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draft-ietf-quic-transport.md
---
title: "QUIC: A UDP-Based Multiplexed and Secure Transport"
abbrev: QUIC Transport Protocol
docname: draft-ietf-quic-transport-latest
date: {DATE}
category: std
ipr: trust200902
area: Transport
workgroup: QUIC
stand_alone: yes
pi: [toc, sortrefs, symrefs, docmapping]
author:
-
ins: J. Iyengar
name: Jana Iyengar
org: Fastly
email: jri.ietf@gmail.com
role: editor
-
ins: M. Thomson
name: Martin Thomson
org: Mozilla
email: mt@lowentropy.net
role: editor
normative:
QUIC-RECOVERY:
title: "QUIC Loss Detection and Congestion Control"
date: {DATE}
seriesinfo:
Internet-Draft: draft-ietf-quic-recovery-latest
author:
-
ins: J. Iyengar
name: Jana Iyengar
org: Fastly
role: editor
-
ins: I. Swett
name: Ian Swett
org: Google
role: editor
QUIC-TLS:
title: "Using Transport Layer Security (TLS) to Secure QUIC"
date: {DATE}
seriesinfo:
Internet-Draft: draft-ietf-quic-tls-latest
author:
-
ins: M. Thomson
name: Martin Thomson
org: Mozilla
role: editor
-
ins: S. Turner
name: Sean Turner
org: sn3rd
role: editor
informative:
QUIC-INVARIANTS:
title: "Version-Independent Properties of QUIC"
date: {DATE}
seriesinfo:
Internet-Draft: draft-ietf-quic-invariants-latest
author:
-
ins: M. Thomson
name: Martin Thomson
org: Mozilla
EARLY-DESIGN:
title: "QUIC: Multiplexed Transport Over UDP"
author:
- ins: J. Roskind
date: 2013-12-02
target: "https://goo.gl/dMVtFi"
SLOWLORIS:
title: "Welcome to Slowloris..."
author:
- ins: R. RSnake Hansen
date: 2009-06
target:
"https://web.archive.org/web/20150315054838/http://ha.ckers.org/slowloris/"
--- abstract
This document defines the core of the QUIC transport protocol. Accompanying
documents describe QUIC's loss detection and congestion control and the use of
TLS for key negotiation.
--- note_Note_to_Readers
Discussion of this draft takes place on the QUIC working group mailing list
(quic@ietf.org), which is archived at
\<https://mailarchive.ietf.org/arch/search/?email_list=quic\>.
Working Group information can be found at \<https://github.com/quicwg\>; source
code and issues list for this draft can be found at
\<https://github.com/quicwg/base-drafts/labels/-transport\>.
--- middle
# Introduction
QUIC is a multiplexed and secure general-purpose transport protocol that
provides:
* Stream multiplexing
* Stream and connection-level flow control
* Low-latency connection establishment
* Connection migration and resilience to NAT rebinding
* Authenticated and encrypted header and payload
QUIC uses UDP as a substrate to avoid requiring changes to legacy client
operating systems and middleboxes. QUIC authenticates all of its headers and
encrypts most of the data it exchanges, including its signaling, to avoid
incurring a dependency on middleboxes.
## Document Structure
This document describes the core QUIC protocol and is structured as follows.
* Streams are the basic service abstraction that QUIC provides.
- {{streams}} describes core concepts related to streams,
- {{stream-states}} provides a reference model for stream states, and
- {{flow-control}} outlines the operation of flow control.
* Connections are the context in which QUIC endpoints communicate.
- {{connections}} describes core concepts related to connections,
- {{version-negotiation}} describes version negotiation,
- {{handshake}} details the process for establishing connections,
- {{address-validation}} specifies critical denial of service mitigation
mechanisms,
- {{migration}} describes how endpoints migrate a connection to a new
network path,
- {{termination}} lists the options for terminating an open connection, and
- {{error-handling}} provides general guidance for error handling.
* Packets and frames are the basic unit used by QUIC to communicate.
- {{packets-frames}} describes concepts related to packets and frames,
- {{packetization}} defines models for the transmission, retransmission, and
acknowledgement of data, and
- {{packet-size}} specifies rules for managing the size of packets.
* Finally, encoding details of QUIC protocol elements are described in:
- {{versions}} (Versions),
- {{integer-encoding}} (Integer Encoding),
- {{packet-formats}} (Packet Headers),
- {{transport-parameter-encoding}} (Transport Parameters),
- {{frame-formats}} (Frames), and
- {{error-codes}} (Errors).
Accompanying documents describe QUIC's loss detection and congestion control
{{QUIC-RECOVERY}}, and the use of TLS for key negotiation {{QUIC-TLS}}.
This document defines QUIC version 1, which conforms to the protocol invariants
in {{QUIC-INVARIANTS}}.
## Terms and Definitions
The keywords "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD",
"SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in BCP 14 {{!RFC2119}} {{!RFC8174}}
when, and only when, they appear in all capitals, as shown here.
Commonly used terms in the document are described below.
QUIC:
: The transport protocol described by this document. QUIC is a name, not an
acronym.
QUIC packet:
: A complete processable unit of QUIC that can be encapsulated in a UDP
datagram. Multiple QUIC packets can be encapsulated in a single UDP datagram.
Endpoint:
: An entity that can participate in a QUIC connection by generating,
receiving, and processing QUIC packets. There are only two types of endpoint
in QUIC: client and server.
Client:
: The endpoint initiating a QUIC connection.
Server:
: The endpoint accepting incoming QUIC connections.
Connection ID:
: An opaque identifier that is used to identify a QUIC connection at an
endpoint. Each endpoint sets a value for its peer to include in packets sent
towards the endpoint.
Stream:
: A unidirectional or bidirectional channel of ordered bytes within a QUIC
connection. A QUIC connection can carry multiple simultaneous streams.
Application:
: An entity that uses QUIC to send and receive data.
## Notational Conventions
Packet and frame diagrams in this document use the format described in Section
3.1 of {{?RFC2360}}, with the following additional conventions:
\[x\]:
: Indicates that x is optional
x (A):
: Indicates that x is A bits long
x (A/B/C) ...:
: Indicates that x is one of A, B, or C bits long
x (i) ...:
: Indicates that x uses the variable-length encoding in {{integer-encoding}}
x (*) ...:
: Indicates that x is variable-length
# Streams {#streams}
Streams in QUIC provide a lightweight, ordered byte-stream abstraction to an
application. Streams can be unidirectional or bidirectional. An alternative
view of QUIC unidirectional streams is a "message" abstraction of practically
unlimited length.
Streams can be created by sending data. Other processes associated with stream
management - ending, cancelling, and managing flow control - are all designed to
impose minimal overheads. For instance, a single STREAM frame ({{frame-stream}})
can open, carry data for, and close a stream. Streams can also be long-lived and
can last the entire duration of a connection.
Streams can be created by either endpoint, can concurrently send data
interleaved with other streams, and can be cancelled. QUIC does not provide any
means of ensuring ordering between bytes on different streams.
QUIC allows for an arbitrary number of streams to operate concurrently and for
an arbitrary amount of data to be sent on any stream, subject to flow control
constraints (see {{flow-control}}) and stream limits.
## Stream Types and Identifiers {#stream-id}
Streams can be unidirectional or bidirectional. Unidirectional streams carry
data in one direction: from the initiator of the stream to its peer.
Bidirectional streams allow for data to be sent in both directions.
Streams are identified within a connection by a numeric value, referred to as
the stream ID. A stream ID is a 62-bit integer (0 to 2^62-1) that is unique for
all streams on a connection. Stream IDs are encoded as variable-length integers
(see {{integer-encoding}}). A QUIC endpoint MUST NOT reuse a stream ID within a
connection.
The least significant bit (0x1) of the stream ID identifies the initiator of the
stream. Client-initiated streams have even-numbered stream IDs (with the bit
set to 0), and server-initiated streams have odd-numbered stream IDs (with the
bit set to 1).
The second least significant bit (0x2) of the stream ID distinguishes between
bidirectional streams (with the bit set to 0) and unidirectional streams (with
the bit set to 1).
The least significant two bits from a stream ID therefore identify a stream as
one of four types, as summarized in {{stream-id-types}}.
| Bits | Stream Type |
|:-----|:---------------------------------|
| 0x0 | Client-Initiated, Bidirectional |
| 0x1 | Server-Initiated, Bidirectional |
| 0x2 | Client-Initiated, Unidirectional |
| 0x3 | Server-Initiated, Unidirectional |
{: #stream-id-types title="Stream ID Types"}
Within each type, streams are created with numerically increasing stream IDs. A
stream ID that is used out of order results in all streams of that type with
lower-numbered stream IDs also being opened.
The first bidirectional stream opened by the client has a stream ID of 0.
## Sending and Receiving Data
STREAM frames ({{frame-stream}}) 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
{{flow-control}}.
## Stream Prioritization {#stream-prioritization}
Stream multiplexing can have a significant effect on application performance if
resources allocated to streams are correctly prioritized.
QUIC does not provide a mechanism for exchanging prioritization information.
Instead, it relies on receiving priority information from the application that
uses QUIC.
A QUIC implementation SHOULD provide ways in which an application can indicate
the relative priority of streams. When deciding which streams to dedicate
resources to, the implementation SHOULD use the information provided by the
application.
## Required Operations on Streams
There are certain operations which an application MUST be able to perform when
interacting with QUIC streams. This document does not specify an API, but
any implementation of this version of QUIC MUST expose the ability to perform
the operations described in this section on a QUIC stream.
On the sending part of a stream, application protocols need to be able to:
- write data, understanding when stream flow control credit
({{data-flow-control}}) has successfully been reserved to send the written
data
- end the stream (clean termination), resulting in a STREAM frame
({{frame-stream}}) with the FIN bit set; and
- reset the stream (abrupt termination), resulting in a RESET_STREAM frame
({{frame-reset-stream}}), even if the stream was already ended.
On the receiving part of a stream, application protocols need to be able to:
- read data
- abort reading of the stream and request closure, possibly resulting in a
STOP_SENDING frame ({{frame-stop-sending}})
Applications also need to be informed of state changes on streams, including
when the peer has opened or reset a stream, when a peer aborts reading on a
stream, when new data is available, and when data can or cannot be written to
the stream due to flow control.
# Stream States {#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 ({{stream-send-states}}), and another for streams on which an
endpoint receives data ({{stream-recv-states}}).
Unidirectional streams use the applicable state machine directly. Bidirectional
streams use both state machines. For the most part, the use of these state
machines is the same whether the stream is unidirectional or bidirectional. The
conditions for opening a stream are slightly more complex for a bidirectional
stream because the opening of either send or receive sides causes the stream
to open in both directions.
An endpoint MUST open streams of the same type in increasing order of stream ID.
Note:
: These states are largely informative. This document uses stream states to
describe rules for when and how different types of frames can be sent and the
reactions that are expected when different types of frames are received.
Though these state machines are intended to be useful in implementing QUIC,
these states aren't intended to constrain implementations. An implementation
can define a different state machine as long as its behavior is consistent
with an implementation that implements these states.
## Sending Stream States {#stream-send-states}
{{fig-stream-send-states}} 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 |
| |
| Peer Creates |
| Bidirectional Stream |
v |
+-------+ |
| Send | Send RESET_STREAM |
| |---------------------->|
+-------+ |
| |
| Send STREAM + FIN |
v v
+-------+ +-------+
| Data | Send RESET_STREAM | Reset |
| Sent |------------------>| Sent |
+-------+ +-------+
| |
| Recv All ACKs | Recv ACK
v v
+-------+ +-------+
| Data | | Reset |
| Recvd | | Recvd |
+-------+ +-------+
~~~
{: #fig-stream-send-states title="States for Sending Parts of Streams"}
The sending part of stream that the endpoint initiates (types 0
and 2 for clients, 1 and 3 for servers) is opened by the application. The
"Ready" state represents a newly created stream that is able to accept data from
the application. Stream data might be buffered in this state in preparation for
sending.
Sending the first STREAM or STREAM_DATA_BLOCKED frame causes a sending part of a
stream to enter the "Send" state. An implementation might choose to defer
allocating a stream ID to a stream until it sends the first STREAM frame and
enters this state, which can allow for better stream prioritization.
The sending part of a bidirectional stream initiated by a peer (type 0 for a
server, type 1 for a client) enters the "Ready" state then immediately
transitions to the "Send" state if the receiving part enters the "Recv" state
({{stream-recv-states}}).
In the "Send" state, an endpoint transmits - and retransmits as necessary -
stream data in STREAM frames. The endpoint respects the flow control limits set
by its peer, and continues to accept and process MAX_STREAM_DATA frames. An
endpoint in the "Send" state generates STREAM_DATA_BLOCKED frames if it is
blocked from sending by stream or connection flow control limits
{{data-flow-control}}.
After the application indicates that all stream data has been sent and a STREAM
frame containing the FIN bit is sent, the sending part of the stream enters the
"Data Sent" state. From this state, the endpoint only retransmits stream data
as necessary. The endpoint does not need to check flow control limits or send
STREAM_DATA_BLOCKED frames for a stream in this state. MAX_STREAM_DATA frames
might be received until the peer receives the final stream offset. The endpoint
can safely ignore any MAX_STREAM_DATA frames it receives from its peer for a
stream in this state.
Once all stream data has been successfully acknowledged, the sending part of the
stream enters the "Data Recvd" state, which is a terminal state.
From any of the "Ready", "Send", or "Data Sent" states, an application can
signal that it wishes to abandon transmission of stream data. Alternatively, an
endpoint might receive a STOP_SENDING frame from its peer. In either case, the
endpoint sends a RESET_STREAM frame, which causes the stream to enter the "Reset
Sent" state.
An endpoint MAY send a RESET_STREAM as the first frame that mentions a stream;
this causes the sending part of that stream to open and then immediately
transition to the "Reset Sent" state.
Once a packet containing a RESET_STREAM has been acknowledged, the sending part
of the stream enters the "Reset Recvd" state, which is a terminal state.
## Receiving Stream States {#stream-recv-states}
{{fig-stream-recv-states}} shows the states for the part of a stream that
receives data from a peer. The states for a receiving part of a stream mirror
only some of the states of the sending part of the stream at the peer. The
receiving part of a stream does not track states on the sending part that cannot
be observed, such as the "Ready" state. Instead, the receiving part of a stream
tracks the delivery of data to the application, some of which cannot be observed
by the sender.
~~~
o
| Recv STREAM / STREAM_DATA_BLOCKED / RESET_STREAM
| Create Bidirectional Stream (Sending)
| Recv MAX_STREAM_DATA / STOP_SENDING (Bidirectional)
| Create Higher-Numbered Stream
v
+-------+
| Recv | Recv RESET_STREAM
| |-----------------------.
+-------+ |
| |
| Recv STREAM + FIN |
v |
+-------+ |
| Size | Recv RESET_STREAM |
| Known |---------------------->|
+-------+ |
| |
| Recv All Data |
v v
+-------+ Recv RESET_STREAM +-------+
| Data |--- (optional) --->| Reset |
| Recvd | Recv All Data | Recvd |
+-------+<-- (optional) ----+-------+
| |
| App Read All Data | App Read RST
v v
+-------+ +-------+
| Data | | Reset |
| Read | | Read |
+-------+ +-------+
~~~
{: #fig-stream-recv-states title="States for Receiving Parts of Streams"}
The receiving part of a stream initiated by a peer (types 1 and 3 for a client,
or 0 and 2 for a server) is created when the first STREAM, STREAM_DATA_BLOCKED,
or RESET_STREAM is received for that stream. For bidirectional streams
initiated by a peer, receipt of a MAX_STREAM_DATA or STOP_SENDING frame for the
sending part of the stream also creates the receiving part. The initial state
for the receiving part of stream is "Recv".
The receiving part of a stream enters the "Recv" state when the sending part of
a bidirectional stream initiated by the endpoint (type 0 for a client, type 1
for a server) enters the "Ready" state.
An endpoint opens a bidirectional stream when a MAX_STREAM_DATA or STOP_SENDING
frame is received from the peer for that stream. Receiving a MAX_STREAM_DATA
frame for an unopened stream indicates that the remote peer has opened the
stream and is providing flow control credit. Receiving a STOP_SENDING frame for
an unopened stream indicates that the remote peer no longer wishes to receive
data on this stream. Either frame might arrive before a STREAM or
STREAM_DATA_BLOCKED frame if packets are lost or reordered.
Before a stream is created, all streams of the same type with lower-numbered
stream IDs MUST be created. This ensures that the creation order for streams is
consistent on both endpoints.
In the "Recv" state, the endpoint receives STREAM and STREAM_DATA_BLOCKED
frames. Incoming data is buffered and can be reassembled into the correct order
for delivery to the application. As data is consumed by the application and
buffer space becomes available, the endpoint sends MAX_STREAM_DATA frames to
allow the peer to send more data.
When a STREAM frame with a FIN bit is received, the final size of the stream is
known (see {{final-size}}). The receiving part of the stream then enters the
"Size Known" state. In this state, the endpoint no longer needs to send
MAX_STREAM_DATA frames, it only receives any retransmissions of stream data.
Once all data for the stream has been received, the receiving part enters the
"Data Recvd" state. This might happen as a result of receiving the same STREAM
frame that causes the transition to "Size Known". After all data has been
received, any STREAM or STREAM_DATA_BLOCKED frames for the stream can be
discarded.
The "Data Recvd" state persists until stream data has been delivered to the
application. Once stream data has been delivered, the stream enters the "Data
Read" state, which is a terminal state.
Receiving a RESET_STREAM frame in the "Recv" or "Size Known" states causes the
stream to enter the "Reset Recvd" state. This might cause the delivery of
stream data to the application to be interrupted.
It is possible that all stream data is received when a RESET_STREAM is received
(that is, from the "Data Recvd" state). Similarly, it is possible for remaining
stream data to arrive after receiving a RESET_STREAM frame (the "Reset Recvd"
state). An implementation is free to manage this situation as it chooses.
Sending RESET_STREAM means that an endpoint cannot guarantee delivery of stream
data; however there is no requirement that stream data not be delivered if a
RESET_STREAM is received. An implementation MAY interrupt delivery of stream
data, discard any data that was not consumed, and signal the receipt of the
RESET_STREAM. A RESET_STREAM signal might be suppressed or withheld if stream
data is completely received and is buffered to be read by the application. If
the RESET_STREAM is suppressed, the receiving part of the stream remains in
"Data Recvd".
Once the application receives the signal indicating that the stream
was reset, the receiving part of the stream transitions to the "Reset Read"
state, which is a terminal state.
## Permitted Frame Types
The sender of a stream sends just three frame types that affect the state of a
stream at either sender or receiver: STREAM ({{frame-stream}}),
STREAM_DATA_BLOCKED ({{frame-stream-data-blocked}}), and RESET_STREAM
({{frame-reset-stream}}).
A sender MUST NOT send any of these frames from a terminal state ("Data Recvd"
or "Reset Recvd"). A sender MUST NOT send STREAM or STREAM_DATA_BLOCKED after
sending a RESET_STREAM; that is, in the terminal states and in the "Reset Sent"
state. A receiver could receive any of these three frames in any state, due to
the possibility of delayed delivery of packets carrying them.
The receiver of a stream sends MAX_STREAM_DATA ({{frame-max-stream-data}}) and
STOP_SENDING frames ({{frame-stop-sending}}).
The receiver only sends MAX_STREAM_DATA in the "Recv" state. A receiver can
send STOP_SENDING in any state where it has not received a RESET_STREAM frame;
that is states other than "Reset Recvd" or "Reset Read". However there is
little value in sending a STOP_SENDING frame in the "Data Recvd" state, since
all stream data has been received. A sender could receive either of these two
frames in any state as a result of delayed delivery of packets.
## Bidirectional Stream States {#stream-bidi-states}
A bidirectional stream is composed of sending and receiving parts.
Implementations may represent states of the bidirectional stream as composites
of sending and receiving stream states. The simplest model presents the stream
as "open" when either sending or receiving parts are in a non-terminal state and
"closed" when both sending and receiving streams are in terminal states.
{{stream-bidi-mapping}} shows a more complex mapping of bidirectional stream
states that loosely correspond to the stream states in HTTP/2
{{?HTTP2=RFC7540}}. This shows that multiple states on sending or receiving
parts of streams are mapped to the same composite state. Note that this is just
one possibility for such a mapping; this mapping requires that data is
acknowledged before the transition to a "closed" or "half-closed" state.
| 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 |
{: #stream-bidi-mapping title="Possible Mapping of Stream States to HTTP/2"}
Note (*1):
: A stream is considered "idle" if it has not yet been created, or if the
receiving part of the stream is in the "Recv" state without yet having
received any frames.
## 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" states, the transport SHOULD
signal this by sending a STOP_SENDING frame to prompt closure of the stream in
the opposite direction. This typically indicates that the receiving application
is no longer reading data it receives from the stream, but it is not a guarantee
that incoming data will be ignored.
STREAM frames received after sending STOP_SENDING are still counted toward
connection and stream flow control, even though these frames can be discarded
upon receipt.
A STOP_SENDING frame requests that the receiving endpoint send a RESET_STREAM
frame. An endpoint that receives a STOP_SENDING frame MUST send a RESET_STREAM
frame if the stream is in the Ready or Send state. If the stream is in the Data
Sent state and any outstanding data is declared lost, an endpoint SHOULD send a
RESET_STREAM frame in lieu of a retransmission.
An endpoint SHOULD copy the error code from the STOP_SENDING frame to the
RESET_STREAM frame it sends, but MAY use any application error code. The
endpoint that sends a STOP_SENDING frame MAY ignore the error code carried in
any RESET_STREAM frame it receives.
If the STOP_SENDING frame is received on a stream that is already in the
"Data Sent" state, an endpoint that wishes to cease retransmission of
previously-sent STREAM frames on that stream MUST first send a RESET_STREAM
frame.
STOP_SENDING SHOULD only be sent for a stream that has not been reset by the
peer. STOP_SENDING is most useful for streams in the "Recv" or "Size Known"
states.
An endpoint is expected to send another STOP_SENDING frame if a packet
containing a previous STOP_SENDING is lost. However, once either all stream
data or a RESET_STREAM frame has been received for the stream - that is, the
stream is in any state other than "Recv" or "Size Known" - sending a
STOP_SENDING frame is unnecessary.
An endpoint that wishes to terminate both directions of a bidirectional stream
can terminate one direction by sending a RESET_STREAM, and it can encourage
prompt termination in the opposite direction by sending a STOP_SENDING frame.
# Flow Control {#flow-control}
It is necessary to limit the amount of data that a receiver could buffer, to
prevent a fast sender from overwhelming a slow receiver, or to prevent a
malicious sender from consuming a large amount of memory at a receiver. To
enable a receiver to limit memory commitment to a connection and to apply back
pressure on the sender, streams are flow controlled both individually and as an
aggregate. A QUIC receiver controls the maximum amount of data the sender can
send on a stream at any time, as described in {{data-flow-control}} and
{{fc-credit}}
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 {{controlling-concurrency}}.
