Martin Thomson · IETF Datatracker

RFC 9000 QUIC Transport Protocol May 2021
Iyengar & Thomson Standards Track [Page]

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 publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.

Table of Contents

1. 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:

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 262-1) 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.

Table 1: Stream ID Types
Bits Stream Type
0x00 Client-Initiated, Bidirectional
0x01 Server-Initiated, Bidirectional
0x02 Client-Initiated, Unidirectional
0x03 Server-Initiated, Unidirectional

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.

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.

Table 2: Possible Mapping of Stream States to HTTP/2
Sending Part Receiving Part Composite State
No Stream / Ready No Stream / Recv (*1) idle
Ready / Send / Data Sent Recv / Size Known open
Ready / Send / Data Sent Data Recvd / Data Read half-closed (remote)
Ready / Send / Data Sent Reset Recvd / Reset Read half-closed (remote)
Data Recvd Recv / Size Known half-closed (local)
Reset Sent / Reset Recvd Recv / Size Known half-closed (local)
Reset Sent / Reset Recvd Data Recvd / Data Read closed
Reset Sent / Reset Recvd Reset Recvd / Reset Read closed
Data Recvd Data Recvd / Data Read closed
Data Recvd Reset Recvd / Reset Read closed

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 260, 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.

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:

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].

10.3. 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 262-1. 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 262-1, 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.

Table 3: Frame Types
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

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.

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