |transport | |
| | |parameters | |
+-------+---------------------------+----------------+--------------+
|0x09 | CONNECTION_ID_LIMIT_ERROR |Too many |Section 20 |
| | |connection IDs | |
| | |received | |
+-------+---------------------------+----------------+--------------+
|0x0a | PROTOCOL_VIOLATION |Generic protocol|Section 20 |
| | |violation | |
+-------+---------------------------+----------------+--------------+
|0x0b | INVALID_TOKEN |Invalid Token |Section 20 |
| | |received | |
+-------+---------------------------+----------------+--------------+
|0x0c | APPLICATION_ERROR |Application |Section 20 |
| | |error | |
+-------+---------------------------+----------------+--------------+
|0x0d | CRYPTO_BUFFER_EXCEEDED |CRYPTO data |Section 20 |
| | |buffer | |
| | |overflowed | |
+-------+---------------------------+----------------+--------------+
|0x0e | KEY_UPDATE_ERROR |Invalid packet |Section 20 |
| | |protection | |
| | |update | |
+-------+---------------------------+----------------+--------------+
|0x0f | AEAD_LIMIT_REACHED |Excessive use of|Section 20 |
| | |packet | |
| | |protection keys | |
+-------+---------------------------+----------------+--------------+
|0x10 | NO_VIABLE_PATH |No viable |Section 20 |
| | |network path | |
| | |exists | |
+-------+---------------------------+----------------+--------------+
|0x0100-| CRYPTO_ERROR |TLS alert code |Section 20 |
|0x01ff | | | |
+-------+---------------------------+----------------+--------------+
Table 7: Initial QUIC Transport Error Codes Registry Entries
23. References
23.1. Normative References
[BCP38] Ferguson, P. and D. Senie, "Network Ingress Filtering:
Defeating Denial of Service Attacks which employ IP Source
Address Spoofing", BCP 38, RFC 2827, May 2000.
<https://www.rfc-editor.org/info/bcp38>
[DPLPMTUD] Fairhurst, G., Jones, T., Tüxen, M., Rüngeler, I., and T.
Völker, "Packetization Layer Path MTU Discovery for
Datagram Transports", RFC 8899, DOI 10.17487/RFC8899,
September 2020, <https://www.rfc-editor.org/info/rfc8899>.
[EARLY-ASSIGN]
Cotton, M., "Early IANA Allocation of Standards Track Code
Points", BCP 100, RFC 7120, DOI 10.17487/RFC7120, January
2014, <https://www.rfc-editor.org/info/rfc7120>.
[IPv4] Postel, J., "Internet Protocol", STD 5, RFC 791,
DOI 10.17487/RFC0791, September 1981,
<https://www.rfc-editor.org/info/rfc791>.
[QUIC-INVARIANTS]
Thomson, M., "Version-Independent Properties of QUIC",
RFC 8999, DOI 10.17487/RFC8999, May 2021,
<https://www.rfc-editor.org/info/rfc8999>.
[QUIC-RECOVERY]
Iyengar, J., Ed. and I. Swett, Ed., "QUIC Loss Detection
and Congestion Control", RFC 9002, DOI 10.17487/RFC9002,
May 2021, <https://www.rfc-editor.org/info/rfc9002>.
[QUIC-TLS] Thomson, M., Ed. and S. Turner, Ed., "Using TLS to Secure
QUIC", RFC 9001, DOI 10.17487/RFC9001, May 2021,
<https://www.rfc-editor.org/info/rfc9001>.
[RFC1191] Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191,
DOI 10.17487/RFC1191, November 1990,
<https://www.rfc-editor.org/info/rfc1191>.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[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>.
[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>.
[UDP] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
DOI 10.17487/RFC0768, August 1980,
<https://www.rfc-editor.org/info/rfc768>.
23.2. Informative References
[AEAD] McGrew, D., "An Interface and Algorithms for Authenticated
Encryption", RFC 5116, DOI 10.17487/RFC5116, January 2008,
<https://www.rfc-editor.org/info/rfc5116>.