Data sent in CRYPTO frames is not flow controlled in the same way as stream
data. QUIC relies on the cryptographic protocol implementation to avoid
excessive buffering of data; see {{QUIC-TLS}}. The implementation SHOULD
provide an interface to QUIC to tell it about its buffering limits so that there
is not excessive buffering at multiple layers.
## Data Flow Control {#data-flow-control}
QUIC employs a credit-based flow-control scheme similar to that in HTTP/2
{{?HTTP2}}, where a receiver advertises the number of bytes it is prepared to
receive on a given stream and for the entire connection. This leads to two
levels of data flow control in QUIC:
* Stream flow control, which prevents a single stream from consuming the entire
receive buffer for a connection by limiting the amount of data that can be
sent on any stream.
* Connection flow control, which prevents senders from exceeding a receiver's
buffer capacity for the connection, by limiting the total bytes of stream data
sent in STREAM frames on all streams.
A receiver sets initial credits for all streams by sending transport parameters
during the handshake ({{transport-parameters}}). A receiver sends
MAX_STREAM_DATA ({{frame-max-stream-data}}) or MAX_DATA ({{frame-max-data}})
frames to the sender to advertise additional credit.
A receiver advertises credit for a stream by sending a MAX_STREAM_DATA frame
with the Stream ID field set appropriately. A MAX_STREAM_DATA frame indicates
the maximum absolute byte offset of a stream. A receiver could use the current
offset of data consumed to determine the flow control offset to be advertised.
A receiver MAY send MAX_STREAM_DATA frames in multiple packets in order to make
sure that the sender receives an update before running out of flow control
credit, even if one of the packets is lost.
A receiver advertises credit for a connection by sending a MAX_DATA frame, which
indicates the maximum of the sum of the absolute byte offsets of all streams. A
receiver maintains a cumulative sum of bytes received on all streams, which is
used to check for flow control violations. A receiver might use a sum of bytes
consumed on all streams to determine the maximum data limit to be advertised.
A receiver can advertise a larger offset by sending MAX_STREAM_DATA or MAX_DATA
frames. Once a receiver advertises an offset, it MAY advertise a smaller
offset, but this has no effect.
A receiver MUST close the connection with a FLOW_CONTROL_ERROR error
({{error-handling}}) if the sender violates the advertised connection or stream
data limits.
A sender MUST ignore any MAX_STREAM_DATA or MAX_DATA frames that do not increase
flow control limits.
If a sender runs out of flow control credit, it will be unable to send new data
and is considered blocked. A sender SHOULD send a STREAM_DATA_BLOCKED or
DATA_BLOCKED frame to indicate it has data to write but is blocked by flow
control limits. These frames are expected to be sent infrequently in common
cases, but they are considered useful for debugging and monitoring purposes.
A sender SHOULD NOT send multiple STREAM_DATA_BLOCKED or DATA_BLOCKED frames
for the same data limit, unless the original frame is determined to be lost.
Another STREAM_DATA_BLOCKED or DATA_BLOCKED frame can be sent after the data
limit is increased.
## Flow Credit Increments {#fc-credit}
This document leaves when and how many bytes to advertise in a MAX_STREAM_DATA
or MAX_DATA frame to implementations, but offers a few considerations. These
frames contribute to connection overhead. Therefore frequently sending frames
with small changes is undesirable. At the same time, larger increments to
limits are necessary to avoid blocking if updates are less frequent, requiring
larger resource commitments at the receiver. Thus 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, sending frames related to flow control
only when there are other frames to send or when a peer is blocked ensures that
flow control doesn't cause extra packets to be sent.
If a sender runs out of flow control credit, it will be unable to send new data
and is considered blocked. It is generally considered best to not let the
sender become blocked. To avoid blocking a sender, and to reasonably account
for the possibility of loss, a receiver should send a MAX_DATA or
MAX_STREAM_DATA frame at least two round trips before it expects the sender to
get blocked.
A receiver MUST NOT wait for a STREAM_DATA_BLOCKED or DATA_BLOCKED frame before
sending MAX_STREAM_DATA or MAX_DATA, since doing so will mean that a sender will
be blocked for at least an entire round trip, and potentially for longer if the
peer chooses to not send STREAM_DATA_BLOCKED or DATA_BLOCKED frames.
## Handling Stream Cancellation {#stream-cancellation}
Endpoints need to eventually agree on the amount of flow control credit that has
been consumed, to avoid either exceeding flow control limits or deadlocking.
On receipt of a RESET_STREAM frame, an endpoint will tear down state for the
matching stream and ignore further data arriving on that stream. Without the
offset included in RESET_STREAM, the two endpoints could disagree on
the number of bytes that count towards connection flow control.
To remedy this issue, a RESET_STREAM frame ({{frame-reset-stream}}) includes the
final size of data sent on the stream. On receiving a RESET_STREAM frame, a
receiver definitively knows how many bytes were sent on that stream before the
RESET_STREAM frame, and the receiver MUST use the final size of the stream to
account for all bytes sent on the stream in its connection level flow
controller.
RESET_STREAM terminates one direction of a stream abruptly. For a bidirectional
stream, RESET_STREAM has no effect on data flow in the opposite direction. Both
endpoints MUST maintain flow control state for the stream in the unterminated
direction until that direction enters a terminal state, or until one of the
endpoints sends CONNECTION_CLOSE.
## Stream Final Size {#final-size}
The final size is the amount of flow control credit that is consumed by a
stream. Assuming that every contiguous byte on the stream was sent once, the
final size is the number of bytes sent. More generally, this is one higher
than the offset of the byte with the largest offset sent on the stream, or zero
if no bytes were sent.
For a stream that is reset, the final size is carried explicitly in a
RESET_STREAM frame. Otherwise, the final size is the offset plus the length of
a STREAM frame marked with a FIN flag, or 0 in the case of incoming
unidirectional streams.
An endpoint will know the final size for a stream when the receiving part of the
stream enters the "Size Known" or "Reset Recvd" state ({{stream-states}}).
An endpoint MUST NOT send data on a stream at or beyond the final size.
Once a final size for a stream is known, it cannot change. If a RESET_STREAM or
STREAM frame is received indicating a change in the final size for the stream,
an endpoint SHOULD respond with a FINAL_SIZE_ERROR error (see
{{error-handling}}). A receiver SHOULD treat receipt of data at or beyond the
final size as a FINAL_SIZE_ERROR error, even after a stream is closed.
Generating these errors is not mandatory, but only because requiring that an
endpoint generate these errors also means that the endpoint needs to maintain
the final size state for closed streams, which could mean a significant state
commitment.
## Controlling Concurrency {#controlling-concurrency}
An endpoint limits the cumulative number of incoming streams a peer can open.
Only streams with a stream ID less than (max_stream * 4 +
initial_stream_id_for_type) can be opened (see {{long-packet-types}}). Initial
limits are set in the transport parameters (see
{{transport-parameter-definitions}}) and subsequently limits are advertised
using MAX_STREAMS frames ({{frame-max-streams}}). Separate limits apply to
unidirectional and bidirectional streams.
If a max_streams transport parameter or MAX_STREAMS frame is received with a
value greater than 2^60, this would allow a maximum stream ID that cannot be
expressed as a variable-length integer (see {{integer-encoding}}).
If either is received, the connection MUST be closed immediately with a
connection error of type STREAM_LIMIT_ERROR (see {{immediate-close}}).
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 ({{error-handling}}).
Once a receiver advertises a stream limit using the MAX_STREAMS frame,
advertising a smaller limit has no effect. A receiver MUST ignore any
MAX_STREAMS frame that does not increase the stream limit.
As with stream and connection flow control, this document leaves when and how
many streams to advertise to a peer via MAX_STREAMS to implementations.
Implementations might choose to increase limits as streams close to keep the
number of streams available to peers roughly consistent.
An endpoint that is unable to open a new stream due to the peer's limits SHOULD
send a STREAMS_BLOCKED frame ({{frame-streams-blocked}}). This signal is
considered useful for debugging. An endpoint MUST NOT wait to receive this
signal before advertising additional credit, since doing so will mean that the
peer will be blocked for at least an entire round trip, and potentially for
longer if the peer chooses to not send STREAMS_BLOCKED frames.
# Connections {#connections}
QUIC's connection establishment combines version negotiation with the
cryptographic and transport handshakes to reduce connection establishment
latency, as described in {{handshake}}. Once established, a connection
may migrate to a different IP or port at either endpoint as
described in {{migration}}. Finally, a connection may be terminated by either
endpoint, as described in {{termination}}.
## Connection ID {#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) don't cause packets for a QUIC
connection to be delivered to the wrong endpoint. Each endpoint selects
connection IDs using an implementation-specific (and perhaps
deployment-specific) method which will allow packets with that connection ID to
be routed back to the endpoint and identified by the endpoint upon receipt.
Connection IDs MUST NOT contain any information that can be used by an external
observer (that is, one that does not cooperate with the issuer) to correlate
them with other connection IDs for the same connection. As a trivial example,
this means the same connection ID MUST NOT be issued more than once on the same
connection.
Packets with long headers include Source Connection ID and Destination
Connection ID fields. These fields are used to set the connection IDs for new
connections; see {{negotiating-connection-ids}} for details.
Packets with short headers ({{short-header}}) 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 ({{packet-version}}) packet echoes the connection IDs
selected by the client, both to ensure correct routing toward the client and to
allow the client to validate that the packet is in response to an Initial
packet.
A zero-length connection ID MAY be used when the connection ID is not needed for
routing and the address/port tuple of packets is sufficient to identify a
connection. An endpoint whose peer has selected a zero-length connection ID MUST
continue to use a zero-length connection ID for the lifetime of the connection
and MUST NOT send packets from any other local address.
When an endpoint has requested 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 ({{frame-new-connection-id}}).
### Issuing Connection IDs {#issue-cid}
Each Connection ID has an associated sequence number to assist in deduplicating
messages. The initial connection ID issued by an endpoint is sent in the Source
Connection ID field of the long packet header ({{long-header}}) 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 ({{frame-new-connection-id}}). The sequence number on each newly-issued
connection ID MUST increase by 1. The connection ID randomly selected by the
client in the Initial packet and any connection ID provided by a Retry packet
are not assigned sequence numbers unless a server opts to retain them as its
initial connection ID.
When an endpoint issues a connection ID, it MUST accept packets that carry this
connection ID for the duration of the connection or until its peer invalidates
the connection ID via a RETIRE_CONNECTION_ID frame
({{frame-retire-connection-id}}).
An endpoint SHOULD ensure that its peer has a sufficient number of available and
unused connection IDs. Endpoints store received connection IDs for future use
and advertise the number of connection IDs they are willing to store with the
active_connection_id_limit transport parameter. An endpoint SHOULD NOT provide
more connection IDs than the peer's limit.
An endpoint SHOULD supply a new connection ID when it receives a packet with a
previously unused connection ID or when the peer retires one, unless providing
the new connection ID would exceed the peer's limit. An endpoint MAY limit the
frequency or the total number of connection IDs issued for each connection to
avoid the risk of running out of connection IDs; see {{reset-token}}.
An endpoint that initiates migration and requires non-zero-length connection IDs
SHOULD ensure that the pool of connection IDs available to its peer allows the
peer to use a new connection ID on migration, as the peer will close the
connection if the pool is exhausted.
### Consuming and Retiring Connection IDs {#retiring-cids}
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 {{migration-linkability}} for more.
An endpoint maintains a set of connection IDs received from its peer, any of
which it can use when sending packets. When the endpoint wishes to remove a
connection ID from use, it sends a RETIRE_CONNECTION_ID frame to its peer.
Sending a RETIRE_CONNECTION_ID frame indicates that the connection ID will not
be used again and requests that the peer replace it with a new connection ID
using a NEW_CONNECTION_ID frame.
As discussed in {{migration-linkability}}, each connection ID MUST be used on
packets sent from only one local address. An endpoint that migrates away from a
local address SHOULD retire all connection IDs used on that address once it no
longer plans to use that address.
An endpoint can request that its peer retire connection IDs by sending a
NEW_CONNECTION_ID frame with an increased Retire Prior To field. Upon receipt,
the peer SHOULD retire the corresponding connection IDs and send the
corresponding RETIRE_CONNECTION_ID frames in a timely manner. Failing to do so
can cause packets to be delayed, lost, or cause the original endpoint to send a
stateless reset in response to a connection ID it can no longer route correctly.
An endpoint MAY discard a connection ID for which retirement has been requested
once an interval of no less than 3 PTO has elapsed since an acknowledgement is
received for the NEW_CONNECTION_ID frame requesting that retirement. Subsequent
incoming packets using that connection ID could elicit a response with the
corresponding stateless reset token.
## Matching Packets to Connections {#packet-handling}
Incoming packets are classified on receipt. Packets can either be associated
with an existing connection, or - for servers - potentially create a new
connection.
Hosts try to associate a packet with an existing connection. If the packet has a
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 {{connection-id}}.
If the Destination Connection ID is zero length and the packet matches the
address/port tuple of a connection where the host did not require connection
IDs, QUIC processes the packet as part of that connection. Endpoints SHOULD
either reject connection attempts that use the same addresses as existing
connections, or use a non-zero-length Destination Connection ID so that packets
can be correctly attributed to connections.
Endpoints can send a Stateless Reset ({{stateless-reset}}) 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 without packet protection, such as Initial, Retry, or Version
Negotiation, MAY be discarded. An endpoint MUST generate a connection error if
it commits changes to state before discovering an error.
### Client Packet Handling {#client-pkt-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 address/port tuple to identify a
connection. Packets that don't match an existing connection are discarded.
Due to packet reordering or loss, a client might receive packets for a
connection that are encrypted with a key it has not yet computed. The client MAY
drop these packets, or MAY buffer them in anticipation of later packets that
allow it to compute the key.
If a client receives a packet that has an unsupported version, it MUST discard
that packet.
### Server Packet Handling {#server-pkt-handling}
If a server receives a packet that has an unsupported version, but the packet is
sufficiently large to initiate a new connection for any version supported by the
server, it SHOULD send a Version Negotiation packet as described in
{{send-vn}}. Servers MAY rate control these packets to avoid storms of Version
Negotiation packets. Otherwise, servers MUST drop packets that specify
unsupported versions.
The first packet for an unsupported version can use different semantics and
encodings for any version-specific field. In particular, different packet
protection keys might be used for different versions. Servers that do not
support a particular version are unlikely to be able to decrypt the payload of
the packet. Servers SHOULD NOT attempt to decode or decrypt a packet from an
unknown version, but instead send a Version Negotiation packet, provided that
the packet is sufficiently long.
Packets with a supported version, or no version field, are matched to a
connection using the connection ID or - for packets with zero-length connection
IDs - the address tuple. If the packet doesn't match an existing connection,
the server continues below.
If the packet is an Initial packet fully conforming with the specification, the
server proceeds with the handshake ({{handshake}}). This commits the server to
the version that the client selected.
If a server isn't currently accepting any new connections, it SHOULD send an
Initial packet containing a CONNECTION_CLOSE frame with error code
SERVER_BUSY.
If the packet is a 0-RTT packet, the server MAY buffer a limited number of these
packets in anticipation of a late-arriving Initial packet. Clients are not able
to send Handshake packets prior to receiving a server response, so servers
SHOULD ignore any such packets.
Servers MUST drop incoming packets under all other circumstances.
## Life of a QUIC Connection {#connection-lifecycle}
TBD.
<!-- Goes into how the next few sections are connected. Specifically, one goal
is to combine the address validation section that shows up below with path
validation that shows up later, and explain why these two mechanisms are
required here.
suggested structure:
- establishment
- VN
- Retry
- Crypto
- use (include migration)
- shutdown
-->
## Required Operations on Connections
There are certain operations which an application MUST be able to perform when
interacting with the QUIC transport. This document does not specify an API, but
any implementation of this version of QUIC MUST expose the ability to perform
the operations described in this section on a QUIC connection.
When implementing the client role, applications need to be able to:
- open a connection, which begins the exchange described in {{handshake}};
- enable 0-RTT; and
- be informed when 0-RTT has been accepted or rejected by a server.
When implementing the server role, applications need to be able to:
- listen for incoming connections, which prepares for the exchange described in
{{handshake}};
- if Early Data is supported, embed application-controlled data in the TLS
resumption ticket sent to the client; and
- if Early Data is supported, retrieve application-controlled data from the
client's resumption ticket and enable rejecting Early Data based on that
information.
In either role, applications need to be able to:
- configure minimum values for the initial number of permitted streams of each
type, as communicated in the transport parameters ({{transport-parameters}});
- control resource allocation of various types, including flow control and the
number of permitted streams of each type;
- identify whether the handshake has completed successfully or is still ongoing
- keep a connection from silently closing, either by generating PING frames
({{frame-ping}}) or by requesting that the transport send additional frames
before the idle timeout expires ({{idle-timeout}}); and
- immediately close ({{immediate-close}}) the connection.
# Version Negotiation {#version-negotiation}
Version negotiation ensures that client and server agree to a QUIC version
that is mutually supported. A server sends a Version Negotiation packet in
response to each packet that might initiate a new connection; see
{{packet-handling}} for details.
The size of the first packet sent by a client will determine whether a server
sends a Version Negotiation packet. Clients that support multiple QUIC versions
SHOULD pad the first packet they send to the largest of the minimum packet sizes
across all versions they support. This ensures that the server responds if there
is a mutually supported version.
## Sending Version Negotiation Packets {#send-vn}
If the version selected by the client is not acceptable to the server, the
server responds with a Version Negotiation packet (see {{packet-version}}).
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.
As a result, the client discards all state for the connection and does not send
any more packets on the connection.
A server MAY limit the number of Version Negotiation packets it sends. For
instance, a server that is able to recognize packets as 0-RTT might choose not
to send Version Negotiation packets in response to 0-RTT packets with the
expectation that it will eventually receive an Initial packet.
## Handling Version Negotiation Packets {#handle-vn}
When a client receives a Version Negotiation packet, it MUST abandon the
current connection attempt. Version Negotiation packets are designed to allow
future versions of QUIC to negotiate the version in use between endpoints.
Future versions of QUIC might change how implementations that support multiple
versions of QUIC react to Version Negotiation packets when attempting to
establish a connection using this version. How to perform version negotiation
is left as future work defined by future versions of QUIC. In particular,
that future work will need to ensure robustness against version downgrade
attacks {{version-downgrade}}.
### Version Negotiation Between Draft Versions
\[\[RFC editor: please remove this section before publication.]]
When a draft implementation receives a Version Negotiation packet, it MAY use
it to attempt a new connection with one of the versions listed in the packet,
instead of abandoning the current connection attempt {{handle-vn}}.
The client MUST check that the Destination and Source Connection ID fields
match the Source and Destination Connection ID fields in a packet that the
client sent. If this check fails, the packet MUST be discarded.
Once the Version Negotiation packet is determined to be valid, the client then
selects an acceptable protocol version from the list provided by the server.
The client then attempts to create a new connection using that version. The new
connection MUST use a new random Destination Connection ID different from the
one it had previously sent.
Note that this mechanism does not protect against downgrade attacks and
MUST NOT be used outside of draft implementations.
## Using Reserved Versions
For a server to use a new version in the future, clients need to correctly
handle unsupported versions. To help ensure this, a server SHOULD include a
version that is reserved for forcing version negotiation (0x?a?a?a?a as defined
in {{versions}}) when generating a Version Negotiation packet.
The design of version negotiation permits a server to avoid maintaining state
for packets that it rejects in this fashion.
A client MAY send a packet using a version that is reserved for forcing version
negotiation. This can be used to solicit a list of supported versions from a
server.
# Cryptographic and Transport Handshake {#handshake}
QUIC relies on a combined cryptographic and transport handshake to minimize
connection establishment latency. QUIC uses the CRYPTO frame {{frame-crypto}}
to transmit the cryptographic handshake. Version 0x00000001 of QUIC uses TLS as
described in {{QUIC-TLS}}; a different QUIC version number could indicate that a
different cryptographic handshake protocol is in use.
QUIC provides reliable, ordered delivery of the cryptographic handshake
data. QUIC packet protection is used to encrypt as much of the handshake
protocol as possible. The cryptographic handshake MUST provide the following
properties:
* authenticated key exchange, where
* a server is always authenticated,
* a client is optionally authenticated,
* every connection produces distinct and unrelated keys,
* keying material is usable for packet protection for both 0-RTT and 1-RTT
packets, and
* 1-RTT keys have forward secrecy
* authenticated values for transport parameters of both endpoints, and
confidentiality protection for server transport parameters (see
{{transport-parameters}})
* authenticated negotiation of an application protocol (TLS uses ALPN
{{?RFC7301}} for this purpose)
The first CRYPTO frame from a client MUST be sent in a single packet. Any
second attempt that is triggered by address validation (see
{{validate-handshake}}) MUST also be sent within a single packet. This avoids
having to reassemble a message from multiple packets.
The first client packet of the cryptographic handshake protocol MUST fit within
a 1232 byte QUIC packet payload. This includes overheads that reduce the space
available to the cryptographic handshake protocol.
An endpoint can verify support for Explicit Congestion Notification (ECN) in the
first packets it sends, as described in {{ecn-validation}}.
The CRYPTO frame can be sent in different packet number spaces. The sequence
numbers used by CRYPTO frames to ensure ordered delivery of cryptographic
handshake data start from zero in each packet number space.
Endpoints MUST explicitly negotiate an application protocol. This avoids
situations where there is a disagreement about the protocol that is in use.
## 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 {{validate-retry}}.
Once any address validation exchanges are complete, the
cryptographic handshake is used to agree on cryptographic keys. The
cryptographic handshake is carried in Initial ({{packet-initial}}) and Handshake
({{packet-handshake}}) packets.
{{tls-1rtt-handshake}} provides an overview of the 1-RTT handshake. Each line
shows a QUIC packet with the packet type and packet number shown first, followed
by the frames that are typically contained in those packets. So, for instance
the first packet is of type Initial, with packet number 0, and contains a CRYPTO
frame carrying the ClientHello.
Note that multiple QUIC packets -- even of different encryption levels -- may be
coalesced into a single UDP datagram (see {{packet-coalesce}}), and so this
handshake may consist of as few as 4 UDP datagrams, or any number more. For
instance, the server's first flight contains packets from the Initial encryption
level (obfuscation), the Handshake level, and "0.5-RTT data" from the server at
the 1-RTT encryption level.
~~~~
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] ->
1-RTT[1]: STREAM[3, "..."], ACK[0]
<- Handshake[1]: ACK[0]
~~~~
{: #tls-1rtt-handshake title="Example 1-RTT Handshake"}
{{tls-0rtt-handshake}} shows an example of a connection with a 0-RTT handshake
and a single packet of 0-RTT data. Note that as described in {{packet-numbers}},
the server acknowledges 0-RTT data at the 1-RTT encryption level, 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] ->
1-RTT[1]: STREAM[3, "..."], ACK[1]
<- Handshake[1]: ACK[0]
~~~~
{: #tls-0rtt-handshake title="Example 0-RTT Handshake"}
## Negotiating Connection IDs {#negotiating-connection-ids}
A connection ID is used to ensure consistent routing of packets, as described in
{{connection-id}}. 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 ({{long-header}}) are used to
establish the connection ID that each endpoint uses. Each endpoint uses the
Source Connection ID field to specify the connection ID that is used in the
Destination Connection ID field of packets being sent to them. Upon receiving a
packet, each endpoint sets the Destination Connection ID it sends to match the
value of the Source Connection ID that they receive.
When an Initial packet is sent by a client that has not previously received an
Initial or Retry packet from the server, it populates the Destination Connection
ID field with an unpredictable value. This MUST be at least 8 bytes in length.
Until a packet is received from the server, the client MUST use the same value
unless it abandons the connection attempt and starts a new one. The initial
Destination Connection ID is used to determine packet protection keys for
Initial packets.
The client populates the Source Connection ID field with a value of its choosing
and sets the SCID Len field to indicate the length.
The first flight of 0-RTT packets use the same Destination and Source Connection
ID values as the client's first Initial.
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 subsequent 0-RTT packets. That means that
a client might change the Destination Connection ID twice during connection
establishment, once in response to a Retry and once in response to the first
Initial packet from the server. Once a client has received an Initial packet
from the server, it MUST discard any packet it receives with a different Source
Connection ID.
A client MUST only change the value it sends in the Destination Connection ID in
response to the first packet of each type it receives from the server (Retry or
Initial); a server MUST set its value based on the Initial packet. Any
additional changes are not permitted; if subsequent packets of those types
include a different Source Connection ID, they MUST be discarded. This avoids
problems that might arise from stateless processing of multiple Initial packets
producing different connection IDs.
The connection ID can change over the lifetime of a connection, especially in
response to connection migration ({{migration}}); see {{issue-cid}} for details.
## Transport Parameters {#transport-parameters}
During connection establishment, both endpoints make authenticated declarations
of their transport parameters. These declarations are made unilaterally by each
endpoint. Endpoints are required to comply with the restrictions implied by
these parameters; the description of each parameter includes rules for its
handling.
The encoding of the transport parameters is detailed in
{{transport-parameter-encoding}}.
QUIC includes the encoded transport parameters in the cryptographic handshake.
Once the handshake completes, the transport parameters declared by the peer are
available. Each endpoint validates the value provided by its peer.
Definitions for each of the defined transport parameters are included in
{{transport-parameter-definitions}}.
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.
A server MUST include the original_connection_id transport parameter
({{transport-parameter-definitions}}) if it sent a Retry packet to enable
validation of the Retry, as described in {{packet-retry}}.
### Values of Transport Parameters for 0-RTT {#zerortt-parameters}
Both endpoints store the value of the server transport parameters from
a connection and apply them to any 0-RTT packets that are sent in
subsequent connections to that peer, except for transport parameters that
are explicitly excluded. Remembered transport parameters apply to the new
connection until the handshake completes and the client starts sending
1-RTT packets. Once the handshake completes, the client uses the transport
parameters established in the handshake.
The definition of new transport parameters ({{new-transport-parameters}}) MUST
specify whether they MUST, MAY, or MUST NOT be stored for 0-RTT. A client need
not store a transport parameter it cannot process.
A client MUST NOT use remembered values for the following parameters:
original_connection_id, preferred_address, stateless_reset_token,
ack_delay_exponent and active_connection_id_limit. The client MUST use the
server's new values in the handshake instead, and absent new values from the
server, the default value.
A client that attempts to send 0-RTT data MUST remember all other transport
parameters used by the server. The server can remember these transport
parameters, or store an integrity-protected copy of the values in the ticket
and recover the information when accepting 0-RTT data. A server uses the
transport parameters in determining whether to accept 0-RTT data.
If 0-RTT data is accepted by the server, the server MUST NOT reduce any
limits or alter any values that might be violated by the client with its
0-RTT data. In particular, a server that accepts 0-RTT data MUST NOT set
values for the following parameters ({{transport-parameter-definitions}})
that are smaller than the remembered value of the parameters.
* initial_max_data
* initial_max_stream_data_bidi_local
* initial_max_stream_data_bidi_remote
* initial_max_stream_data_uni
* initial_max_streams_bidi
* initial_max_streams_uni
Omitting or setting a zero value for certain transport parameters can result in
0-RTT data being enabled, but not usable. The applicable subset of transport
parameters that permit sending of application data SHOULD be set to non-zero
values for 0-RTT. This includes initial_max_data and either
initial_max_streams_bidi and initial_max_stream_data_bidi_remote, or
initial_max_streams_uni and initial_max_stream_data_uni.
A server MUST either reject 0-RTT data or abort a handshake if the implied
values for transport parameters cannot be supported.
When sending frames in 0-RTT packets, a client MUST only use remembered
transport parameters; importantly, it MUST NOT use updated values that it learns
from the server's updated transport parameters or from frames received in 1-RTT
packets. Updated values of transport parameters from the handshake apply only
to 1-RTT packets. For instance, flow control limits from remembered transport
parameters apply to all 0-RTT packets even if those values are increased by the
handshake or by frames sent in 1-RTT packets. A server MAY treat use of updated
transport parameters in 0-RTT as a connection error of type PROTOCOL_VIOLATION.
### New Transport Parameters {#new-transport-parameters}
New transport parameters can be used to negotiate new protocol behavior. An
endpoint MUST ignore transport parameters that it does not support. Absence of
a transport parameter therefore disables any optional protocol feature that is
negotiated using the parameter. As described in {{transport-parameter-grease}},
some identifiers are reserved in order to exercise this requirement.
New transport parameters can be registered according to the rules in
{{iana-transport-parameters}}.
## Cryptographic Message Buffering
Implementations need to maintain a buffer of CRYPTO data received out of order.
Because there is no flow control of CRYPTO frames, an endpoint could
potentially force its peer to buffer an unbounded amount of data.
Implementations MUST support buffering at least 4096 bytes of data received in
CRYPTO frames out of order. Endpoints MAY choose to allow more data to be
buffered during the handshake. A larger limit during the handshake could allow
for larger keys or credentials to be exchanged. An endpoint's buffer size does
not need to remain constant during the life of the connection.
Being unable to buffer CRYPTO frames during the handshake can lead to a
connection failure. If an endpoint's buffer is exceeded during the handshake, it
can expand its buffer temporarily to complete the handshake. If an endpoint
does not expand its buffer, it MUST close the connection with a
CRYPTO_BUFFER_EXCEEDED error code.
Once the handshake completes, if an endpoint is unable to buffer all data in a
CRYPTO frame, it MAY discard that CRYPTO frame and all CRYPTO frames received in
the future, or it MAY close the connection with a CRYPTO_BUFFER_EXCEEDED error
code. Packets containing discarded CRYPTO frames MUST be acknowledged because
the packet has been received and processed by the transport even though the
CRYPTO frame was discarded.
# Address Validation
Address validation is used by QUIC to avoid being used for a traffic
amplification attack. In such an attack, a packet is sent to a server with
spoofed source address information that identifies a victim. If a server
generates more or larger packets in response to that packet, the attacker can
use the server to send more data toward the victim than it would be able to send
on its own.
The primary defense against amplification attack is verifying that an endpoint
is able to receive packets at the transport address that it claims. Address
validation is performed both during connection establishment (see
{{validate-handshake}}) and during connection migration (see
{{migrate-validate}}).
## Address Validation During Connection Establishment {#validate-handshake}
Connection establishment implicitly provides address validation for both
endpoints. In particular, receipt of a packet protected with Handshake keys
confirms that the client received the Initial packet from the server. Once the
server has successfully processed a Handshake packet from the client, it can
consider the client address to have been validated.
Prior to validating the client address, servers MUST NOT send more than three
times as many bytes as the number of bytes they have received. This limits the
magnitude of any amplification attack that can be mounted using spoofed source
addresses. In determining this limit, servers only count the size of
successfully processed packets.
Clients MUST ensure that UDP datagrams containing only Initial packets are sized
to at least 1200 bytes, adding padding to packets in the datagram as necessary.
Sending padded datagrams ensures that the server is not overly constrained by
the amplification restriction.
Packet loss, in particular loss of a Handshake packet from the server, can cause
a situation in which the server cannot send when the client has no data to send
and the anti-amplification limit is reached. In order to avoid this causing a
handshake deadlock, clients SHOULD send a packet upon a crypto retransmission
timeout, as described in {{QUIC-RECOVERY}}. If the client has no data to
retransmit and does not have Handshake keys, it SHOULD send an Initial packet in
a UDP datagram of at least 1200 bytes. If the client has Handshake keys, it
SHOULD 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
{{validate-retry}}) or in a previous connection using the NEW_TOKEN frame (see
{{validate-future}}).
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.
### Address Validation using Retry Packets {#validate-retry}
Upon receiving the client's Initial packet, the server can request address
validation by sending a Retry packet ({{packet-retry}}) containing a token. This
token MUST be repeated by the client in all Initial packets it sends for that
connection after it receives the Retry packet. In response to processing an
Initial containing a token, a server can either abort the connection or permit
it to proceed.
As long as it is not possible for an attacker to generate a valid token for
its own address (see {{token-integrity}}) 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. By giving the client a different connection ID to
use, a server can cause the connection to be routed to a server instance with
more resources available for new connections.
A flow showing the use of a Retry packet is shown in {{fig-retry}}.
~~~~
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, "..."]
~~~~
{: #fig-retry title="Example Handshake with Retry"}
### Address Validation for Future Connections {#validate-future}
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 {{frame-new-token}} to provide the client
with an address validation token that can be used to validate future
connections. The client includes this token in Initial packets to provide
address validation in a future connection. The client MUST include the token in
all Initial packets it sends, unless a Retry replaces the token with a newer
one. The client MUST NOT use the token provided in a Retry for future
connections. Servers MAY discard any Initial packet that does not carry the
expected token.
A token SHOULD be constructed in a way that allows the server to distinguish it
from tokens that are sent in Retry packets as they are carried in the same
field.
The token MUST NOT include information that would allow it to be linked by an
on-path observer to the connection on which it was issued. For example, it
cannot include the connection ID or addressing information unless the values are
encrypted.
Unlike the token that is created for a Retry packet, there might be some time
between when the token is created and when the token is subsequently used.
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.
It is unlikely that the client port number is the same on two different
connections; validating the port is therefore unlikely to be successful.
If the client has a token received in a NEW_TOKEN frame on a previous connection
to what it believes to be the same server, it SHOULD include that value 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 token allows a server to correlate activity between the connection where the
token was issued and any connection where it is used. Clients that want to
break continuity of identity with a server MAY discard tokens provided using the
NEW_TOKEN frame. 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 in different connections. Reusing a token
allows connections to be linked by entities on the network path; see
{{migration-linkability}}. A client MUST NOT reuse a token if it believes that
its point of network attachment has changed since the token was last used; that
is, if there is a change in its local IP address or network interface. A client
needs to start the connection process over if there is any change in its local
address prior to completing the handshake.
Clients might receive multiple tokens on a single connection. Aside from
preventing linkability, any token can be used in any connection attempt.
Servers can send additional tokens to either enable address validation for
multiple connection attempts or to replace older tokens that might become
invalid. For a client, this ambiguity means that sending the most recent unused
token is most likely to be effective. Though saving and using older tokens has
no negative consequences, clients can regard older tokens as being less likely
be useful to the server for address validation.
When a server receives an Initial packet with an address validation token, it
MUST attempt to validate the token, unless it has already completed address
validation. If the token is invalid then the server SHOULD proceed as if
the client did not have a validated address, including potentially sending
a Retry. If the validation succeeds, the server SHOULD then allow the
handshake to proceed.
Note:
: The rationale for treating the client as unvalidated rather than discarding
the packet is that the client might have received the token in a previous
connection using the NEW_TOKEN frame, and if the server has lost state, it
might be unable to validate the token at all, leading to connection failure if
the packet is discarded. A server SHOULD encode tokens provided with
NEW_TOKEN frames and Retry packets differently, and validate the latter more
strictly.
In a stateless design, a server can use encrypted and authenticated tokens to
pass information to clients that the server can later recover and use to
validate a client address. Tokens are not integrated into the cryptographic
handshake and so they are not authenticated. For instance, a client might be
able to reuse a token. To avoid attacks that exploit this property, a server
can limit its use of tokens to only the information needed to validate client
addresses.
Attackers could replay tokens to use servers as amplifiers in DDoS attacks. To
protect against such attacks, servers SHOULD ensure that tokens sent in Retry
packets are only accepted for a short time. Tokens that are provided in
NEW_TOKEN frames (see {{frame-new-token}}) need to be valid for longer, but
SHOULD NOT be accepted multiple times in a short period. Servers are encouraged
to allow tokens to be used only once, if possible.
### Address Validation Token Integrity {#token-integrity}
An address validation token MUST be difficult to guess. Including a large
enough random value in the token would be sufficient, but this depends on the
server remembering the value it sends to clients.
A token-based scheme allows the server to offload any state associated with
validation to the client. For this design to work, the token MUST be covered by
integrity protection against modification or falsification by clients. Without
integrity protection, malicious clients could generate or guess values for
tokens that would be accepted by the server. Only the server requires access to
the integrity protection key for tokens.
There is no need for a single well-defined format for the token because the
server that generates the token also consumes it. A token could include
information about the claimed client address (IP and port), a timestamp, and any
other supplementary information the server will need to validate the token in
the future.
## Path Validation {#migrate-validate}
Path validation is used during connection migration (see {{migration}} and
{{preferred-address}}) by the migrating endpoint to verify reachability of a
peer from a new local address. In path validation, endpoints test reachability
between a specific local address and a specific peer address, where an address
is the two-tuple of IP address and port.
Path validation tests that packets (PATH_CHALLENGE) can be both sent to and
received (PATH_RESPONSE) from a peer on the path. Importantly, it validates
that the packets received from the migrating endpoint do not carry a spoofed
source address.
Path validation can be used at any time by either endpoint. For instance, an
endpoint might check that a peer is still in possession of its address after a
period of quiescence.
Path validation is not designed as a NAT traversal mechanism. Though the
mechanism described here might be effective for the creation of NAT bindings
that support NAT traversal, the expectation is that one or other peer is able to
receive packets without first having sent a packet on that path. Effective NAT
traversal needs additional synchronization mechanisms that are not provided
here.
An endpoint MAY bundle PATH_CHALLENGE and PATH_RESPONSE frames that are used for
path validation with other frames. In particular, an endpoint may pad a packet
carrying a PATH_CHALLENGE for PMTU discovery, or an endpoint may bundle a
PATH_RESPONSE with its own PATH_CHALLENGE.
When probing a new path, an endpoint might want to ensure that its peer has an
unused connection ID available for responses. The endpoint can send
NEW_CONNECTION_ID and PATH_CHALLENGE frames in the same packet. This ensures
that an unused connection ID will be available to the peer when sending a
response.
## Initiating Path Validation
To initiate path validation, an endpoint sends a PATH_CHALLENGE frame containing
a random payload on the path to be validated.
An endpoint MAY send multiple PATH_CHALLENGE frames to guard against packet
loss, however an endpoint SHOULD NOT send multiple PATH_CHALLENGE frames in a
single packet. An endpoint SHOULD NOT send a PATH_CHALLENGE more frequently
than it would an Initial packet, ensuring that connection migration is no more
load on a new path than establishing a new connection.
The endpoint MUST use unpredictable data in every PATH_CHALLENGE frame so that
it can associate the peer's response with the corresponding PATH_CHALLENGE.
## Path Validation Responses
On receiving a PATH_CHALLENGE frame, an endpoint MUST respond immediately by
echoing the data contained in the PATH_CHALLENGE frame in a PATH_RESPONSE frame.
An endpoint MUST NOT send more than one PATH_RESPONSE frame in response to one
PATH_CHALLENGE frame (see {{retransmission-of-information}}). The peer is
expected to send more PATH_CHALLENGE frames as necessary to evoke additional
PATH_RESPONSE frames.
## Successful Path Validation
A new address is considered valid when a PATH_RESPONSE frame is received that
contains the data that was sent in a previous PATH_CHALLENGE. Receipt of an
acknowledgment for a packet containing a PATH_CHALLENGE frame is not adequate
validation, since the acknowledgment can be spoofed by a malicious peer.
Note that receipt on a different local address does not result in path
validation failure, as it might be a result of a forwarded packet (see
{{off-path-forward}}) or misrouting. It is possible that a valid PATH_RESPONSE
might be received in the future.
## Failed Path Validation
Path validation only fails when the endpoint attempting to validate the path
abandons its attempt to validate the path.
Endpoints SHOULD abandon path validation based on a timer. When setting this
timer, implementations are cautioned that the new path could have a longer
round-trip time than the original. A value of three times the larger of the
current Probe Timeout (PTO) or the initial timeout (that is, 2*kInitialRtt) as
defined in {{QUIC-RECOVERY}} is RECOMMENDED. That is:
~~~
validation_timeout = max(3*PTO, 6*kInitialRtt)
~~~
Note that the endpoint might receive packets containing other frames on the new
path, but a PATH_RESPONSE frame with appropriate data is required for path
validation to succeed.
When an endpoint abandons path validation, it determines that the path is
unusable. This does not necessarily imply a failure of the connection -
endpoints can continue sending packets over other paths as appropriate. If no
paths are available, an endpoint can wait for a new path to become available or
close the connection.
A path validation might be abandoned for other reasons besides
failure. Primarily, this happens if a connection migration to a new path is
initiated while a path validation on the old path is in progress.
# Connection Migration {#migration}
The use of a connection ID allows connections to survive changes to endpoint
addresses (IP address and port), such as those caused by an
endpoint migrating to a new network. This section describes the process by
which an endpoint migrates to a new address.
The design of QUIC relies on endpoints retaining a stable address for the
duration of the handshake. An endpoint MUST NOT initiate connection migration
before the handshake is confirmed, as defined in section 4.1.2 of {{QUIC-TLS}}.
An endpoint also MUST NOT send packets from a different local address, actively
initiating migration, if the peer sent the `disable_active_migration` transport
parameter during the handshake. An endpoint which has sent this transport
parameter, but detects that a peer has nonetheless migrated to a different
network MUST either drop the incoming packets on that path without generating a
stateless reset or proceed with path validation and allow the peer to migrate.
Generating a stateless reset or closing the connection would allow third parties
in the network to cause connections to close by spoofing or otherwise
manipulating observed traffic.
Not all changes of peer address are intentional, or active, migrations. The peer
could experience NAT rebinding: a change of address due to a middlebox, usually
a NAT, allocating a new outgoing port or even a new outgoing IP address for a
flow. An endpoint MUST perform path validation ({{migrate-validate}}) 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 {{preferred-address}}. Clients are responsible for initiating all
migrations. Servers do not send non-probing packets (see {{probing}}) 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.
## Probing a New Path {#probing}
An endpoint MAY probe for peer reachability from a new local address using path
validation {{migrate-validate}} prior to migrating the connection to the new
local address. Failure of path validation simply means that the new path is not
usable for this connection. Failure to validate a path does not cause the
connection to end unless there are no valid alternative paths available.
An endpoint uses a new connection ID for probes sent from a new local address,
see {{migration-linkability}} for further discussion. An endpoint that uses
a new local address needs to ensure that at least one new connection ID is
available at the peer. That can be achieved by including a NEW_CONNECTION_ID
frame in the probe.
Receiving a PATH_CHALLENGE frame from a peer indicates that the peer is probing
for reachability on a path. An endpoint sends a PATH_RESPONSE in response as per
{{migrate-validate}}.
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".
## Initiating Connection Migration {#initiating-migration}
An endpoint can migrate a connection to a new local address by sending packets
containing non-probing frames from that address.
Each endpoint validates its peer's address during connection establishment.
Therefore, a migrating endpoint can send to its peer knowing that the peer is
willing to receive at the peer's current address. Thus an endpoint can migrate
to a new local address without first validating the peer's address.
When migrating, the new path might not support the endpoint's current sending
rate. Therefore, the endpoint resets its congestion controller, as described in
{{migration-cc}}.
The new path might not have the same ECN capability. Therefore, the endpoint
verifies ECN capability as described in {{ecn}}.
Receiving acknowledgments for data sent on the new path serves as proof of the
peer's reachability from the new address. Note that since acknowledgments may
be received on any path, return reachability on the new path is not
established. To establish return reachability on the new path, an endpoint MAY
concurrently initiate path validation {{migrate-validate}} on the new path.
## Responding to Connection Migration {#migration-response}
Receiving a packet from a new peer address containing a non-probing frame
indicates that the peer has migrated to that address.
In response to such a packet, an endpoint MUST start sending subsequent packets
to the new peer address and MUST initiate path validation ({{migrate-validate}})
to verify the peer's ownership of the unvalidated address.
An endpoint MAY send data to an unvalidated peer address, but it MUST protect
against potential attacks as described in {{address-spoofing}} and
{{on-path-spoofing}}. An endpoint MAY skip validation of a peer address if that
address has been seen recently.
An endpoint only changes the address that it sends packets to in response to the
highest-numbered non-probing packet. This ensures that an endpoint does not send
packets to an old peer address in the case that it receives reordered packets.
After changing the address to which it sends non-probing packets, an endpoint
could abandon any path validation for other addresses.
Receiving a packet from a new peer address might be the result of a NAT
rebinding at the peer.
After verifying a new client address, the server SHOULD send new address
validation tokens ({{address-validation}}) to the client.
### Peer Address Spoofing {#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 {{migration-response}}, an endpoint is required to validate a
peer's new address to confirm the peer's possession of the new address. Until a
peer's address is deemed valid, an endpoint MUST limit the rate at which it
sends data to this address. The endpoint MUST NOT send more than a minimum
congestion window's worth of data per estimated round-trip time (kMinimumWindow,
as defined in {{QUIC-RECOVERY}}). In the absence of this limit, an endpoint
risks being used for a denial of service attack against an unsuspecting victim.
Note that since the endpoint will not have any round-trip time measurements to
this address, the estimate SHOULD be the default initial value (see
{{QUIC-RECOVERY}}).
If an endpoint skips validation of a peer address as described in
{{migration-response}}, it does not need to limit its sending rate.
### On-Path Address Spoofing {#on-path-spoofing}
An on-path attacker could cause a spurious connection migration by copying and
forwarding a packet with a spoofed address such that it arrives before the
original packet. The packet with the spoofed address will be seen to come from
a migrating connection, and the original packet will be seen as a duplicate and
dropped. After a spurious migration, validation of the source address will fail
because the entity at the source address does not have the necessary
cryptographic keys to read or respond to the PATH_CHALLENGE frame that is sent
to it even if it wanted to.
To protect the connection from failing due to such a spurious migration, an
endpoint MUST revert to using the last validated peer address when validation of
a new peer address fails.
If an endpoint has no state about the last validated peer address, it MUST close
the connection silently by discarding all connection state. This results in new
packets on the connection being handled generically. For instance, an endpoint
MAY send a stateless reset in response to any further incoming packets.