[ALPN] 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>.
[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>.
[COOKIE] Barth, A., "HTTP State Management Mechanism", RFC 6265,
DOI 10.17487/RFC6265, April 2011,
<https://www.rfc-editor.org/info/rfc6265>.
[CSRF] Barth, A., Jackson, C., and J. Mitchell, "Robust defenses
for cross-site request forgery", Proceedings of the 15th
ACM conference on Computer and communications security -
CCS '08, DOI 10.1145/1455770.1455782, 2008,
<https://doi.org/10.1145/1455770.1455782>.
[EARLY-DESIGN]
Roskind, J., "QUIC: Multiplexed Stream Transport Over
UDP", 2 December 2013, <https://docs.google.com/document/
d/1RNHkx_VvKWyWg6Lr8SZ-saqsQx7rFV-ev2jRFUoVD34/
edit?usp=sharing>.
[GATEWAY] Hätönen, S., Nyrhinen, A., Eggert, L., Strowes, S.,
Sarolahti, P., and M. Kojo, "An experimental study of home
gateway characteristics", Proceedings of the 10th ACM
SIGCOMM conference on Internet measurement - IMC ‘10, '10,
DOI 10.1145/1879141.1879174, November 2010,
<https://doi.org/10.1145/1879141.1879174>.
[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>.
[IPv6] 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>.
[QUIC-MANAGEABILITY]
Kuehlewind, M. and B. Trammell, "Manageability of the QUIC
Transport Protocol", Work in Progress, Internet-Draft,
draft-ietf-quic-manageability-11, 21 April 2021,
<https://tools.ietf.org/html/draft-ietf-quic-
manageability-11>.
[RANDOM] 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>.
[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>.
[RFC1918] Rekhter, Y., Moskowitz, B., Karrenberg, D., de Groot, G.
J., and E. Lear, "Address Allocation for Private
Internets", BCP 5, RFC 1918, DOI 10.17487/RFC1918,
February 1996, <https://www.rfc-editor.org/info/rfc1918>.
[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>.
[RFC3449] Balakrishnan, H., Padmanabhan, V., Fairhurst, G., and M.
Sooriyabandara, "TCP Performance Implications of Network
Path Asymmetry", BCP 69, RFC 3449, DOI 10.17487/RFC3449,
December 2002, <https://www.rfc-editor.org/info/rfc3449>.
[RFC4193] Hinden, R. and B. Haberman, "Unique Local IPv6 Unicast
Addresses", RFC 4193, DOI 10.17487/RFC4193, October 2005,
<https://www.rfc-editor.org/info/rfc4193>.
[RFC4291] Hinden, R. and S. Deering, "IP Version 6 Addressing
Architecture", RFC 4291, DOI 10.17487/RFC4291, February
2006, <https://www.rfc-editor.org/info/rfc4291>.
[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>.
[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>.
[RFC7983] Petit-Huguenin, M. and G. Salgueiro, "Multiplexing Scheme
Updates for Secure Real-time Transport Protocol (SRTP)
Extension for Datagram Transport Layer Security (DTLS)",
RFC 7983, DOI 10.17487/RFC7983, September 2016,
<https://www.rfc-editor.org/info/rfc7983>.
[RFC8087] Fairhurst, G. and M. Welzl, "The Benefits of Using
Explicit Congestion Notification (ECN)", RFC 8087,
DOI 10.17487/RFC8087, March 2017,
<https://www.rfc-editor.org/info/rfc8087>.
[RFC8981] Gont, F., Krishnan, S., Narten, T., and R. Draves,
"Temporary Address Extensions for Stateless Address
Autoconfiguration in IPv6", RFC 8981,
DOI 10.17487/RFC8981, February 2021,
<https://www.rfc-editor.org/info/rfc8981>.