Note that receipt of packets with higher packet numbers from the legitimate peer
address will trigger another connection migration. This will cause the
validation of the address of the spurious migration to be abandoned.
### Off-Path Packet Forwarding {#off-path-forward}
An off-path attacker that can observe packets might forward copies of genuine
packets to endpoints. If the copied packet arrives before the genuine packet,
this will appear as a NAT rebinding. Any genuine packet will be discarded as a
duplicate. If the attacker is able to continue forwarding packets, it might be
able to cause migration to a path via the attacker. This places the attacker on
path, giving it the ability to observe or drop all subsequent packets.
Unlike the attack described in {{on-path-spoofing}}, the attacker can ensure
that the new path is successfully validated.
This style of attack relies on the attacker using a path that is approximately
as fast as the direct path between endpoints. The attack is more reliable if
relatively few packets are sent or if packet loss coincides with the attempted
attack.
A non-probing packet received on the original path that increases the maximum
received packet number will cause the endpoint to move back to that path.
Eliciting packets on this path increases the likelihood that the attack is
unsuccessful. Therefore, mitigation of this attack relies on triggering the
exchange of packets.
In response to an apparent migration, endpoints MUST validate the previously
active path using a PATH_CHALLENGE frame. This induces the sending of new
packets on that path. If the path is no longer viable, the validation attempt
will time out and fail; if the path is viable, but no longer desired, the
validation will succeed, but only results in probing packets being sent on the
path.
An endpoint that receives a PATH_CHALLENGE on an active path SHOULD send a
non-probing packet in response. If the non-probing packet arrives before any
copy made by an attacker, this results in the connection being migrated back to
the original path. Any subsequent migration to another path restarts this
entire process.
This defense is imperfect, but this is not considered a serious problem. If the
path via the attack is reliably faster than the original path despite multiple
attempts to use that original path, it is not possible to distinguish between
attack and an improvement in routing.
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.
## Loss Detection and Congestion Control {#migration-cc}
The capacity available on the new path might not be the same as the old path.
Packets sent on the old path SHOULD NOT contribute to congestion control or RTT
estimation for the new path.
On confirming a peer's ownership of its new address, an endpoint MUST
immediately reset the congestion controller and round-trip time estimator for
the new path to initial values (see Sections A.3 and B.3 in {{QUIC-RECOVERY}})
unless it has knowledge that a previous send rate or round-trip time estimate is
valid for the new path. For instance, an endpoint might infer that a change in
only the client's port number is indicative of a NAT rebinding, meaning that the
new path is likely to have similar bandwidth and round-trip time. However, this
determination will be imperfect. If the determination is incorrect, the
congestion controller and the RTT estimator are expected to adapt to the new
path. Generally, implementations are advised to be cautious when using previous
values on a new path.
There may be apparent reordering at the receiver when an endpoint sends data and
probes from/to multiple addresses during the migration period, since the two
resulting paths may have different round-trip times. A receiver of packets on
multiple paths will still send ACK frames covering all received packets.
While multiple paths might be used during connection migration, a single
congestion control context and a single loss recovery context (as described in
{{QUIC-RECOVERY}}) may be adequate. For instance, an endpoint might delay
switching to a new congestion control context until it is confirmed that an old
path is no longer needed (such as the case in {{off-path-forward}}).
A sender can make exceptions for probe packets so that their loss detection is
independent and does not unduly cause the congestion controller to reduce its
sending rate. An endpoint might set a separate timer when a PATH_CHALLENGE is
sent, which is cancelled when 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.
## Privacy Implications of Connection Migration {#migration-linkability}
Using a stable connection ID on multiple network paths allows a passive observer
to correlate activity between those paths. An endpoint that moves between
networks might not wish to have their activity correlated by any entity other
than their peer, so different connection IDs are used when sending from
different local addresses, as discussed in {{connection-id}}. For this to be
effective endpoints need to ensure that connections IDs they provide cannot be
linked by any other entity.
At any time, endpoints MAY change the Destination Connection ID they send to a
value that has not been used on another path.
An endpoint MUST use a new connection ID if it initiates connection migration.
Using a new connection ID eliminates the use of the connection ID for linking
activity from the same connection on different networks. 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.
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.
A client might wish to reduce linkability by employing a new connection ID and
source UDP port when sending traffic after a period of inactivity. Changing the
UDP port from which it sends packets at the same time might cause the packet to
appear as a connection migration. This ensures that the mechanisms that support
migration are exercised even for clients that don't experience NAT rebindings or
genuine migrations. Changing port number can cause a peer to reset its
congestion state (see {{migration-cc}}), so the port SHOULD only be changed
infrequently.
An endpoint that exhausts available connection IDs cannot 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.
## Server's Preferred Address {#preferred-address}
QUIC allows servers to accept connections on one IP address and attempt to
transfer these connections to a more preferred address shortly after the
handshake. This is particularly useful when clients initially connect to an
address shared by multiple servers but would prefer to use a unicast address to
ensure connection stability. This section describes the protocol for migrating a
connection to a preferred server address.
Migrating a connection to a new server address mid-connection is left for future
work. If a client receives packets from a new server address not indicated by
the preferred_address transport parameter, the client SHOULD discard these
packets.
### 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 finished, the client SHOULD select one of the two
server's preferred addresses and initiate path validation (see
{{migrate-validate}}) of that address using the connection ID provided in the
preferred_address transport parameter.
If path validation succeeds, the client SHOULD immediately begin sending all
future packets to the new server address using the new connection ID and
discontinue use of the old server address. If path validation fails, the client
MUST continue sending all future packets to the server's original IP address.
### Responding to Connection Migration
A server might receive a packet addressed to its preferred IP address at any
time after it accepts a connection. If this packet contains a PATH_CHALLENGE
frame, the server sends a PATH_RESPONSE frame as per {{migrate-validate}}. The
server MUST send other non-probing frames from its original address until it
receives a non-probing packet from the client at its preferred address and until
the server has validated the new path.
The server MUST probe on the path toward the client from its preferred address.
This helps to guard against spurious migration initiated by an attacker.
Once the server has completed its path validation and has received a non-probing
packet with a new largest packet number on its preferred address, the server
begins sending non-probing packets to the client exclusively from its preferred
IP address. It SHOULD drop packets for this connection received on the old IP
address, but MAY continue to process delayed packets.
### 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 the connection to the server's preferred address is not from the same client
address, the server MUST protect against potential attacks as described in
{{address-spoofing}} and {{on-path-spoofing}}. In addition to intentional
simultaneous migration, this might also occur because the client's access
network used a different NAT binding for the server's preferred address.
Servers SHOULD initiate path validation to the client's new address upon
receiving a probe packet from a different address. Servers MUST NOT send more
than a minimum congestion window's worth of non-probing packets to the new
address before path validation is complete.
A client that migrates to a new address SHOULD use a preferred address from the
same address family for the server.
## Use of IPv6 Flow-Label and Migration {#ipv6-flow-label}
Endpoints that send data using IPv6 SHOULD apply an IPv6 flow label
in compliance with {{!RFC6437}}, unless the local API does not allow
setting IPv6 flow labels.
The IPv6 flow label SHOULD be a pseudo-random function of the source
and destination addresses, source and destination UDP ports, and the destination
CID. The flow label generation MUST be designed to minimize the chances of
linkability with a previously used flow label, as this would enable correlating
activity on multiple paths (see {{migration-linkability}}).
A possible implementation is to compute the flow label as a cryptographic hash
function of the source and destination addresses, source and destination
UDP ports, destination CID, and a local secret.
# Connection Termination {#termination}
An established QUIC connection can be terminated in one of three ways:
* idle timeout ({{idle-timeout}})
* immediate close ({{immediate-close}})
* stateless reset ({{stateless-reset}})
An endpoint MAY discard connection state if it does not have a validated path on
which it can send packets (see {{migrate-validate}}).
## Closing and Draining Connection States {#draining}
The closing and draining connection states exist to ensure that connections
close cleanly and that delayed or reordered packets are properly discarded.
These states SHOULD persist for at least three times the current Probe Timeout
(PTO) interval as defined in {{QUIC-RECOVERY}}.
An endpoint enters a closing period after initiating an immediate close
({{immediate-close}}). While closing, an endpoint MUST NOT send packets unless
they contain a CONNECTION_CLOSE frame (see {{immediate-close}} for details). An
endpoint retains only enough information to generate a packet containing a
CONNECTION_CLOSE frame and to identify packets as belonging to the connection.
The endpoint's selected connection ID and the QUIC version are sufficient
information to identify packets for a closing connection; an endpoint can
discard all other connection state. An endpoint MAY retain packet protection
keys for incoming packets to allow it to read and process a CONNECTION_CLOSE
frame.
The draining state is entered once an endpoint receives a signal that its peer
is closing or draining. While otherwise identical to the closing state, an
endpoint in the draining state MUST NOT send any packets. Retaining packet
protection keys is unnecessary once a connection is in the draining state.
An endpoint MAY transition from the closing period to the draining period if it
receives a CONNECTION_CLOSE frame or stateless reset, both of which indicate
that the peer is also closing or draining. The draining period SHOULD end when
the closing period would have ended. In other words, the endpoint can use the
same end time, but cease retransmission of the closing packet.
Disposing of connection state prior to the end of the closing or draining period
could cause delayed or reordered packets to generate an unnecessary stateless
reset. Endpoints that have some alternative means to ensure that late-arriving
packets on the connection do not induce a response, such as those that are able
to close the UDP socket, MAY use an abbreviated draining period which can allow
for faster resource recovery. Servers that retain an open socket for accepting
new connections SHOULD NOT exit the closing or draining period early.
Once the closing or draining period has ended, an endpoint SHOULD discard all
connection state. This results in new packets on the connection being handled
generically. For instance, an endpoint MAY send a stateless reset in response
to any further incoming packets.
The draining and closing periods do not apply when a stateless reset
({{stateless-reset}}) is sent.
An endpoint is not expected to handle key updates when it is closing or
draining. A key update might prevent the endpoint from moving from the closing
state to draining, but it otherwise has no impact.
While in the closing period, an endpoint could receive packets from a new source
address, indicating a connection migration ({{migration}}). An endpoint in the
closing state MUST strictly limit the number of packets it sends to this new
address until the address is validated (see {{migrate-validate}}). A server in
the closing state MAY instead choose to discard packets received from a new
source address.
## Idle Timeout {#idle-timeout}
If the idle timeout is enabled, a connection is silently closed and the state is
discarded when it remains idle for longer than both the advertised
idle timeout (see {{transport-parameter-definitions}}) and three times the
current Probe Timeout (PTO).
Each endpoint advertises its own idle timeout to its peer. An endpoint
restarts any timer it maintains when a packet from its peer is received and
processed successfully. The timer is also restarted when sending a packet
containing frames other than ACK or PADDING (an ACK-eliciting packet; see
{{QUIC-RECOVERY}}), but only if no other ACK-eliciting packets have been sent
since last receiving a packet. Restarting when sending packets ensures that
connections do not prematurely time out when initiating new activity.
The value for an idle timeout can be asymmetric. The value advertised by an
endpoint is only used to determine whether the connection is live at that
endpoint. An endpoint that sends packets near the end of the idle timeout
period of a peer risks having those packets discarded if its peer enters the
draining state before the packets arrive. If a peer could timeout within a
Probe Timeout (PTO; see Section 6.3 of {{QUIC-RECOVERY}}), it is advisable to
test for liveness before sending any data that cannot be retried safely. Note
that it is likely that only applications or application protocols will
know what information can be retried.
## Immediate Close
An endpoint sends a CONNECTION_CLOSE frame ({{frame-connection-close}}) to
terminate the connection immediately. A CONNECTION_CLOSE frame causes all
streams to immediately become closed; open streams can be assumed to be
implicitly reset.
After sending a CONNECTION_CLOSE frame, endpoints immediately enter the closing
state. During the closing period, an endpoint that sends a CONNECTION_CLOSE
frame SHOULD respond to any packet that it receives with another packet
containing a CONNECTION_CLOSE frame. To minimize the state that an endpoint
maintains for a closing connection, endpoints MAY send the exact same packet.
However, endpoints SHOULD limit the number of packets they generate containing a
CONNECTION_CLOSE frame. For instance, an endpoint could progressively increase
the number of packets that it receives before sending additional packets or
increase the time between packets.
Note:
: Allowing retransmission of a closing packet contradicts other advice in this
document that recommends the creation of new packet numbers for every packet.
Sending new packet numbers is primarily of advantage to loss recovery and
congestion control, which are not expected to be relevant for a closed
connection. Retransmitting the final packet requires less state.
New packets from unverified addresses could be used to create an amplification
attack (see {{address-validation}}). To avoid this, endpoints MUST either limit
transmission of CONNECTION_CLOSE frames to validated addresses or drop packets
without response if the response would be more than three times larger than the
received packet.
After receiving a CONNECTION_CLOSE frame, endpoints enter the draining state.
An endpoint that receives a CONNECTION_CLOSE frame MAY send a single packet
containing a CONNECTION_CLOSE frame before entering the draining state, using a
CONNECTION_CLOSE frame and a NO_ERROR code if appropriate. An endpoint MUST NOT
send further packets, which could result in a constant exchange of
CONNECTION_CLOSE frames until the closing period on either peer ended.
An immediate close can be used after an application protocol has arranged to
close a connection. This might be after the application protocols negotiates a
graceful shutdown. The application protocol exchanges whatever messages that
are needed to cause both endpoints to agree to close the connection, after which
the application requests that the connection be closed. The application
protocol can use a CONNECTION_CLOSE frame with an appropriate error code to
signal closure.
When sending CONNECTION_CLOSE, the goal is to ensure that the peer will process
the frame. Generally, this means sending the frame in a packet with the highest
level of packet protection to avoid the packet being discarded. However, during
the handshake, it is possible that more advanced packet protection keys are not
available to the peer, so the frame MAY be replicated in a packet that uses a
lower packet protection level.
After the handshake is confirmed, an endpoint MUST send any CONNECTION_CLOSE
frames in a 1-RTT packet. Prior to handshake confirmation, the peer might not
have 1-RTT keys, so the endpoint SHOULD send CONNECTION_CLOSE frames in a
Handshake packet. If the endpoint does not have Handshake keys, it SHOULD send
CONNECTION_CLOSE frames in an Initial packet.
A client will always know whether the server has Handshake keys
(see {{discard-initial}}), but it is possible that a server does not know
whether the client has Handshake keys. Under these circumstances, a server
SHOULD send a CONNECTION_CLOSE frame in both Handshake and Initial packets
to ensure that at least one of them is processable by the client. These
packets can be coalesced into a single UDP datagram (see {{packet-coalesce}}).
## Stateless Reset {#stateless-reset}
A stateless reset is provided as an option of last resort for an endpoint that
does not have access to the state of a connection. A crash or outage might
result in peers continuing to send data to an endpoint that is unable to
properly continue the connection. An endpoint MAY send a stateless reset in
response to receiving a packet that it cannot associate with an active
connection.
A stateless reset is not appropriate for signaling error conditions. An
endpoint that wishes to communicate a fatal connection error MUST use a
CONNECTION_CLOSE frame if it has sufficient state to do so.
To support this process, a token is sent by endpoints. The token is carried in
the NEW_CONNECTION_ID frame sent by either peer, and servers can specify the
stateless_reset_token transport parameter during the handshake (clients cannot
because their transport parameters don't have confidentiality protection). This
value is protected by encryption, so only client and server know this value.
Tokens are invalidated when their associated connection ID is retired via a
RETIRE_CONNECTION_ID frame ({{frame-retire-connection-id}}).
An endpoint that receives packets that it cannot process sends a packet in the
following layout:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0|1| Unpredictable Bits (38 ..) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ +
| |
+ Stateless Reset Token (128) +
| |
+ +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #fig-stateless-reset title="Stateless Reset Packet"}
This design ensures that a stateless reset packet is - to the extent possible -
indistinguishable from a regular packet with a short header.
A stateless reset uses an entire UDP datagram, starting with the first two bits
of the packet header. The remainder of the first byte and an arbitrary number
of bytes following it that are set to unpredictable values. The last 16 bytes
of the datagram contain a Stateless Reset Token.
To entities other than its intended recipient, a stateless reset will appear to
be a packet with a short header. For the stateless reset to appear as a valid
QUIC packet, the Unpredictable Bits field needs to include at least 38 bits of
data (or 5 bytes, less the two fixed bits).
A minimum size of 21 bytes does not guarantee that a stateless reset is
difficult to distinguish from other packets if the recipient requires the use of
a connection ID. To prevent a resulting stateless reset from being trivially
distinguishable from a valid packet, all packets sent by an endpoint SHOULD be
padded to at least 22 bytes longer than the minimum connection ID that the
endpoint might use. An endpoint that sends a stateless reset in response to
packet that is 43 bytes or less in length SHOULD send a stateless reset that is
one byte shorter than the packet it responds to.
These values assume that the Stateless Reset Token is the same as the minimum
expansion of the packet protection AEAD. Additional unpredictable bytes are
necessary if the endpoint could have negotiated a packet protection scheme with
a larger minimum expansion.
An endpoint MUST NOT send a stateless reset that is three times or more larger
than the packet it receives to avoid being used for amplification.
{{reset-looping}} describes additional limits on stateless reset size.
Endpoints MUST discard packets that are too small to be valid QUIC packets.
With the set of AEAD functions defined in {{QUIC-TLS}}, packets that are smaller
than 21 bytes are never valid.
Endpoints MUST send stateless reset packets formatted as a packet with a short
header. However, endpoints MUST treat any packet ending in a valid stateless
reset token as a stateless reset, as other QUIC versions might allow the use of
a long header.
An endpoint MAY send a stateless reset in response to a packet with a long
header. Sending a stateless reset is not effective prior to the stateless reset
token being available to a peer. In this QUIC version, packets with a long
header are only used during connection establishment. Because the stateless
reset token is not available until connection establishment is complete or near
completion, ignoring an unknown packet with a long header might be as effective
as sending a stateless reset.
An endpoint cannot determine the Source Connection ID from a packet with a short
header, therefore it cannot set the Destination Connection ID in the stateless
reset packet. The Destination 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 ({{frame-new-connection-id}}).
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
{{reset-looping}}. A Stateless Reset that is not correctly routed is
an ineffective error detection and recovery mechanism. In this
case, endpoints will need to rely on other methods - such as timers - to
detect that the connection has failed.
* The randomly generated connection ID can be used by entities other than the
peer to identify this as a potential stateless reset. An endpoint that
occasionally uses different connection IDs might introduce some uncertainty
about this.
This stateless reset design is specific to QUIC version 1. An endpoint that
supports multiple versions of QUIC needs to generate a stateless reset that will
be accepted by peers that support any version that the endpoint might support
(or might have supported prior to losing state). Designers of new versions of
QUIC need to be aware of this and either reuse this design, or use a portion of
the packet other than the last 16 bytes for carrying data.
### Detecting a Stateless Reset
An endpoint detects a potential stateless reset when an incoming packet either
cannot be associated with a connection, cannot be decrypted, or is marked as a
duplicate packet. The endpoint MUST then compare the last 16 bytes of the
packet with all Stateless Reset Tokens that are associated with connection IDs
that the endpoint recently used to send packets from the IP address and port on
which the datagram is received. This includes Stateless Reset Tokens from
NEW_CONNECTION_ID frames and the server's transport parameters. 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.
If the last 16 bytes of the packet values are identical to a Stateless Reset
Token, the endpoint MUST enter the draining period and not send any further
packets on this connection. If the comparison fails, the packet can be
discarded.
### Calculating a Stateless Reset Token {#reset-token}
The stateless reset token MUST be difficult to guess. In order to create a
Stateless Reset Token, an endpoint could randomly generate {{!RFC4086}} a secret
for every connection that it creates. However, this presents a coordination
problem when there are multiple instances in a cluster or a storage problem for
an endpoint that might lose state. Stateless reset specifically exists to
handle the case where state is lost, so this approach is suboptimal.
A single static key can be used across all connections to the same endpoint by
generating the proof using a second iteration of a preimage-resistant function
that takes a static key and the connection ID chosen by the endpoint (see
{{connection-id}}) as input. An endpoint could use HMAC {{?RFC2104}} (for
example, HMAC(static_key, connection_id)) or HKDF {{?RFC5869}} (for example,
using the static key as input keying material, with the connection ID as salt).
The output of this function is truncated to 16 bytes to produce the Stateless
Reset Token for that connection.
An endpoint that loses state can use the same method to generate a valid
Stateless Reset Token. The connection ID comes from the packet that the
endpoint receives.
This design relies on the peer always sending a connection ID in its packets so
that the endpoint can use the connection ID from a packet to reset the
connection. An endpoint that uses this design MUST 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 {{reset-oracle}}. 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 MAY be used for multiple connection IDs on the
same connection. However, reuse of a Stateless Reset Token might expose an
endpoint to denial of service if associated connection IDs are forgotten while
the associated token is still active at a peer. An endpoint MUST ensure that
packets with Destination Connection ID field values that correspond to a reused
Stateless Reset Token are attributed to the same connection as long as the
Stateless Reset Token is still usable, even when the connection ID has been
retired. Otherwise, an attacker might be able to send a packet with a retired
connection ID and cause the endpoint to produce a Stateless Reset that it can
use to disrupt the connection, just as with the attacks in {{reset-oracle}}.
Note that Stateless Reset packets do not have any cryptographic protection.
### Looping {#reset-looping}
The design of a Stateless Reset is such that without knowing the stateless reset
token it is indistinguishable from a valid packet. For instance, if a server
sends a Stateless Reset to another server it might receive another Stateless
Reset in response, which could lead to an infinite exchange.
An endpoint MUST ensure that every Stateless Reset that it sends is smaller than
the packet which triggered it, unless it maintains state sufficient to prevent
looping. In the event of a loop, this results in packets eventually being too
small to trigger a response.
An endpoint can remember the number of Stateless Reset packets that it has sent
and stop generating new Stateless Reset packets once a limit is reached. Using
separate limits for different remote addresses will ensure that Stateless Reset
packets can be used to close connections when other peers or connections have
exhausted limits.
Reducing the size of a Stateless Reset below 41 bytes means that the packet
could reveal to an observer that it is a Stateless Reset, depending upon the
length of the peer's connection IDs. Conversely, refusing to send a Stateless
Reset in response to a small packet might result in Stateless Reset not being
useful in detecting cases of broken connections where only very small packets
are sent; such failures might only be detected by other means, such as timers.
# Error Handling {#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 {{connection-errors}}), while only application-level
errors can be isolated to a single stream (see {{stream-errors}}).
The most appropriate error code ({{error-codes}}) 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 ({{stateless-reset}}) 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.
## 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
({{frame-connection-close}}). An endpoint MAY close the connection in this
manner even if the error only affects a single stream.
Application protocols can signal application-specific protocol errors using the
application-specific variant of the CONNECTION_CLOSE frame. Errors that are
specific to the transport, including all those described in this document, are
carried in the QUIC-specific variant of the CONNECTION_CLOSE frame.
A CONNECTION_CLOSE frame could be sent in a packet that is lost. An endpoint
SHOULD be prepared to retransmit a packet containing a CONNECTION_CLOSE frame if
it receives more packets on a terminated connection. Limiting the number of
retransmissions and the time over which this final packet is sent limits the
effort expended on terminated connections.
An endpoint that chooses not to retransmit packets containing a CONNECTION_CLOSE
frame risks a peer missing the first such packet. The only mechanism available
to an endpoint that continues to receive data for a terminated connection is to
use the stateless reset process ({{stateless-reset}}).
An endpoint that receives an invalid CONNECTION_CLOSE frame MUST NOT signal the
existence of the error to its peer.
## 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
({{frame-reset-stream}}) with an appropriate error code to terminate just the
affected stream.
RESET_STREAM MUST be instigated by the protocol using QUIC. RESET_STREAM
carries an application error code. 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.
Resetting a stream without knowledge of the application protocol could cause the
protocol to enter an unrecoverable state. Application protocols might require
certain streams to be reliably delivered in order to guarantee consistent state
between endpoints. Application protocols SHOULD define rules for handling
streams that are prematurely cancelled by either endpoint.
# Packets and Frames {#packets-frames}
QUIC endpoints communicate by exchanging packets. Packets have confidentiality
and integrity protection (see {{packet-protected}}) and are carried in UDP
datagrams (see {{packet-coalesce}}).
This version of QUIC uses the long packet header (see {{long-header}}) during
connection establishment. Packets with the long header are Initial
({{packet-initial}}), 0-RTT ({{packet-0rtt}}), Handshake ({{packet-handshake}}),
and Retry ({{packet-retry}}). Version negotiation uses a version-independent
packet with a long header (see {{packet-version}}).
Packets with the short header ({{short-header}}) are designed for minimal
overhead and are used after a connection is established and 1-RTT keys are
available.
## Protected Packets {#packet-protected}
All QUIC packets except Version Negotiation and Retry packets use authenticated
encryption with additional data (AEAD) {{!RFC5116}} to provide confidentiality
and integrity protection. Details of packet protection are found in
{{QUIC-TLS}}; this section includes an overview of the process.
Initial packets are protected using keys that are statically derived. This
packet protection is not effective confidentiality protection. Initial
protection only exists to ensure that the sender of the packet is on the network
path. Any entity that receives the Initial packet from a client can recover the
keys necessary to remove packet protection or to generate packets that will be
successfully authenticated.
All other packets are protected with keys derived from the cryptographic
handshake. The type of the packet from the long header or key phase from the
short header are used to identify which encryption level - and therefore the
keys - that are used. Packets protected with 0-RTT and 1-RTT keys are expected
to have confidentiality and data origin authentication; the cryptographic
handshake ensures that only the communicating endpoints receive the
corresponding keys.
The packet number field contains a packet number, which has additional
confidentiality protection that is applied after packet protection is applied
(see {{QUIC-TLS}} for details). The underlying packet number increases with
each packet sent in a given packet number space; see {{packet-numbers}} for
details.
## Coalescing Packets {#packet-coalesce}
Initial ({{packet-initial}}), 0-RTT ({{packet-0rtt}}), and Handshake
({{packet-handshake}}) packets contain a Length field, which determines the end
of the packet. The length includes both the Packet Number and Payload
fields, both of which are confidentiality protected and initially of unknown
length. The length of the Payload field is learned once header protection is
removed.
Using the Length field, a sender can coalesce multiple QUIC packets into one UDP
datagram. This can reduce the number of UDP datagrams needed to complete the
cryptographic handshake and start sending data. This can also be used to
construct PMTU probes (see {{pmtu-probes-src-cid}}). Receivers MUST be able to
process coalesced packets.
Coalescing packets in order of increasing encryption levels (Initial, 0-RTT,
Handshake, 1-RTT) makes it more likely 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 NOT coalesce multiple packets at the same encryption level.
Senders MUST NOT coalesce QUIC packets for different connections into a single
UDP datagram. Receivers SHOULD ignore any subsequent packets with a different
Destination Connection ID than the first packet in the datagram.
Every QUIC packet that is coalesced into a single UDP datagram is separate and
complete. Though the values of some fields in the packet header might be
redundant, no fields are omitted. The receiver of coalesced QUIC packets MUST
individually process each QUIC packet and separately acknowledge them, as if
they were received as the payload of different UDP datagrams. For example, if
decryption fails (because the keys are not available or any other reason),
the receiver MAY either discard or buffer the packet for later processing and
MUST attempt to process the remaining packets.
Retry packets ({{packet-retry}}), Version Negotiation packets
({{packet-version}}), and packets with a short header ({{short-header}}) 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.
## Packet Numbers {#packet-numbers}
The packet number is an integer in the range 0 to 2^62-1. This number is used
in determining the cryptographic nonce for packet protection. Each endpoint
maintains a separate packet number for sending and receiving.
Packet numbers are limited to this range because they need to be representable
in whole in the Largest Acknowledged field of an ACK frame ({{frame-ack}}).
When present in a long or short header however, packet numbers are reduced and
encoded in 1 to 4 bytes (see {{packet-encoding}}).
Version Negotiation ({{packet-version}}) and Retry ({{packet-retry}}) packets
do not include a packet number.
Packet numbers are divided into 3 spaces in QUIC:
- Initial space: All Initial packets ({{packet-initial}}) are in this space.
- Handshake space: All Handshake packets ({{packet-handshake}}) are in this
space.
- Application data space: All 0-RTT and 1-RTT encrypted packets
({{packet-protected}}) are in this space.
As described in {{QUIC-TLS}}, each packet type uses different protection keys.
Conceptually, a packet number space is the context in which a packet can be
processed and acknowledged. Initial packets can only be sent with Initial
packet protection keys and acknowledged in packets which are also Initial
packets. Similarly, Handshake packets are sent at the Handshake encryption
level and can only be acknowledged in Handshake packets.
This enforces cryptographic separation between the data sent in the different
packet sequence number spaces. Packet numbers in each space start at packet
number 0. Subsequent packets sent in the same packet number space MUST increase
the packet number by at least one.
0-RTT and 1-RTT data exist in the same packet number space to make loss recovery
algorithms easier to implement between the two packet types.
A QUIC endpoint MUST NOT reuse a packet number within the same packet number
space in one connection. If the packet number for sending reaches 2^62 - 1, the
sender MUST close the connection without sending a CONNECTION_CLOSE frame or any
further packets; an endpoint MAY send a Stateless Reset ({{stateless-reset}}) in
response to further packets that it receives.
A receiver MUST discard a newly unprotected packet unless it is certain that it
has not processed another packet with the same packet number from the same
packet number space. Duplicate suppression MUST happen after removing packet
protection for the reasons described in Section 9.3 of {{QUIC-TLS}}. An
efficient algorithm for duplicate suppression can be found in Section 3.4.3 of
{{?RFC4303}}.
Packet number encoding at a sender and decoding at a receiver are described in
{{packet-encoding}}.
## Frames and Frame Types {#frames}
The payload of QUIC packets, after removing packet protection, consists of a
sequence of complete frames, as shown in {{packet-frames}}. Version
Negotiation, Stateless Reset, and Retry packets do not contain frames.
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Frame 1 (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Frame 2 (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Frame N (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #packet-frames title="QUIC Payload"}
The payload of a packet that contains frames MUST contain at least one frame,
and MAY contain multiple frames and multiple frame types. Frames always fit
within a single QUIC packet and cannot span multiple packets.
Each frame begins with a Frame Type, indicating its type, followed by
additional type-dependent fields:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Frame Type (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Type-Dependent Fields (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #frame-layout title="Generic Frame Layout"}
The frame types defined in this specification are listed in {{frame-types}}.
The Frame Type in ACK, STREAM, MAX_STREAMS, STREAMS_BLOCKED, and
CONNECTION_CLOSE frames is used to carry other frame-specific flags. For all
other frames, the Frame Type field simply identifies the frame. These
frames are explained in more detail in {{frame-formats}}.
| Type Value | Frame Type Name | Definition |
|:------------|:---------------------|:-------------------------------|
| 0x00 | PADDING | {{frame-padding}} |
| 0x01 | PING | {{frame-ping}} |
| 0x02 - 0x03 | ACK | {{frame-ack}} |
| 0x04 | RESET_STREAM | {{frame-reset-stream}} |
| 0x05 | STOP_SENDING | {{frame-stop-sending}} |
| 0x06 | CRYPTO | {{frame-crypto}} |
| 0x07 | NEW_TOKEN | {{frame-new-token}} |
| 0x08 - 0x0f | STREAM | {{frame-stream}} |
| 0x10 | MAX_DATA | {{frame-max-data}} |
| 0x11 | MAX_STREAM_DATA | {{frame-max-stream-data}} |
| 0x12 - 0x13 | MAX_STREAMS | {{frame-max-streams}} |
| 0x14 | DATA_BLOCKED | {{frame-data-blocked}} |
| 0x15 | STREAM_DATA_BLOCKED | {{frame-stream-data-blocked}} |
| 0x16 - 0x17 | STREAMS_BLOCKED | {{frame-streams-blocked}} |
| 0x18 | NEW_CONNECTION_ID | {{frame-new-connection-id}} |
| 0x19 | RETIRE_CONNECTION_ID | {{frame-retire-connection-id}} |
| 0x1a | PATH_CHALLENGE | {{frame-path-challenge}} |
| 0x1b | PATH_RESPONSE | {{frame-path-response}} |
| 0x1c - 0x1d | CONNECTION_CLOSE | {{frame-connection-close}} |
{: #frame-types title="Frame Types"}
An endpoint MUST treat the receipt of a frame of unknown type as a connection
error of type FRAME_ENCODING_ERROR.
All QUIC frames are idempotent in this version of QUIC. That is, a valid
frame does not cause undesirable side effects or errors when received more
than once.
The Frame Type field uses a variable length integer encoding (see
{{integer-encoding}}) with one exception. To ensure simple and efficient
implementations of frame parsing, a frame type MUST use the shortest possible
encoding. Though a two-, four- or eight-byte encoding of the frame types
defined in this document is possible, the Frame Type field for these frames is
encoded on a single byte. 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. 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.
# Packetization and Reliability {#packetization}
A sender bundles one or more frames in a QUIC packet (see {{frames}}).
A sender can minimize per-packet bandwidth and computational costs by bundling
as many frames as possible within a QUIC packet. A sender MAY wait for a short
period of time to bundle multiple frames before sending a packet that is not
maximally packed, to avoid sending out large numbers of small packets. An
implementation MAY use knowledge about application sending behavior or
heuristics to determine whether and for how long to wait. This waiting period
is an implementation decision, and an implementation should be careful to delay
conservatively, since any delay is likely to increase application-visible
latency.
Stream multiplexing is achieved by interleaving STREAM frames from multiple
streams into one or more QUIC packets. A single QUIC packet can include
multiple STREAM frames from one or more streams.
One of the benefits of QUIC is avoidance of head-of-line blocking across
multiple streams. When a packet loss occurs, only streams with data in that
packet are blocked waiting for a retransmission to be received, while other
streams can continue making progress. Note that when data from multiple streams
is bundled into a single QUIC packet, loss of that packet blocks all those
streams from making progress. Implementations are advised to bundle as few
streams as necessary in outgoing packets without losing transmission efficiency
to underfilled packets.
## Packet Processing {#processing}
A packet MUST NOT be acknowledged until packet protection has been successfully
removed and all frames contained in the packet have been processed. For STREAM
frames, this means the data has been enqueued in preparation to be received by
the application protocol, but it does not require that data is delivered and
consumed.
Once the packet has been fully processed, a receiver acknowledges receipt by
sending one or more ACK frames containing the packet number of the received
packet.
<!-- TODO: Do we need to say anything about partial processing. And our
expectations about what implementations do with packets that have errors after
valid frames? -->
## Generating Acknowledgements {#generating-acks}
Endpoints acknowledge all packets they receive and process. However, only
ack-eliciting packets (see {{QUIC-RECOVERY}}) trigger the sending of an ACK
frame. Packets that are not ack-eliciting are only acknowledged when an ACK
frame is sent for other reasons.
When sending a packet for any reason, an endpoint should attempt to bundle an
ACK frame if one has not been sent recently. Doing so helps with timely loss
detection at the peer.
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.
### Sending ACK Frames
An ACK frame SHOULD be generated for at least every second ack-eliciting packet.
This recommendation is in keeping with standard practice for TCP {{?RFC5681}}.
A receiver's delayed acknowledgment timer SHOULD NOT exceed the current RTT
estimate or the value it indicates in the `max_ack_delay` transport parameter.
This ensures an acknowledgment is sent at least once per RTT when packets
needing acknowledgement are received. The sender can use the receiver's
`max_ack_delay` value in determining timeouts for timer-based retransmission.
In order to assist loss detection at the sender, an endpoint SHOULD send an ACK
frame immediately on receiving an ack-eliciting packet that is out of order. The
endpoint MAY continue sending ACK frames immediately on each subsequently
received packet, but the endpoint SHOULD return to acknowledging every other
packet after a period of 1/8 x RTT, unless more ACK-eliciting packets are
received out of order. If every subsequent ACK-eliciting packet arrives out of
order, then an ACK frame SHOULD be sent immediately for every received
ACK-eliciting packet.
Similarly, packets marked with the ECN Congestion Experienced (CE) codepoint in
the IP header SHOULD be acknowledged immediately, to reduce the peer's response
time to congestion events.
As an optimization, a receiver MAY process multiple packets before sending any
ACK frames in response. In this case the receiver can determine whether an
immediate or delayed acknowledgement should be generated after processing
incoming packets.
Acknowledgements of packets carrying CRYPTO frames SHOULD be minimally delayed,
to complete the handshake with minimal latency. Delaying them by a small amount,
such as the local timer granularity, allows the endpoint to bundle any data sent
in response with the ACK frame. ACK frames SHOULD be sent immediately when the
crypto stack indicates all data for that packet number space has been received.
Packets containing PADDING frames are considered to be in flight for congestion
control purposes {{QUIC-RECOVERY}}. Sending only PADDING frames might cause the
sender to become limited by the congestion controller (as described in
{{QUIC-RECOVERY}}) with no acknowledgments forthcoming from the
receiver. Therefore, a sender SHOULD ensure that other frames are sent in
addition to PADDING frames to elicit acknowledgments from the receiver.
An endpoint that is only sending ACK frames will not receive acknowledgments
from its peer unless those acknowledgements are included in packets with
ACK-eliciting frames. An endpoint SHOULD bundle ACK frames with other frames
when there are new ACK-eliciting packets to acknowledge. When only
non-ACK-eliciting packets need to be acknowledged, an endpoint MAY wait until an
ACK-eliciting packet has been received to bundle an ACK frame with outgoing
frames.
The algorithms in {{QUIC-RECOVERY}} are resilient to receivers that do not
follow guidance offered above. However, an implementor should only deviate from
these requirements after careful consideration of the performance implications
of doing so.
Packets containing only ACK frames are not congestion controlled, so there are
limits on how frequently they can be sent. An endpoint MUST NOT send more than
one ACK-frame-only packet in response to receiving an ACK-eliciting packet (one
containing frames other than ACK and/or PADDING). An endpoint MUST NOT send a
packet containing only an ACK frame in response to a non-ACK-eliciting packet
(one containing only ACK and/or PADDING frames), even if there are packet gaps
which precede the received packet. Limiting ACK frames avoids an infinite
feedback loop of acknowledgements, which could prevent the connection from ever
becoming idle. However, the endpoint acknowledges non-ACK-eliciting packets when
it sends an ACK frame.
An endpoint SHOULD treat receipt of an acknowledgment for a packet it did not
send as a connection error of type PROTOCOL_VIOLATION, if it is able to detect
the condition.
### Managing ACK Ranges
When an ACK frame is sent, one or more ranges of acknowledged packets are
included. Including older packets reduces the chance of spurious retransmits
caused by losing previously sent ACK frames, at the cost of larger ACK frames.
ACK frames SHOULD always acknowledge the most recently received packets, and the
more out-of-order the packets are, the more important it is to send an updated
ACK frame quickly, to prevent the peer from declaring a packet as lost and
spuriously retransmitting the frames it contains.
{{ack-tracking}} and {{ack-limiting}} describe an exemplary approach for
determining what packets to acknowledge in each ACK frame.
### Receiver Tracking of ACK Frames {#ack-tracking}
When a packet containing an ACK frame is sent, the largest acknowledged in that
frame may be saved. When a packet containing an ACK frame is acknowledged, the
receiver can stop acknowledging packets less than or equal to the largest
acknowledged in the sent ACK frame.
In cases without ACK frame loss, this algorithm allows for a minimum of 1 RTT
of reordering. In cases with ACK frame loss and reordering, this approach does
not guarantee that every acknowledgement is seen by the sender before it is no
longer included in the ACK frame. Packets could be received out of order and
all subsequent ACK frames containing them could be lost. In this case, the
loss recovery algorithm could cause spurious retransmits, but the sender will
continue making forward progress.
### Limiting ACK Ranges {#ack-limiting}
To limit ACK Ranges (see {{ack-ranges}}) to those that have not yet been
received by the sender, the receiver SHOULD track which ACK frames have been
acknowledged by its peer. The receiver SHOULD exclude already acknowledged
packets from future ACK frames whenever these packets would unnecessarily
contribute to the ACK frame size. When the receiver is only sending
non-ACK-eliciting packets, it can bundle a PING or other small ACK-eliciting
frame with a fraction of them, such as once per round trip, to enable
dropping unnecessary ACK ranges and any state for previously sent packets.
The receiver MUST NOT bundle an ACK-eliciting frame, such as a PING, with all
packets that would otherwise be non-ACK-eliciting, in order to avoid an
infinite feedback loop of acknowledgements.
To limit receiver state or the size of ACK frames, a receiver MAY limit the
number of ACK Ranges it sends. A receiver can do this even without receiving
acknowledgment of its ACK frames, with the knowledge this could cause the sender
to unnecessarily retransmit some data. Standard QUIC algorithms
({{QUIC-RECOVERY}}) declare packets lost after sufficiently newer packets are
acknowledged. Therefore, the receiver SHOULD repeatedly acknowledge newly
received packets in preference to packets received in the past.
### Measuring and Reporting Host Delay {#host-delay}
An endpoint measures the delays intentionally introduced between when an
ACK-eliciting packet is received and the corresponding acknowledgment is sent.
The endpoint encodes this delay for the largest acknowledged packet in the Ack
Delay field of an ACK frame (see {{frame-ack}}). This allows the receiver of the
ACK to adjust for any intentional delays, which is important for getting a
better estimate of the path RTT when acknowledgments are delayed. A packet might
be held in the OS kernel or elsewhere on the host before being processed. An
endpoint MUST NOT include delays that is does not control when populating the
Ack Delay field in an ACK frame.
An endpoint MUST NOT excessively delay acknowledgements of ack-eliciting
packets. An endpoint commits to a maximum delay using the max_ack_delay
transport parameter; see {{transport-parameter-definitions}}. max_ack_delay
declares an explicit contract: an endpoint promises to never 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 delayed
retransmissions from the peer. For Initial and Handshake packets, a
max_ack_delay of 0 is used.
### ACK Frames and Packet Protection
ACK frames MUST only be carried in a packet that has the same packet
number space as the packet being ACKed (see {{packet-protected}}). 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.
## 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 encryption level 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.
* The most recent set of acknowledgments are sent in ACK frames. An ACK frame
SHOULD contain all unacknowledged acknowledgments, as described in
{{sending-ack-frames}}.
* 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
{{solicited-state-transitions}}.
* Connection close signals, including packets that contain CONNECTION_CLOSE
frames, are not sent again when packet loss is detected, but as described in
{{termination}}.
* 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.
* The current maximum stream data offset is sent in MAX_STREAM_DATA frames.
Like MAX_DATA, an updated value is sent when the packet containing the most
recent MAX_STREAM_DATA frame for a stream is lost or when the limit is
updated, with care taken to prevent the frame from being sent too often. An
endpoint SHOULD stop sending MAX_STREAM_DATA frames when the receiving part of
the stream enters a "Size Known" state.
* The limit on streams of a given type is sent in MAX_STREAMS frames. Like
MAX_DATA, an updated value is sent when a packet containing the most recent
MAX_STREAMS for a stream type frame is declared lost or when the limit is
updated, with care taken to prevent the frame from being sent too often.
* Blocked signals are carried in DATA_BLOCKED, STREAM_DATA_BLOCKED, and
STREAMS_BLOCKED frames. DATA_BLOCKED frames have connection scope,
STREAM_DATA_BLOCKED frames have stream scope, and STREAMS_BLOCKED frames are
scoped to a specific stream type. New frames are sent if packets containing
the most recent frame for a scope is lost, but only while the endpoint is
blocked on the corresponding limit. These frames always include the limit that
is causing blocking at the time that they are transmitted.
* A liveness or path validation check using PATH_CHALLENGE frames is sent
periodically until a matching PATH_RESPONSE frame is received or until there
is no remaining need for liveness or path validation checking. PATH_CHALLENGE
frames include a different payload each time they are sent.
* Responses to path validation using PATH_RESPONSE frames are sent just once.
The peer is expected to send more PATH_CHALLENGE frames as necessary to evoke
additional PATH_RESPONSE frames.
* New connection IDs are sent in NEW_CONNECTION_ID frames and retransmitted if
the packet containing them is lost. Retransmissions of this frame carry the
same sequence number value. Likewise, retired connection IDs are sent in
RETIRE_CONNECTION_ID frames and retransmitted if the packet containing them is
lost.
* 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.
Endpoints SHOULD prioritize retransmission of data over sending new data, unless
priorities specified by the application indicate otherwise (see
{{stream-prioritization}}).
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 receiver MUST accept packets containing
an outdated frame, such as a MAX_DATA frame carrying a smaller maximum data than
one found in an older packet.
Upon detecting losses, a sender MUST take appropriate congestion control action.
The details of loss detection and congestion control are described in
{{QUIC-RECOVERY}}.
## Explicit Congestion Notification {#ecn}
QUIC endpoints can use Explicit Congestion Notification (ECN) {{!RFC3168}} to
detect and respond to network congestion. ECN allows a network node to indicate
congestion in the network by setting a codepoint in the IP header of a packet
instead of dropping it. Endpoints react to congestion by reducing their sending
rate in response, as described in {{QUIC-RECOVERY}}.
To use ECN, QUIC endpoints first determine whether a path supports ECN marking
and the peer is able to access the ECN codepoint in the IP header. A network
path does not support ECN if ECN marked packets get dropped or ECN markings are
rewritten on the path. An endpoint validates the use of ECN on the path, both
during connection establishment and when migrating to a new path
({{migration}}).