[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 - the low
bandwidth, yet greedy and poisonous HTTP client!", June
2009, <https://web.archive.org/web/20150315054838/
http://ha.ckers.org/slowloris/>.
Appendix A. Pseudocode
The pseudocode in this section describes sample algorithms. These
algorithms are intended to be correct and clear, rather than being
optimally performant.
The pseudocode segments in this section are licensed as Code
Components; see the Copyright Notice.
A.1. Sample Variable-Length Integer Decoding
The pseudocode in Figure 45 shows how a variable-length integer can
be read from a stream of bytes. The function ReadVarint takes a
single argument -- a sequence of bytes, which can be read in network
byte order.
ReadVarint(data):
// The length of variable-length integers is encoded in the
// first two bits of the first byte.
v = data.next_byte()
prefix = v >> 6
length = 1 << prefix
// Once the length is known, remove these bits and read any
// remaining bytes.
v = v & 0x3f
repeat length-1 times:
v = (v << 8) + data.next_byte()
return v
Figure 45: Sample Variable-Length Integer Decoding Algorithm
For example, the eight-byte sequence 0xc2197c5eff14e88c decodes to
the decimal value 151,288,809,941,952,652; the four-byte sequence
0x9d7f3e7d decodes to 494,878,333; the two-byte sequence 0x7bbd
decodes to 15,293; and the single byte 0x25 decodes to 37 (as does
the two-byte sequence 0x4025).
A.2. Sample Packet Number Encoding Algorithm
The pseudocode in Figure 46 shows how an implementation can select an
appropriate size for packet number encodings.
The EncodePacketNumber function takes two arguments:
* full_pn is the full packet number of the packet being sent.
* largest_acked is the largest packet number that has been
acknowledged by the peer in the current packet number space, if
any.
EncodePacketNumber(full_pn, largest_acked):
// The number of bits must be at least one more
// than the base-2 logarithm of the number of contiguous
// unacknowledged packet numbers, including the new packet.
if largest_acked is None:
num_unacked = full_pn + 1
else:
num_unacked = full_pn - largest_acked
min_bits = log(num_unacked, 2) + 1
num_bytes = ceil(min_bits / 8)
// Encode the integer value and truncate to
// the num_bytes least significant bytes.
return encode(full_pn, num_bytes)
Figure 46: Sample Packet Number Encoding Algorithm
For example, if an endpoint has received an acknowledgment for packet
0xabe8b3 and is sending a packet with a number of 0xac5c02, there are
29,519 (0x734f) outstanding packet numbers. In order to represent at
least twice this range (59,038 packets, or 0xe69e), 16 bits are
required.
In the same state, sending a packet with a number of 0xace8fe uses
the 24-bit encoding, because at least 18 bits are required to
represent twice the range (131,222 packets, or 0x020096).
A.3. Sample Packet Number Decoding Algorithm
The pseudocode in Figure 47 includes an example algorithm for
decoding packet numbers after header protection has been removed.
The DecodePacketNumber function takes three arguments:
* largest_pn is the largest packet number that has been successfully
processed in the current packet number space.
* truncated_pn is the value of the Packet Number field.
* pn_nbits is the number of bits in the Packet Number field (8, 16,
24, or 32).
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 cannot 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 47: Sample Packet Number Decoding Algorithm
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.
A.4. 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 --
ECT(0) by default; 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.
The testing period runs for a number of packets or a limited time, 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 suggests limiting this to ten
packets or three times the PTO.
After the testing period ends, the ECN state for the path becomes
"unknown". From the "unknown" state, successful validation of the
ECN counts in an ACK frame (see Section 13.4.2.1) 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 ECN-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.
Contributors
The original design and rationale behind this protocol draw
significantly from work by Jim Roskind [EARLY-DESIGN].
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)
* Lars Eggert
* Lucas Pardue
* Magnus Westerlund
* Marten Seemann
* Martin Duke
* Mike Bishop
* Mikkel Fahnøe Jørgensen
* Mirja Kühlewind
* Nick Banks
* 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