### ECN Counts
On receiving a QUIC packet with an ECT or CE codepoint, an ECN-enabled endpoint
that can access the ECN codepoints from the enclosing IP packet increases the
corresponding ECT(0), ECT(1), or CE count, and includes these counts in
subsequent ACK frames (see {{generating-acks}} and {{frame-ack}}). Note
that this requires being able to read the ECN codepoints from the enclosing IP
packet, which is not possible on all platforms.
A packet detected by a receiver as a duplicate does not affect the receiver's
local ECN codepoint counts; see ({{security-ecn}}) for relevant security
concerns.
If an endpoint receives a QUIC packet without an ECT or CE codepoint in the IP
packet header, it responds per {{generating-acks}} with an ACK frame without
increasing any ECN counts. If an endpoint does not implement ECN
support or does not have access to received ECN codepoints, it does not increase
ECN counts.
Coalesced packets (see {{packet-coalesce}}) mean that several packets can share
the same IP header. The ECN counter for the ECN codepoint received in the
associated IP header are incremented once for each QUIC packet, not per
enclosing IP packet or UDP datagram.
Each packet number space maintains separate acknowledgement state and separate
ECN counts. For example, if one each of an Initial, 0-RTT, Handshake, and 1-RTT
QUIC packet are coalesced, the corresponding counts for the Initial and
Handshake packet number space will be incremented by one and the counts for the
1-RTT packet number space will be increased by two.
### ECN Validation {#ecn-validation}
It is possible for faulty network devices to corrupt or erroneously drop packets
with ECN markings. To provide robust connectivity in the presence of such
devices, each endpoint independently validates ECN counts and disables ECN if
errors are detected.
Endpoints validate ECN for packets sent on each network path independently. An
endpoint thus validates ECN on new connection establishment, when switching to a
new server preferred address, and on active connection migration to a new path.
Even if an endpoint does not use ECN markings on packets it transmits, the
endpoint MUST provide feedback about ECN markings received from the peer if they
are accessible. Failing to report ECN counts will cause the peer to disable ECN
marking.
#### Sending ECN Markings
To start ECN validation, an endpoint SHOULD do the following when sending
packets on a new path to a peer:
* Set the ECT(0) codepoint in the IP header of early outgoing packets sent on a
new path to the peer {{!RFC8311}}.
* If all packets that were sent with the ECT(0) codepoint are eventually deemed
lost {{QUIC-RECOVERY}}, validation is deemed to have failed.
To reduce the chances of misinterpreting congestive loss as packets dropped by a
faulty network element, an endpoint could set the ECT(0) codepoint in the first
ten outgoing packets on a path, or for a period of three RTTs, whichever occurs
first.
Implementations MAY experiment with and use other strategies for use of ECN.
Other methods of probing paths for ECN support are possible, as are different
marking strategies. Implementations can also use the ECT(1) codepoint, as
specified in {{?RFC8311}}.
#### Receiving ACK Frames
An endpoint that sets ECT(0) or ECT(1) codepoints on packets it transmits MUST
use the following steps on receiving an ACK frame to validate ECN.
* If this ACK frame newly acknowledges a packet that the endpoint sent with
either ECT(0) or ECT(1) codepoints set, and if no ECN feedback is present in
the ACK frame, validation fails. This step protects against both a network
element that zeroes out ECN bits and a peer that is unable to access ECN
markings, since the peer could respond without ECN feedback in these cases.
* For validation to succeed, the total increase in ECT(0), ECT(1), and CE counts
MUST be no smaller than the total number of QUIC packets sent with an ECT
codepoint that are newly acknowledged in this ACK frame. This step detects
any network remarking from ECT(0), ECT(1), or CE codepoints to Not-ECT.
* Any increase in either ECT(0) or ECT(1) counts, plus any increase in the CE
count, MUST be no smaller than the number of packets sent with the
corresponding ECT codepoint that are newly acknowledged in this ACK frame.
This step detects any erroneous network remarking from ECT(0) to ECT(1) (or
vice versa).
Processing ECN counts out of order can result in validation failure. An
endpoint SHOULD NOT perform this validation if this ACK frame does not advance
the largest packet number acknowledged in this connection.
An endpoint could miss acknowledgements for a packet when ACK frames are lost.
It is therefore possible for the total increase in ECT(0), ECT(1), and CE counts
to be greater than the number of packets acknowledged in an ACK frame. When
this happens, and if validation succeeds, the local reference counts MUST be
increased to match the counts in the ACK frame.
#### Validation Outcomes
If validation fails, then the endpoint stops sending ECN markings in subsequent
IP packets with the expectation that either the network path or the peer does
not support ECN.
Upon successful validation, an endpoint can continue to set ECT codepoints in
subsequent packets with the expectation that the path is ECN-capable. Network
routing and path elements can change mid-connection however; an endpoint MUST
disable ECN if validation fails at any point in the connection.
Even if validation fails, an endpoint MAY revalidate ECN on the same path at any
later time in the connection.
# Packet Size {#packet-size}
The QUIC packet size includes the QUIC header and protected payload, but not the
UDP or IP header.
Clients MUST ensure they send the first Initial packet in a single IP packet.
Similarly, the first Initial packet sent after receiving a Retry packet MUST be
sent in a single IP packet.
The payload of a UDP datagram carrying the first Initial packet MUST be expanded
to at least 1200 bytes, by adding PADDING frames to the Initial packet and/or by
coalescing the Initial packet (see {{packet-coalesce}}). Sending a UDP datagram
of this size ensures that the network path supports a reasonable Maximum
Transmission Unit (MTU), and helps reduce the amplitude of amplification attacks
caused by server responses toward an unverified client address; see
{{address-validation}}.
The datagram containing the first Initial packet from a client MAY exceed 1200
bytes if the client believes that the Path Maximum Transmission Unit (PMTU)
supports the size that it chooses.
A server MAY send a CONNECTION_CLOSE frame with error code PROTOCOL_VIOLATION in
response to the first Initial packet it receives from a client if the UDP
datagram is smaller than 1200 bytes. It MUST NOT send any other frame type in
response, or otherwise behave as if any part of the offending packet was
processed as valid.
The server MUST also limit the number of bytes it sends before validating the
address of the client; see {{address-validation}}.
## Path Maximum Transmission Unit (PMTU)
The PMTU is the maximum size of the entire IP packet including the IP header,
UDP header, and UDP payload. The UDP payload includes the QUIC packet header,
protected payload, and any authentication fields. The PMTU can depend upon the
current path characteristics. Therefore, the current largest UDP payload an
implementation will send is referred to as the QUIC maximum packet size.
QUIC depends on a PMTU of at least 1280 bytes. This is the IPv6 minimum size
{{?RFC8200}} and is also supported by most modern IPv4 networks. All QUIC
packets (except for PMTU probe packets) SHOULD be sized to fit within the
maximum packet size to avoid the packet being fragmented or dropped
{{?RFC8085}}.
An endpoint SHOULD use Datagram Packetization Layer PMTU Discovery
({{!DPLPMTUD=I-D.ietf-tsvwg-datagram-plpmtud}}) or implement Path MTU Discovery
(PMTUD) {{!RFC1191}} {{!RFC8201}} to determine whether the path to a destination
will support a desired message size without fragmentation.
In the absence of these mechanisms, QUIC endpoints SHOULD NOT send IP packets
larger than 1280 bytes. Assuming the minimum IP header size, this results in a
QUIC maximum packet size of 1232 bytes for IPv6 and 1252 bytes for IPv4. A QUIC
implementation MAY be more conservative in computing the QUIC maximum packet
size to allow for unknown tunnel overheads or IP header options/extensions.
Each pair of local and remote addresses could have a different PMTU. QUIC
implementations that implement any kind of PMTU discovery therefore SHOULD
maintain a maximum packet size for each combination of local and remote IP
addresses.
If a QUIC endpoint determines that the PMTU between any pair of local and remote
IP addresses has fallen below the size needed to support the smallest allowed
maximum packet size, it MUST immediately cease sending QUIC packets, except for
PMTU probe packets, on the affected path. An endpoint MAY terminate the
connection if an alternative path cannot be found.
## ICMP Packet Too Big Messages {#icmp-pmtud}
PMTU discovery {{!RFC1191}} {{!RFC8201}} relies on reception of ICMP messages
(e.g., IPv6 Packet Too Big messages) that indicate when a packet is dropped
because it is larger than the local router MTU. DPLPMTUD can also optionally use
these messages. This use of ICMP messages is potentially vulnerable to off-path
attacks that successfully guess the addresses used on the path and reduce the
PMTU to a bandwidth-inefficient value.
An endpoint MUST ignore an ICMP message that claims the PMTU has decreased below
1280 bytes.
The requirements for generating ICMP ({{?RFC1812}}, {{?RFC4443}}) state that the
quoted packet should contain as much of the original packet as possible without
exceeding the minimum MTU for the IP version. The size of the quoted packet can
actually be smaller, or the information unintelligible, as described in Section
1.1 of {{!DPLPMTUD}}.
QUIC endpoints SHOULD validate ICMP messages to protect from off-path injection
as specified in {{!RFC8201}} and Section 5.2 of {{!RFC8085}}. This validation
SHOULD use the quoted packet supplied in the payload of an ICMP message to
associate the message with a corresponding transport connection {{!DPLPMTUD}}.
ICMP message validation MUST include matching IP addresses and UDP ports
{{!RFC8085}} and, when possible, connection IDs to an active QUIC session.
Further validation can also be provided:
* An IPv4 endpoint could set the Don't Fragment (DF) bit on a small proportion
of packets, so that most invalid ICMP messages arrive when there are no DF
packets outstanding, and can therefore be identified as spurious.
* An endpoint could store additional information from the IP or UDP headers to
use for validation (for example, the IP ID or UDP checksum).
The endpoint SHOULD ignore all ICMP messages that fail validation.
An endpoint MUST NOT increase PMTU based on ICMP messages. Any reduction in the
QUIC maximum packet size MAY be provisional until QUIC's loss detection
algorithm determines that the quoted packet has actually been lost.
## Datagram Packetization Layer PMTU Discovery
Section 6.4 of {{!DPLPMTUD}} provides considerations for implementing Datagram
Packetization Layer PMTUD (DPLPMTUD) with QUIC.
When implementing the algorithm in Section 5.3 of {{!DPLPMTUD}}, the initial
value of BASE_PMTU SHOULD be consistent with the minimum QUIC packet size (1232
bytes for IPv6 and 1252 bytes for IPv4).
PING and PADDING frames can be used to generate PMTU probe packets. These frames
might not be retransmitted if a probe packet containing them is lost. However,
these frames do consume congestion window, which could delay the transmission of
subsequent application data.
A PING frame can be included in a PMTU probe to ensure that a valid probe is
acknowledged.
The considerations for processing ICMP messages in the previous section also
apply if these messages are used by DPLPMTUD.
### PMTU Probes Containing Source Connection ID {#pmtu-probes-src-cid}
Endpoints that rely on the destination connection ID for routing QUIC packets
are likely to require that the connection ID be included in PMTU probe packets
to route any resulting ICMP messages ({{icmp-pmtud}}) back to the correct
endpoint. However, only long header packets ({{long-header}}) contain source
connection IDs, and long header packets are not decrypted or acknowledged by
the peer once the handshake is complete. One way to construct a PMTU probe is
to coalesce (see {{packet-coalesce}}) a Handshake packet ({{packet-handshake}})
with a short header packet in a single UDP datagram. If the UDP datagram
reaches the endpoint, the Handshake packet will be ignored, but the short header
packet will be acknowledged. If the UDP datagram elicits an ICMP message, that
message will likely contain the source connection ID within the quoted portion
of the UDP datagram.
# Versions {#versions}
QUIC versions are identified using a 32-bit unsigned number.
The version 0x00000000 is reserved to represent version negotiation. This
version of the specification is identified by the number 0x00000001.
Other versions of QUIC might have different properties to this version. The
properties of QUIC that are guaranteed to be consistent across all versions of
the protocol are described in {{QUIC-INVARIANTS}}.
Version 0x00000001 of QUIC uses TLS as a cryptographic handshake protocol, as
described in {{QUIC-TLS}}.
Versions with the most significant 16 bits of the version number cleared are
reserved for use in future IETF consensus documents.
Versions that follow the pattern 0x?a?a?a?a are reserved for use in forcing
version negotiation to be exercised. That is, any version number where the low
four bits of all bytes is 1010 (in binary). A client or server MAY advertise
support for any of these reserved versions.
Reserved version numbers will probably never represent a real protocol; a client
MAY use one of these version numbers with the expectation that the server will
initiate version negotiation; a server MAY advertise support for one of these
versions and can expect that clients ignore the value.
\[\[RFC editor: please remove the remainder of this section before
publication.]]
The version number for the final version of this specification (0x00000001), is
reserved for the version of the protocol that is published as an RFC.
Version numbers used to identify IETF drafts are created by adding the draft
number to 0xff000000. For example, draft-ietf-quic-transport-13 would be
identified as 0xff00000D.
Implementors are encouraged to register version numbers of QUIC that they are
using for private experimentation on the GitHub wiki at
\<https://github.com/quicwg/base-drafts/wiki/QUIC-Versions\>.
# Variable-Length Integer Encoding {#integer-encoding}
QUIC packets and frames commonly use a variable-length encoding for non-negative
integer values. This encoding ensures that smaller integer values need fewer
bytes to encode.
The QUIC variable-length integer encoding reserves the two most significant bits
of the first byte to encode the base 2 logarithm of the integer encoding length
in bytes. The integer value is encoded on the remaining bits, in network byte
order.
This means that integers are encoded on 1, 2, 4, or 8 bytes and can encode 6,
14, 30, or 62 bit values respectively. {{integer-summary}} summarizes the
encoding properties.
| 2Bit | Length | Usable Bits | Range |
|:-----|:-------|:------------|:----------------------|
| 00 | 1 | 6 | 0-63 |
| 01 | 2 | 14 | 0-16383 |
| 10 | 4 | 30 | 0-1073741823 |
| 11 | 8 | 62 | 0-4611686018427387903 |
{: #integer-summary title="Summary of Integer Encodings"}
For example, the eight byte sequence c2 19 7c 5e ff 14 e8 8c (in hexadecimal)
decodes to the decimal value 151288809941952652; the four byte sequence 9d 7f 3e
7d decodes to 494878333; the two byte sequence 7b bd decodes to 15293; and the
single byte 25 decodes to 37 (as does the two byte sequence 40 25).
Error codes ({{error-codes}}) and versions ({{versions}}) are described using
integers, but do not use this encoding.
# Packet Formats {#packet-formats}
All numeric values are encoded in network byte order (that is, big-endian) and
all field sizes are in bits. Hexadecimal notation is used for describing the
value of fields.
## Packet Number Encoding and Decoding {#packet-encoding}
Packet numbers are integers in the range 0 to 2^62-1 ({{packet-numbers}}). When
present in long or short packet headers, they are encoded in 1 to 4 bytes. The
number of bits required to represent the packet number is reduced by including
the least significant bits of the packet number.
The encoded packet number is protected as described in Section 5.4 of
{{QUIC-TLS}}.
The sender MUST use a packet number size able to represent more than twice as
large a range than the difference between the largest acknowledged packet and
packet number being sent. A peer receiving the packet will then correctly
decode the packet number, unless the packet is delayed in transit such that it
arrives after many higher-numbered packets have been received. An endpoint
SHOULD use a large enough packet number encoding to allow the packet number to
be recovered even if the packet arrives after packets that are sent afterwards.
As a result, the size of the packet number encoding is at least one bit more
than the base-2 logarithm of the number of contiguous unacknowledged packet
numbers, including the new packet.
For example, if an endpoint has received an acknowledgment for packet 0xabe8bc,
sending a packet with a number of 0xac5c02 requires a packet number encoding
with 16 bits or more; whereas the 24-bit packet number encoding is needed to
send a packet with a number of 0xace8fe.
At a receiver, protection of the packet number is removed prior to recovering
the full packet number. The full packet number is then reconstructed based on
the number of significant bits present, the value of those bits, and the largest
packet number received on a successfully authenticated packet. Recovering the
full packet number is necessary to successfully remove packet protection.
Once header protection is removed, the packet number is decoded by finding the
packet number value that is closest to the next expected packet. The next
expected packet is the highest received packet number plus one. For example, if
the highest successfully authenticated packet had a packet number of 0xa82f30ea,
then a packet containing a 16-bit value of 0x9b32 will be decoded as 0xa82f9b32.
Example pseudo-code for packet number decoding can be found in
{{sample-packet-number-decoding}}.
## Long Header Packets {#long-header}
~~~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+
|1|1|T T|X X X X|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Version (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| DCID Len (8) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Connection ID (0..160) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SCID Len (8) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Source Connection ID (0..160) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~~~
{: #fig-long-header title="Long Header Packet Format"}
Long headers are used for packets that are sent prior to the establishment
of 1-RTT keys. Once both conditions are
met, a sender switches to sending packets using the short header
({{short-header}}). The long form allows for special packets - such as the
Version Negotiation packet - to be represented in this uniform fixed-length
packet format. Packets that use the long header contain the following fields:
Header Form:
: The most significant bit (0x80) of byte 0 (the first byte) is set to 1 for
long headers.
Fixed Bit:
: The next bit (0x40) of byte 0 is set to 1. Packets containing a zero value
for this bit are not valid packets in this version and MUST be discarded.
Long Packet Type (T):
: The next two bits (those with a mask of 0x30) of byte 0 contain a packet type.
Packet types are listed in {{long-packet-types}}.
Type-Specific Bits (X):
: The lower four bits (those with a mask of 0x0f) of byte 0 are type-specific.
Version:
: The QUIC Version is a 32-bit field that follows the first byte. This field
indicates which version of QUIC is in use and determines how the rest of the
protocol fields are interpreted.
DCID Len:
: The byte following the version contains the length in bytes of the Destination
Connection ID field that follows it. This length is encoded as an 8-bit
unsigned integer. In QUIC version 1, this value MUST NOT exceed 20.
Endpoints that receive a version 1 long header with a value larger than
20 MUST drop the packet. Servers SHOULD be able to read longer connection IDs
from other QUIC versions in order to properly form a version negotiation
packet.
Destination Connection ID:
: The Destination Connection ID field follows the DCID Len and is between 0 and
20 bytes in length. {{negotiating-connection-ids}} describes the use of this
field in more detail.
SCID Len:
: The byte following the Destination Connection ID contains the length in bytes
of the Source Connection ID field that follows it. This length is encoded as
a 8-bit unsigned integer. In QUIC version 1, this value MUST NOT exceed 20
bytes. Endpoints that receive a version 1 long header with a value larger than
20 MUST drop the packet. Servers SHOULD be able to read longer connection IDs
from other QUIC versions in order to properly form a version negotiation
packet.
Source Connection ID:
: The Source Connection ID field follows the SCID Len and is between 0 and 20
bytes in length. {{negotiating-connection-ids}} describes the use of this
field in more detail.
In this version of QUIC, the following packet types with the long header are
defined:
| Type | Name | Section |
|-----:|:------------------------------|:----------------------------|
| 0x0 | Initial | {{packet-initial}} |
| 0x1 | 0-RTT | {{packet-0rtt}} |
| 0x2 | Handshake | {{packet-handshake}} |
| 0x3 | Retry | {{packet-retry}} |
{: #long-packet-types title="Long Header Packet Types"}
The header form bit, connection ID lengths byte, Destination and Source
Connection ID fields, and Version fields of a long header packet are
version-independent. The other fields in the first byte are version-specific.
See {{QUIC-INVARIANTS}} for details on how packets from different versions of
QUIC are interpreted.
The interpretation of the fields and the payload are specific to a version and
packet type. While type-specific semantics for this version are described in
the following sections, several long-header packets in this version of QUIC
contain these additional fields:
Reserved Bits (R):
: Two bits (those with a mask of 0x0c) of byte 0 are reserved across multiple
packet types. These bits are protected using header protection (see Section
5.4 of {{QUIC-TLS}}). The value included prior to protection MUST be set to 0.
An endpoint MUST treat receipt of a packet that has a non-zero value for these
bits, after removing both packet and header protection, as a connection error
of type PROTOCOL_VIOLATION. Discarding such a packet after only removing
header protection can expose the endpoint to attacks (see Section 9.3 of
{{QUIC-TLS}}).
Packet Number Length (P):
: In packet types which contain a Packet Number field, the least significant two
bits (those with a mask of 0x03) of byte 0 contain the length of the packet
number, encoded as an unsigned, two-bit integer that is one less than the
length of the packet number field in bytes. That is, the length of the packet
number field is the value of this field, plus one. These bits are protected
using header protection (see Section 5.4 of {{QUIC-TLS}}).
Length:
: The length of the remainder of the packet (that is, the Packet Number and
Payload fields) in bytes, encoded as a variable-length integer
({{integer-encoding}}).
Packet Number:
: The packet number field is 1 to 4 bytes long. The packet number has
confidentiality protection separate from packet protection, as described in
Section 5.4 of {{QUIC-TLS}}. The length of the packet number field is encoded
in the Packet Number Length bits of byte 0 (see above).
### Version Negotiation Packet {#packet-version}
A Version Negotiation packet is inherently not version-specific. Upon receipt by
a client, it will be identified as a Version Negotiation packet based on the
Version field having a value of 0.
The Version Negotiation packet is a response to a client packet that contains a
version that is not supported by the server, and is only sent by servers.
The layout of a Version Negotiation packet is:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+
|1| Unused (7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Version (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| DCID Len (8) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Connection ID (0..2040) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SCID Len (8) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Source Connection ID (0..2040) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Supported Version 1 (32) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Supported Version 2 (32)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Supported Version N (32)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #version-negotiation-format title="Version Negotiation Packet"}
The value in the Unused field is selected randomly by the server. Clients MUST
ignore the value of this field. Servers SHOULD set the most significant bit of
this field (0x40) to 1 so that Version Negotiation packets appear to have the
Fixed Bit field.
The Version field of a Version Negotiation packet MUST be set to 0x00000000.
The server MUST include the value from the Source Connection ID field of the
packet it receives in the Destination Connection ID field. The value for Source
Connection ID MUST be copied from the Destination Connection ID of the received
packet, which is initially randomly selected by a client. Echoing both
connection IDs gives clients some assurance that the server received the packet
and that the Version Negotiation packet was not generated by an off-path
attacker.
As future versions of QUIC may support Connection IDs larger than the version 1
limit, Version Negotiation packets could carry Connection IDs that are longer
than 20 bytes.
The remainder of the Version Negotiation packet is a list of 32-bit versions
which the server supports.
A Version Negotiation packet cannot be explicitly acknowledged in an ACK frame
by a client. Receiving another Initial packet implicitly acknowledges a Version
Negotiation packet.
The Version Negotiation packet does not include the Packet Number and Length
fields present in other packets that use the long header form. Consequently,
a Version Negotiation packet consumes an entire UDP datagram.
A server MUST NOT send more than one Version Negotiation packet in response to a
single UDP datagram.
See {{version-negotiation}} for a description of the version negotiation
process.
### Initial Packet {#packet-initial}
An Initial packet uses long headers with a type value of 0x0. It carries the
first CRYPTO frames sent by the client and server to perform key exchange, and
carries ACKs in either direction.
~~~
+-+-+-+-+-+-+-+-+
|1|1| 0 |R R|P P|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Version (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| DCID Len (8) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Connection ID (0..160) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SCID Len (8) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Source Connection ID (0..160) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Token Length (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Token (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Length (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Packet Number (8/16/24/32) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Payload (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #initial-format title="Initial Packet"}
The Initial packet contains a long header as well as the Length and Packet
Number fields. The first byte contains the Reserved and Packet Number Length
bits. Between the SCID and Length fields, there are two additional
field specific to the Initial packet.
Token Length:
: A variable-length integer specifying the length of the Token field, in bytes.
This value is zero if no token is present. Initial packets sent by the server
MUST set the Token Length field to zero; clients that receive an Initial
packet with a non-zero Token Length field MUST either discard the packet or
generate a connection error of type PROTOCOL_VIOLATION.
Token:
: The value of the token that was previously provided in a Retry packet or
NEW_TOKEN frame.
Payload:
: The payload of the packet.
In order to prevent tampering by version-unaware middleboxes, Initial packets
are protected with connection- and version-specific keys (Initial keys) as
described in {{QUIC-TLS}}. This protection does not provide confidentiality or
integrity against on-path attackers, but provides some level of protection
against off-path attackers.
The client and server use the Initial packet type for any packet that contains
an initial cryptographic handshake message. This includes all cases where a new
packet containing the initial cryptographic message needs to be created, such as
the packets sent after receiving a Retry packet ({{packet-retry}}).
A server sends its first Initial packet in response to a client Initial. A
server may send multiple Initial packets. The cryptographic key exchange could
require multiple round trips or retransmissions of this data.
The payload of an Initial packet includes a CRYPTO frame (or frames) containing
a cryptographic handshake message, ACK frames, or both. PADDING and
CONNECTION_CLOSE frames are also permitted. An endpoint that receives an
Initial packet containing other frames can either discard the packet as spurious
or treat it as a connection error.
The first packet sent by a client always includes a CRYPTO frame that contains
the entirety of the first cryptographic handshake message. This packet, and the
cryptographic handshake message, MUST fit in a single UDP datagram (see
{{handshake}}). The first CRYPTO frame sent always begins at an offset of 0
(see {{handshake}}).
Note that if the server sends a HelloRetryRequest, the client will send a second
Initial packet. This Initial packet will continue the cryptographic handshake
and will contain a CRYPTO frame with an offset matching the size of the CRYPTO
frame sent in the first Initial packet. Cryptographic handshake messages
subsequent to the first do not need to fit within a single UDP datagram.
#### Abandoning Initial Packets {#discard-initial}
A client stops both sending and processing Initial packets when it sends its
first Handshake packet. A server stops sending and processing Initial packets
when it receives its first Handshake packet. Though packets might still be in
flight or awaiting acknowledgment, no further Initial packets need to be
exchanged beyond this point. Initial packet protection keys are discarded (see
Section 4.9.1 of {{QUIC-TLS}}) along with any loss recovery and congestion
control state (see Section 6.5 of {{QUIC-RECOVERY}}).
Any data in CRYPTO frames is discarded - and no longer retransmitted - when
Initial keys are discarded.
### 0-RTT {#packet-0rtt}
A 0-RTT packet uses long headers with a type value of 0x1, followed by the
Length and Packet Number fields. The first byte contains the Reserved and Packet
Number Length bits. It is used to carry "early" data from the client to the
server as part of the first flight, prior to handshake completion. As part of
the TLS handshake, the server can accept or reject this early data.
See Section 2.3 of {{!TLS13}} for a discussion of 0-RTT data and its
limitations.
~~~
+-+-+-+-+-+-+-+-+
|1|1| 1 |R R|P P|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Version (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| DCID Len (8) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Connection ID (0..160) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SCID Len (8) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Source Connection ID (0..160) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Length (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Packet Number (8/16/24/32) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Payload (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #0rtt-format title="0-RTT Packet"}
Packet numbers for 0-RTT protected packets use the same space as 1-RTT protected
packets.
After a client receives a Retry packet, 0-RTT packets are likely to have been
lost or discarded by the server. A client SHOULD attempt to resend data in
0-RTT packets after it sends a new Initial packet.
A client MUST NOT reset the packet number it uses for 0-RTT packets, since the
keys used to protect 0-RTT packets will not change as a result of responding to
a Retry packet. Sending packets with the same packet number in that case is
likely to compromise the packet protection for all 0-RTT packets because the
same key and nonce could be used to protect different content.
A client only receives acknowledgments for its 0-RTT packets once the handshake
is complete. Consequently, a server might expect 0-RTT packets to start with a
packet number of 0. Therefore, in determining the length of the packet number
encoding for 0-RTT packets, a client MUST assume that all packets up to the
current packet number are in flight, starting from a packet number of 0. Thus,
0-RTT packets could need to use a longer packet number encoding.
A client MUST NOT send 0-RTT packets once it starts processing 1-RTT packets
from the server. This means that 0-RTT packets cannot contain any response to
frames from 1-RTT packets. For instance, a client cannot send an ACK frame in a
0-RTT packet, because that can only acknowledge a 1-RTT packet. An
acknowledgment for a 1-RTT packet MUST be carried in a 1-RTT packet.
A server SHOULD treat a violation of remembered limits as a connection error of
an appropriate type (for instance, a FLOW_CONTROL_ERROR for exceeding stream
data limits).
### Handshake Packet {#packet-handshake}
A Handshake packet uses long headers with a type value of 0x2, followed by the
Length and Packet Number fields. The first byte contains the Reserved and
Packet Number Length bits. It is used to carry acknowledgments and
cryptographic handshake messages from the server and client.
~~~
+-+-+-+-+-+-+-+-+
|1|1| 2 |R R|P P|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Version (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| DCID Len (8) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Connection ID (0..160) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SCID Len (8) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Source Connection ID (0..160) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Length (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Packet Number (8/16/24/32) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Payload (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #handshake-format title="Handshake Protected Packet"}
Once a client has received a Handshake packet from a server, it uses Handshake
packets to send subsequent cryptographic handshake messages and acknowledgments
to the server.
The Destination Connection ID field in a Handshake packet contains a connection
ID that is chosen by the recipient of the packet; the Source Connection ID
includes the connection ID that the sender of the packet wishes to use (see
{{negotiating-connection-ids}}).
Handshake packets are their own packet number space, and thus the first
Handshake packet sent by a server contains a packet number of 0.
The payload of this packet contains CRYPTO frames and could contain PADDING, or
ACK frames. Handshake packets MAY contain CONNECTION_CLOSE frames. Endpoints
MUST treat receipt of Handshake packets with other frames as a connection error.
Like Initial packets (see {{discard-initial}}), data in CRYPTO frames at the
Handshake encryption level is discarded - and no longer retransmitted - when
Handshake protection keys are discarded.
### Retry Packet {#packet-retry}
A Retry packet uses a long packet header with a type value of 0x3. It carries
an address validation token created by the server. It is used by a server that
wishes to perform a retry (see {{validate-handshake}}).
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+
|1|1| 3 | Unused|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Version (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| DCID Len (8) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Connection ID (0..160) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SCID Len (8) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Source Connection ID (0..160) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ODCID Len (8) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Original Destination Connection ID (0..160) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Retry Token (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #retry-format title="Retry Packet"}
A Retry packet (shown in {{retry-format}}) does not contain any protected
fields. The value in the Unused field is selected randomly by the server. In
addition to the long header, it contains these additional fields:
ODCID Len:
: The ODCID Len contains the length in bytes of the Original Destination
Connection ID field that follows it. This length is encoded as a 8-bit
unsigned integer. In QUIC version 1, this value MUST NOT exceed 20 bytes.
Clients that receive a version 1 Retry Packet with a value larger than 20 MUST
drop the packet.
Original Destination Connection ID:
: The Original Destination Connection ID contains the value of the Destination
Connection ID from the Initial packet that this Retry is in response to. The
length of this field is given in ODCID Len.
Retry Token:
: An opaque token that the server can use to validate the client's address.
<!-- Break this stuff up a little, maybe into "Sending Retry" and "Processing
Retry" sections. -->
The server populates the Destination Connection ID with the connection ID that
the client included in the Source Connection ID of the Initial packet.
The server includes a connection ID of its choice in the Source Connection ID
field. This value MUST not be equal to the Destination Connection ID field of
the packet sent by the client. The client MUST use this connection ID in the
Destination Connection ID of subsequent packets that it sends.
A server MAY send Retry packets in response to Initial and 0-RTT packets. A
server can either discard or buffer 0-RTT packets that it receives. A server
can send multiple Retry packets as it receives Initial or 0-RTT packets. A
server MUST NOT send more than one Retry packet in response to a single UDP
datagram.
A client MUST accept and process at most one Retry packet for each connection
attempt. After the client has received and processed an Initial or Retry packet
from the server, it MUST discard any subsequent Retry packets that it receives.
Clients MUST discard Retry packets that contain an Original Destination
Connection ID field that does not match the Destination Connection ID from its
Initial packet. This prevents an off-path attacker from injecting a Retry
packet.
The client responds to a Retry packet with an Initial packet that includes the
provided Retry Token to continue connection establishment.
A client sets the Destination Connection ID field of this Initial packet to the
value from the Source Connection ID in the Retry packet. Changing Destination
Connection ID also results in a change to the keys used to protect the Initial
packet. It also sets the Token field to the token provided in the Retry. The
client MUST NOT change the Source Connection ID because the server could include
the connection ID as part of its token validation logic (see
{{token-integrity}}).
The next Initial packet from the client uses the connection ID and token values
from the Retry packet (see {{negotiating-connection-ids}}). Aside from this,
the Initial packet sent by the client is subject to the same restrictions as the
first Initial packet. A client MUST use the same cryptographic handshake
message it includes in this packet. A server MAY treat a packet that
contains a different cryptographic handshake message as a connection error or
discard it.
A client MAY attempt 0-RTT after receiving a Retry packet by sending 0-RTT
packets to the connection ID provided by the server. A client MUST NOT change
the cryptographic handshake message it sends in response to receiving a Retry.
A client MUST NOT reset the packet number for any packet number space after
processing a Retry packet; {{packet-0rtt}} contains more information on this.
A server acknowledges the use of a Retry packet for a connection using the
original_connection_id transport parameter (see
{{transport-parameter-definitions}}). If the server sends a Retry packet, it
MUST include the value of the Original Destination Connection ID field of the
Retry packet (that is, the Destination Connection ID field from the client's
first Initial packet) in the transport parameter.
If the client received and processed a Retry packet, it MUST validate that the
original_connection_id transport parameter is present and correct; otherwise, it
MUST validate that the transport parameter is absent. A client MUST treat a
failed validation as a connection error of type TRANSPORT_PARAMETER_ERROR.
A Retry packet does not include a packet number and cannot be explicitly
acknowledged by a client.
## Short Header Packets {#short-header}
This version of QUIC defines a single packet type which uses the
short packet header.
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+
|0|1|S|R|R|K|P P|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Connection ID (0..160) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Packet Number (8/16/24/32) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Protected Payload (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~~~
{: #fig-short-header title="Short Header Packet Format"}
The short header can be used after the version and 1-RTT keys are negotiated.
Packets that use the short header contain the following fields:
Header Form:
: The most significant bit (0x80) of byte 0 is set to 0 for the short header.
Fixed Bit:
: The next bit (0x40) of byte 0 is set to 1. Packets containing a zero value
for this bit are not valid packets in this version and MUST be discarded.
Spin Bit (S):
: The third most significant bit (0x20) of byte 0 is the latency spin bit, set
as described in {{spin-bit}}.
Reserved Bits (R):
: The next two bits (those with a mask of 0x18) of byte 0 are reserved. These
bits are protected using header protection (see Section 5.4 of
{{QUIC-TLS}}). The value included prior to protection MUST be set to 0. An
endpoint MUST treat receipt of a packet that has a non-zero value for these
bits, after removing both packet and header protection, as a connection error
of type PROTOCOL_VIOLATION. Discarding such a packet after only removing
header protection can expose the endpoint to attacks (see Section 9.3 of
{{QUIC-TLS}}).
Key Phase (K):
: The next bit (0x04) of byte 0 indicates the key phase, which allows a
recipient of a packet to identify the packet protection keys that are used to
protect the packet. See {{QUIC-TLS}} for details. This bit is protected
using header protection (see Section 5.4 of {{QUIC-TLS}}).
Packet Number Length (P):
: The least significant two bits (those with a mask of 0x03) of byte 0 contain
the length of the packet number, encoded as an unsigned, two-bit integer that
is one less than the length of the packet number field in bytes. That is, the
length of the packet number field is the value of this field, plus one. These
bits are protected using header protection (see Section 5.4 of {{QUIC-TLS}}).
Destination Connection ID:
: The Destination Connection ID is a connection ID that is chosen by the
intended recipient of the packet. See {{connection-id}} for more details.
Packet Number:
: The packet number field is 1 to 4 bytes long. The packet number has
confidentiality protection separate from packet protection, as described in
Section 5.4 of {{QUIC-TLS}}. The length of the packet number field is encoded
in Packet Number Length field. See {{packet-encoding}} for details.
Protected Payload:
: Packets with a short header always include a 1-RTT protected payload.
The header form bit and the connection ID field of a short header packet are
version-independent. The remaining fields are specific to the selected QUIC
version. See {{QUIC-INVARIANTS}} for details on how packets from different
versions of QUIC are interpreted.
### Latency Spin Bit {#spin-bit}
The latency spin bit enables passive latency monitoring from observation points
on the network path throughout the duration of a connection. The spin bit is
only present in the short packet header, since it is possible to measure the
initial RTT of a connection by observing the handshake. Therefore, the spin bit
is available after version negotiation and connection establishment are
completed. On-path measurement and use of the latency spin bit is further
discussed in {{?QUIC-MANAGEABILITY=I-D.ietf-quic-manageability}}.
The spin bit is an OPTIONAL feature of QUIC. A QUIC stack that chooses to
support the spin bit MUST implement it as specified in this section.
Each endpoint unilaterally decides if the spin bit is enabled or disabled for a
connection. Implementations MUST allow administrators of clients and servers
to disable the spin bit either globally or on a per-connection basis. Even when
the spin bit is not disabled by the administrator, implementations MUST disable
the spin bit for a given connection with a certain likelihood. The random
selection process SHOULD be designed such that on average the spin bit is
disabled for at least one eighth of network paths. The selection process
performed at the beginning of the connection SHOULD be applied for all paths
used by the connection.
When the spin bit is disabled, endpoints MAY set the spin bit to any value, and
MUST ignore any incoming value. It is RECOMMENDED that endpoints set the spin
bit to a random value either chosen independently for each packet or chosen
independently for each connection ID.
If the spin bit is enabled for the connection, the endpoint maintains a spin
value and sets the spin bit in the short header to the currently stored
value when a packet with a short header is sent out. The spin value is
initialized to 0 in the endpoint at connection start. Each endpoint also
remembers the highest packet number seen from its peer on the connection.
When a server receives a short header packet that increments the highest
packet number seen by the server from the client, it sets the spin value to be
equal to the spin bit in the received packet.
When a client receives a short header packet that increments the highest
packet number seen by the client from the server, it sets the spin value to the
inverse of the spin bit in the received packet.
An endpoint resets its spin value to zero when sending the first packet of a
given connection with a new connection ID. This reduces the risk that transient
spin bit state can be used to link flows across connection migration or ID
change.
With this mechanism, the server reflects the spin value received, while the
client 'spins' it after one RTT. On-path observers can measure the time
between two spin bit toggle events to estimate the end-to-end RTT of a
connection.
# Transport Parameter Encoding {#transport-parameter-encoding}
The format of the transport parameters is the TransportParameters struct from
{{figure-transport-parameters}}. This is described using the presentation
language from Section 3 of {{!TLS13=RFC8446}}.
~~~
enum {
original_connection_id(0),
idle_timeout(1),
stateless_reset_token(2),
max_packet_size(3),
initial_max_data(4),
initial_max_stream_data_bidi_local(5),
initial_max_stream_data_bidi_remote(6),
initial_max_stream_data_uni(7),
initial_max_streams_bidi(8),
initial_max_streams_uni(9),
ack_delay_exponent(10),
max_ack_delay(11),
disable_active_migration(12),
preferred_address(13),
active_connection_id_limit(14),
(65535)
} TransportParameterId;
struct {
TransportParameterId parameter;
opaque value<0..2^16-1>;
} TransportParameter;
TransportParameter TransportParameters<0..2^16-1>;
~~~
{: #figure-transport-parameters title="Definition of TransportParameters"}
The `extension_data` field of the quic_transport_parameters extension defined in
{{QUIC-TLS}} contains a TransportParameters value. TLS encoding rules are
therefore used to describe the encoding of transport parameters.
QUIC encodes transport parameters into a sequence of bytes, which are then
included in the cryptographic handshake.
## Reserved Transport Parameters {#transport-parameter-grease}
Transport parameters with an identifier of the form `31 * N + 27` for integer
values of N are reserved to exercise the requirement that unknown transport
parameters be ignored. These transport parameters have no semantics, and may
carry arbitrary values.
## Transport Parameter Definitions {#transport-parameter-definitions}
This section details the transport parameters defined in this document.
Many transport parameters listed here have integer values. Those transport
parameters that are identified as integers use a variable-length integer
encoding (see {{integer-encoding}}) and have a default value of 0 if the
transport parameter is absent, unless otherwise stated.
The following transport parameters are defined:
original_connection_id (0x0000):
: The value of the Destination Connection ID field from the first Initial packet
sent by the client. This transport parameter is only sent by a server. A
server MUST include the original_connection_id transport parameter if it sent
a Retry packet.
idle_timeout (0x0001):
: The idle timeout is a value in milliseconds that is encoded as an integer; see
({{idle-timeout}}). If this parameter is absent or zero then the idle
timeout is disabled.
stateless_reset_token (0x0002):
: A stateless reset token is used in verifying a stateless reset; see
{{stateless-reset}}. This parameter is a sequence of 16 bytes. This
transport parameter MUST NOT be sent by a client, but MAY be sent by a server.
A server that does not send this transport parameter cannot use stateless
reset ({{stateless-reset}}) for the connection ID negotiated during the
handshake.
max_packet_size (0x0003):
: The maximum packet size parameter is an integer value that limits the size of
packets that the endpoint is willing to receive. This indicates that packets
larger than this limit will be dropped. The default for this parameter is the
maximum permitted UDP payload of 65527. Values below 1200 are invalid. This
limit only applies to protected packets ({{packet-protected}}).
initial_max_data (0x0004):
: The initial maximum data parameter is an integer value that contains the
initial value for the maximum amount of data that can be sent on the
connection. This is equivalent to sending a MAX_DATA ({{frame-max-data}}) for
the connection immediately after completing the handshake.
initial_max_stream_data_bidi_local (0x0005):
: This parameter is an integer value specifying the initial flow control limit
for locally-initiated bidirectional streams. This limit applies to newly
created bidirectional streams opened by the endpoint that sends the transport
parameter. In client transport parameters, this applies to streams with an
identifier with the least significant two bits set to 0x0; in server transport
parameters, this applies to streams with the least significant two bits set to
0x1.
initial_max_stream_data_bidi_remote (0x0006):
: This parameter is an integer value specifying the initial flow control limit
for peer-initiated bidirectional streams. This limit applies to newly created
bidirectional streams opened by the endpoint that receives the transport
parameter. In client transport parameters, this applies to streams with an
identifier with the least significant two bits set to 0x1; in server transport
parameters, this applies to streams with the least significant two bits set to
0x0.
initial_max_stream_data_uni (0x0007):
: This parameter is an integer value specifying the initial flow control limit
for unidirectional streams. This limit applies to newly created
unidirectional streams opened by the endpoint that receives the transport
parameter. In client transport parameters, this applies to streams with an
identifier with the least significant two bits set to 0x3; in server transport
parameters, this applies to streams with the least significant two bits set to
0x2.
initial_max_streams_bidi (0x0008):
: The initial maximum bidirectional streams parameter is an integer value that
contains the initial maximum number of bidirectional streams the peer may
initiate. If this parameter is absent or zero, the peer cannot open
bidirectional streams until a MAX_STREAMS frame is sent. Setting this
parameter is equivalent to sending a MAX_STREAMS ({{frame-max-streams}}) of
the corresponding type with the same value.
initial_max_streams_uni (0x0009):
: The initial maximum unidirectional streams parameter is an integer value that
contains the initial maximum number of unidirectional streams the peer may
initiate. If this parameter is absent or zero, the peer cannot open
unidirectional streams until a MAX_STREAMS frame is sent. Setting this
parameter is equivalent to sending a MAX_STREAMS ({{frame-max-streams}}) of
the corresponding type with the same value.
ack_delay_exponent (0x000a):
: The ACK delay exponent is an integer value indicating an
exponent used to decode the ACK Delay field in the ACK frame ({{frame-ack}}).
If this value is absent, a default value of 3 is assumed (indicating a
multiplier of 8). Values above 20 are invalid.
max_ack_delay (0x000b):
: The maximum ACK delay is an integer value indicating the
maximum amount of time in milliseconds by which the endpoint will delay
sending acknowledgments. This value SHOULD include the receiver's expected
delays in alarms firing. For example, if a receiver sets a timer for 5ms
and alarms commonly fire up to 1ms late, then it should send a max_ack_delay
of 6ms. If this value is absent, a default of 25 milliseconds is assumed.
Values of 2^14 or greater are invalid.
disable_active_migration (0x000c):
: The disable active migration transport parameter is included if the endpoint
does not support active connection migration ({{migration}}). Peers of an
endpoint that sets this transport parameter MUST NOT send any packets,
including probing packets ({{probing}}), from a local address or port other
than that used to perform the handshake. This parameter is a zero-length
value.
preferred_address (0x000d):
: The server's preferred address is used to effect a change in server address at
the end of the handshake, as described in {{preferred-address}}. The format
of this transport parameter is the PreferredAddress struct shown in
{{fig-preferred-address}}. This transport parameter is only sent by a server.
Servers MAY choose to only send a preferred address of one address family by
sending an all-zero address and port (0.0.0.0:0 or ::.0) for the other family.
IP addresses are encoded in network byte order.
~~~
struct {
opaque ipv4Address[4];
uint16 ipv4Port;
opaque ipv6Address[16];
uint16 ipv6Port;
opaque connectionId<0..20>;
opaque statelessResetToken[16];
} PreferredAddress;
~~~
{: #fig-preferred-address title="Preferred Address format"}
active_connection_id_limit (0x000e):
: The maximum number of connection IDs from the peer that an endpoint is willing
to store. This value includes only connection IDs sent in NEW_CONNECTION_ID
frames. If this parameter is absent, a default of 0 is assumed.
If present, transport parameters that set initial flow control limits
(initial_max_stream_data_bidi_local, initial_max_stream_data_bidi_remote, and
initial_max_stream_data_uni) are equivalent to sending a MAX_STREAM_DATA frame
({{frame-max-stream-data}}) on every stream of the corresponding type
immediately after opening. If the transport parameter is absent, streams of
that type start with a flow control limit of 0.
A client MUST NOT include an original connection ID, a stateless reset token, or
a preferred address. A server MUST treat receipt of any of these transport
parameters as a connection error of type TRANSPORT_PARAMETER_ERROR.
# Frame Types and Formats {#frame-formats}
As described in {{frames}}, packets contain one or more frames. This section
describes the format and semantics of the core QUIC frame types.
## PADDING Frame {#frame-padding}
The PADDING frame (type=0x00) has no semantic value. PADDING frames can be used
to increase the size of a packet. Padding can be used to increase an initial
client packet to the minimum required size, or to provide protection against
traffic analysis for protected packets.
A PADDING frame has no content. That is, a PADDING frame consists of the single
byte that identifies the frame as a PADDING frame.
## PING Frame {#frame-ping}
Endpoints can use PING frames (type=0x01) to verify that their peers are still
alive or to check reachability to the peer. The PING frame contains no
additional fields.
The receiver of a PING frame simply needs to acknowledge the packet containing
this frame.
The PING frame can be used to keep a connection alive when an application or
application protocol wishes to prevent the connection from timing out. An
application protocol SHOULD provide guidance about the conditions under which
generating a PING is recommended. This guidance SHOULD indicate whether it is
the client or the server that is expected to send the PING. Having both
endpoints send PING frames without coordination can produce an excessive number
of packets and poor performance.
A connection will time out if no packets are sent or received for a period
longer than the time specified in the idle_timeout transport parameter (see
{{termination}}). However, state in middleboxes might time out earlier than
that. Though REQ-5 in {{?RFC4787}} recommends a 2 minute timeout interval,
experience shows that sending packets every 15 to 30 seconds is necessary to
prevent the majority of middleboxes from losing state for UDP flows.
## ACK Frames {#frame-ack}
Receivers send ACK frames (types 0x02 and 0x03) to inform senders of packets
they have received and processed. The ACK frame contains one or more ACK Ranges.
ACK Ranges identify acknowledged packets. If the frame type is 0x03, ACK frames
also contain the sum of QUIC packets with associated ECN marks received on the
connection up until this point. QUIC implementations MUST properly handle both
types and, if they have enabled ECN for packets they send, they SHOULD use the
information in the ECN section to manage their congestion state.
QUIC acknowledgements are irrevocable. Once acknowledged, a packet remains
acknowledged, even if it does not appear in a future ACK frame. This is unlike
TCP SACKs ({{?RFC2018}}).
It is expected that a sender will reuse the same packet number across different
packet number spaces. ACK frames only acknowledge the packet numbers that were
transmitted by the sender in the same packet number space of the packet that the
ACK was received in.
Version Negotiation and Retry packets cannot be acknowledged because they do not
contain a packet number. Rather than relying on ACK frames, these packets are
implicitly acknowledged by the next Initial packet sent by the client.
An ACK frame is as follows:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Largest Acknowledged (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ACK Delay (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ACK Range Count (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| First ACK Range (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ACK Ranges (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [ECN Counts] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #ack-format title="ACK Frame Format"}
ACK frames contain the following fields:
Largest Acknowledged:
: A variable-length integer representing the largest packet number the peer is
acknowledging; this is usually the largest packet number that the peer has
received prior to generating the ACK frame. Unlike the packet number in the
QUIC long or short header, the value in an ACK frame is not truncated.
ACK Delay:
: A variable-length integer representing the time delta in microseconds between
when this ACK was sent and when the largest acknowledged packet, as indicated
in the Largest Acknowledged field, was received by this peer. The value of
the ACK Delay field is scaled by multiplying the encoded value by 2 to the
power of the value of the `ack_delay_exponent` transport parameter set by the
sender of the ACK frame (see {{transport-parameter-definitions}}). Scaling in
this fashion allows for a larger range of values with a shorter encoding at
the cost of lower resolution. Because the receiver doesn't use the ACK Delay
for Initial and Handshake packets, a sender SHOULD send a value of 0.
ACK Range Count:
: A variable-length integer specifying the number of Gap and ACK Range fields in
the frame.
First ACK Range:
: A variable-length integer indicating the number of contiguous packets
preceding the Largest Acknowledged that are being acknowledged. The First ACK
Range is encoded as an ACK Range (see {{ack-ranges}}) starting from the
Largest Acknowledged. That is, the smallest packet acknowledged in the
range is determined by subtracting the First ACK Range value from the Largest
Acknowledged.
ACK Ranges:
: Contains additional ranges of packets which are alternately not
acknowledged (Gap) and acknowledged (ACK Range); see {{ack-ranges}}.
ECN Counts:
: The three ECN Counts; see {{ack-ecn-counts}}.
### ACK Ranges {#ack-ranges}
The ACK Ranges field consists of alternating Gap and ACK Range values in
descending packet number order. The number of Gap and ACK Range values is
determined by the ACK Range Count field; one of each value is present for each
value in the ACK Range Count field.
ACK Ranges are structured as follows:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Gap (i)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [ACK Range (i)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Gap (i)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [ACK Range (i)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Gap (i)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [ACK Range (i)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #ack-range-format title="ACK Ranges"}
The fields that form the ACK Ranges are:
Gap (repeated):
: A variable-length integer indicating the number of contiguous unacknowledged
packets preceding the packet number one lower than the smallest in the
preceding ACK Range.
ACK Range (repeated):
: A variable-length integer indicating the number of contiguous acknowledged
packets preceding the largest packet number, as determined by the
preceding Gap.
Gap and ACK Range value use a relative integer encoding for efficiency. Though
each encoded value is positive, the values are subtracted, so that each ACK
Range describes progressively lower-numbered packets.
Each ACK Range acknowledges a contiguous range of packets by indicating the
number of acknowledged packets that precede the largest packet number in that
range. A value of zero indicates that only the largest packet number is
acknowledged. Larger ACK Range values indicate a larger range, with
corresponding lower values for the smallest packet number in the range. Thus,
given a largest packet number for the range, the smallest value is determined by
the formula:
~~~
smallest = largest - ack_range
~~~
An ACK Range acknowledges all packets between the smallest packet number and the
largest, inclusive.
The largest value for an ACK Range is determined by cumulatively subtracting the
size of all preceding ACK Ranges and Gaps.
Each Gap indicates a range of packets that are not being acknowledged. The
number of packets in the gap is one higher than the encoded value of the Gap
field.
The value of the Gap field establishes the largest packet number value for the
subsequent ACK Range using the following formula:
~~~
largest = previous_smallest - gap - 2
~~~
If any computed packet number is negative, an endpoint MUST generate a
connection error of type FRAME_ENCODING_ERROR indicating an error in an ACK
frame.
### ECN Counts {#ack-ecn-counts}
The ACK frame uses the least significant bit (that is, type 0x03) to indicate
ECN feedback and report receipt of QUIC packets with associated ECN codepoints
of ECT(0), ECT(1), or CE in the packet's IP header. ECN Counts are only present
when the ACK frame type is 0x03.
ECN Counts are only parsed when the ACK frame type is 0x03. There are 3 ECN
counts, as follows:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ECT(0) Count (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ECT(1) Count (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ECN-CE Count (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
The three ECN Counts are:
ECT(0) Count:
: A variable-length integer representing the total number of packets received
with the ECT(0) codepoint in the packet number space of the ACK frame.
ECT(1) Count:
: A variable-length integer representing the total number of packets received
with the ECT(1) codepoint in the packet number space of the ACK frame.
CE Count:
: A variable-length integer representing the total number of packets received
with the CE codepoint in the packet number space of the ACK frame.
ECN counts are maintained separately for each packet number space.
## RESET_STREAM Frame {#frame-reset-stream}
An endpoint uses a RESET_STREAM frame (type=0x04) to abruptly terminate the
sending part of a stream.
After sending a RESET_STREAM, an endpoint ceases transmission and retransmission
of STREAM frames on the identified stream. A receiver of RESET_STREAM can
discard any data that it already received on that stream.
An endpoint that receives a RESET_STREAM frame for a send-only stream MUST
terminate the connection with error STREAM_STATE_ERROR.
The RESET_STREAM frame is as follows:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Application Error Code (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Final Size (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
RESET_STREAM frames contain the following fields:
Stream ID:
: A variable-length integer encoding of the Stream ID of the stream being
terminated.
Application Protocol Error Code:
: A variable-length integer containing the application protocol error
code (see {{app-error-codes}}) which indicates why the stream is being
closed.
Final Size:
: A variable-length integer indicating the final size of the stream by the
RESET_STREAM sender, in unit of bytes.
## STOP_SENDING Frame {#frame-stop-sending}
An endpoint uses a STOP_SENDING frame (type=0x05) to communicate that incoming
data is being discarded on receipt at application request. STOP_SENDING
requests that a peer cease transmission on a stream.
A STOP_SENDING frame can be sent for streams in the Recv or Size Known states
(see {{stream-send-states}}). Receiving a STOP_SENDING frame for a
locally-initiated stream that has not yet been created MUST be treated as a
connection error of type STREAM_STATE_ERROR. An endpoint that receives a
STOP_SENDING frame for a receive-only stream MUST terminate the connection with
error STREAM_STATE_ERROR.
The STOP_SENDING frame is as follows:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Application Error Code (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
STOP_SENDING frames contain the following fields:
Stream ID:
: A variable-length integer carrying the Stream ID of the stream being ignored.
Application Error Code:
: A variable-length integer containing the application-specified reason the
sender is ignoring the stream (see {{app-error-codes}}).
## CRYPTO Frame {#frame-crypto}
The CRYPTO frame (type=0x06) is used to transmit cryptographic handshake
messages. It can be sent in all packet types except 0-RTT. The CRYPTO frame
offers the cryptographic protocol an in-order stream of bytes. CRYPTO frames
are functionally identical to STREAM frames, except that they do not bear a
stream identifier; they are not flow controlled; and they do not carry markers
for optional offset, optional length, and the end of the stream.
The CRYPTO frame is as follows:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Offset (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Length (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Crypto Data (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #crypto-format title="CRYPTO Frame Format"}
CRYPTO frames contain the following fields:
Offset:
: A variable-length integer specifying the byte offset in the stream for the
data in this CRYPTO frame.
Length:
: A variable-length integer specifying the length of the Crypto Data field in
this CRYPTO frame.
Crypto Data:
: The cryptographic message data.
There is a separate flow of cryptographic handshake data in each encryption
level, each of which starts at an offset of 0. This implies that each encryption
level is treated as a separate CRYPTO stream of data.
Unlike STREAM frames, which include a Stream ID indicating to which stream the
data belongs, the CRYPTO frame carries data for a single stream per encryption
level. The stream does not have an explicit end, so CRYPTO frames do not have a
FIN bit.
## NEW_TOKEN Frame {#frame-new-token}
A server sends a NEW_TOKEN frame (type=0x07) to provide the client with a token
to send in the header of an Initial packet for a future connection.
The NEW_TOKEN frame is as follows:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Token Length (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Token (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
NEW_TOKEN frames contain the following fields:
Token Length:
: A variable-length integer specifying the length of the token in bytes.
Token:
: An opaque blob that the client may use with a future Initial packet.
An endpoint might receive multiple NEW_TOKEN frames that contain the same token
value. Endpoints are responsible for discarding duplicate values, which might
be used to link connection attempts; see {{validate-future}}.
Clients MUST NOT send NEW_TOKEN frames. Servers MUST treat receipt of a
NEW_TOKEN frame as a connection error of type PROTOCOL_VIOLATION.
## STREAM Frames {#frame-stream}
STREAM frames implicitly create a stream and carry stream data. The STREAM
frame takes the form 0b00001XXX (or the set of values from 0x08 to 0x0f). The
value of the three low-order bits of the frame type determines the fields that
are present in the frame.
* The OFF bit (0x04) in the frame type is set to indicate that there is an
Offset field present. When set to 1, the Offset field is present. When set
to 0, the Offset field is absent and the Stream Data starts at an offset of 0
(that is, the frame contains the first bytes of the stream, or the end of a
stream that includes no data).
* The LEN bit (0x02) in the frame type is set to indicate that there is a Length
field present. If this bit is set to 0, the Length field is absent and the
Stream Data field extends to the end of the packet. If this bit is set to 1,
the Length field is present.
* The FIN bit (0x01) of the frame type is set only on frames that contain the
final size of the stream. Setting this bit indicates that the frame
marks the end of the stream.
An endpoint that receives a STREAM frame for a send-only stream MUST terminate
the connection with error STREAM_STATE_ERROR.
The STREAM frames are as follows:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Offset (i)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Length (i)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream Data (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #stream-format title="STREAM Frame Format"}
STREAM frames contain the following fields:
Stream ID:
: A variable-length integer indicating the stream ID of the stream (see
{{stream-id}}).
Offset:
: A variable-length integer specifying the byte offset in the stream for the
data in this STREAM frame. This field is present when the OFF bit is set to
1. When the Offset field is absent, the offset is 0.
Length:
: A variable-length integer specifying the length of the Stream Data field in
this STREAM frame. This field is present when the LEN bit is set to 1. When
the LEN bit is set to 0, the Stream Data field consumes all the remaining
bytes in the packet.
Stream Data:
: The bytes from the designated stream to be delivered.
When a Stream Data field has a length of 0, the offset in the STREAM frame is
the offset of the next byte that would be sent.
The first byte in the stream has an offset of 0. The largest offset delivered
on a stream - the sum of the offset and data length - cannot exceed 2^62-1, as
it is not possible to provide flow control credit for that data. Receipt of a
frame that exceeds this limit will be treated as a connection error of type
FLOW_CONTROL_ERROR.
## MAX_DATA Frame {#frame-max-data}
The MAX_DATA frame (type=0x10) is used in flow control to inform the peer of
the maximum amount of data that can be sent on the connection as a whole.
The MAX_DATA frame is as follows:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Maximum Data (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
MAX_DATA frames contain the following fields:
Maximum Data:
: A variable-length integer indicating the maximum amount of data that can be
sent on the entire connection, in units of bytes.
All data sent in STREAM frames counts toward this limit. The sum of the largest
received offsets on all streams - including streams in terminal states - MUST
NOT exceed the value advertised by a receiver. An endpoint MUST terminate a
connection with a FLOW_CONTROL_ERROR error if it receives more data than the
maximum data value that it has sent, unless this is a result of a change in
the initial limits (see {{zerortt-parameters}}).
## MAX_STREAM_DATA Frame {#frame-max-stream-data}
The MAX_STREAM_DATA frame (type=0x11) is used in flow control to inform a peer
of the maximum amount of data that can be sent on a stream.
A MAX_STREAM_DATA frame can be sent for streams in the Recv state (see
{{stream-send-states}}). Receiving a MAX_STREAM_DATA frame for a
locally-initiated stream that has not yet been created MUST be treated as a
connection error of type STREAM_STATE_ERROR. An endpoint that receives a
MAX_STREAM_DATA frame for a receive-only stream MUST terminate the connection
with error STREAM_STATE_ERROR.
The MAX_STREAM_DATA frame is as follows:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Maximum Stream Data (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
MAX_STREAM_DATA frames contain the following fields:
Stream ID:
: The stream ID of the stream that is affected encoded as a variable-length
integer.
Maximum Stream Data:
: A variable-length integer indicating the maximum amount of data that can be
sent on the identified stream, in units of bytes.
When counting data toward this limit, an endpoint accounts for the largest
received offset of data that is sent or received on the stream. Loss or
reordering can mean that the largest received offset on a stream can be greater
than the total size of data received on that stream. Receiving STREAM frames
might not increase the largest received offset.
The data sent on a stream MUST NOT exceed the largest maximum stream data value
advertised by the receiver. An endpoint MUST terminate a connection with a
FLOW_CONTROL_ERROR error if it receives more data than the largest maximum
stream data that it has sent for the affected stream, unless this is a result of
a change in the initial limits (see {{zerortt-parameters}}).
## MAX_STREAMS Frames {#frame-max-streams}
The MAX_STREAMS frames (type=0x12 and 0x13) inform the peer of the cumulative
number of streams of a given type it is permitted to open. A MAX_STREAMS frame
with a type of 0x12 applies to bidirectional streams, and a MAX_STREAMS frame
with a type of 0x13 applies to unidirectional streams.
The MAX_STREAMS frames are as follows:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Maximum Streams (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
MAX_STREAMS frames contain the following fields:
Maximum Streams:
: A count of the cumulative number of streams of the corresponding type that
can be opened over the lifetime of the connection.
Loss or reordering can cause a MAX_STREAMS frame to be received which states a
lower stream limit than an endpoint has previously received. MAX_STREAMS frames
which do not increase the stream limit MUST be ignored.
An endpoint MUST NOT open more streams than permitted by the current stream
limit set by its peer. For instance, a server that receives a unidirectional
stream limit of 3 is permitted to open stream 3, 7, and 11, but not stream 15.
An endpoint MUST terminate a connection with a STREAM_LIMIT_ERROR error if a
peer opens more streams than was permitted.
Note that these frames (and the corresponding transport parameters) do not
describe the number of streams that can be opened concurrently. The limit
includes streams that have been closed as well as those that are open.
## DATA_BLOCKED Frame {#frame-data-blocked}
A sender SHOULD send a DATA_BLOCKED frame (type=0x14) when it wishes to send
data, but is unable to due to connection-level flow control (see
{{flow-control}}). DATA_BLOCKED frames can be used as input to tuning of flow
control algorithms (see {{fc-credit}}).
The DATA_BLOCKED frame is as follows:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Data Limit (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
DATA_BLOCKED frames contain the following fields:
Data Limit:
: A variable-length integer indicating the connection-level limit at which
blocking occurred.
## STREAM_DATA_BLOCKED Frame {#frame-stream-data-blocked}
A sender SHOULD send a STREAM_DATA_BLOCKED frame (type=0x15) when it wishes to
send data, but is unable to due to stream-level flow control. This frame is
analogous to DATA_BLOCKED ({{frame-data-blocked}}).
An endpoint that receives a STREAM_DATA_BLOCKED frame for a send-only stream
MUST terminate the connection with error STREAM_STATE_ERROR.
The STREAM_DATA_BLOCKED frame is as follows:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream Data Limit (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
STREAM_DATA_BLOCKED frames contain the following fields:
Stream ID:
: A variable-length integer indicating the stream which is flow control blocked.
Stream Data Limit:
: A variable-length integer indicating the offset of the stream at which the
blocking occurred.
## STREAMS_BLOCKED Frames {#frame-streams-blocked}
A sender SHOULD send a STREAMS_BLOCKED frame (type=0x16 or 0x17) when it wishes
to open a stream, but is unable to due to the maximum stream limit set by its
peer (see {{frame-max-streams}}). A STREAMS_BLOCKED frame of type 0x16 is used
to indicate reaching the bidirectional stream limit, and a STREAMS_BLOCKED frame
of type 0x17 indicates reaching the unidirectional stream limit.
A STREAMS_BLOCKED frame does not open the stream, but informs the peer that a
new stream was needed and the stream limit prevented the creation of the stream.
The STREAMS_BLOCKED frames are as follows:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream Limit (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
STREAMS_BLOCKED frames contain the following fields:
Stream Limit:
: A variable-length integer indicating the stream limit at the time the frame
was sent.
## NEW_CONNECTION_ID Frame {#frame-new-connection-id}
An endpoint sends a NEW_CONNECTION_ID frame (type=0x18) to provide its peer with
alternative connection IDs that can be used to break linkability when migrating
connections (see {{migration-linkability}}).
The NEW_CONNECTION_ID frame is as follows:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Sequence Number (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Retire Prior To (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Length (8) | |
+-+-+-+-+-+-+-+-+ Connection ID (8..160) +
| ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ +
| |
+ Stateless Reset Token (128) +
| |
+ +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
NEW_CONNECTION_ID frames contain the following fields:
Sequence Number:
: The sequence number assigned to the connection ID by the sender. See
{{issue-cid}}.
Retire Prior To:
: A variable-length integer indicating which connection IDs should be retired.
See {{retiring-cids}}.
Length:
: An 8-bit unsigned integer containing the length of the connection ID. Values
less than 1 and greater than 20 are invalid and MUST be treated as a
connection error of type PROTOCOL_VIOLATION.
Connection ID:
: A connection ID of the specified length.
Stateless Reset Token:
: A 128-bit value that will be used for a stateless reset when the associated
connection ID is used (see {{stateless-reset}}).
An endpoint MUST NOT send this frame if it currently requires that its peer send
packets with a zero-length Destination Connection ID. Changing the length of a
connection ID to or from zero-length makes it difficult to identify when the
value of the connection ID changed. An endpoint that is sending packets with a
zero-length Destination Connection ID MUST treat receipt of a NEW_CONNECTION_ID
frame as a connection error of type PROTOCOL_VIOLATION.
Transmission errors, timeouts and retransmissions might cause the same
NEW_CONNECTION_ID frame to be received multiple times. Receipt of the same
frame multiple times MUST NOT be treated as a connection error. A receiver can
use the sequence number supplied in the NEW_CONNECTION_ID frame to identify new
connection IDs from old ones.
If an endpoint receives a NEW_CONNECTION_ID frame that repeats a previously
issued connection ID with a different Stateless Reset Token or a different
sequence number, or if a sequence number is used for different connection
IDs, the endpoint MAY treat that receipt as a connection error of type
PROTOCOL_VIOLATION.
The Retire Prior To field is a request for the peer to retire all connection IDs
with a sequence number less than the specified value. This includes the initial
and preferred_address transport parameter connection IDs. The peer SHOULD
retire the corresponding connection IDs and send the corresponding
RETIRE_CONNECTION_ID frames in a timely manner.
The Retire Prior To field MUST be less than or equal to the Sequence Number
field. Receiving a value greater than the Sequence Number MUST be treated as a
connection error of type PROTOCOL_VIOLATION.
Once a sender indicates a Retire Prior To value, smaller values sent in
subsequent NEW_CONNECTION_ID frames have no effect. A receiver MUST ignore any
Retire Prior To fields that do not increase the largest received Retire Prior To
value.
## RETIRE_CONNECTION_ID Frame {#frame-retire-connection-id}
An endpoint sends a RETIRE_CONNECTION_ID frame (type=0x19) to indicate that it
will no longer use a connection ID that was issued by its peer. This may include
the connection ID provided during the handshake. Sending a RETIRE_CONNECTION_ID
frame also serves as a request to the peer to send additional connection IDs for
future use (see {{connection-id}}). New connection IDs can be delivered to a
peer using the NEW_CONNECTION_ID frame ({{frame-new-connection-id}}).
Retiring a connection ID invalidates the stateless reset token associated with
that connection ID.
The RETIRE_CONNECTION_ID frame is as follows:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Sequence Number (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
RETIRE_CONNECTION_ID frames contain the following fields:
Sequence Number:
: The sequence number of the connection ID being retired. See
{{retiring-cids}}.
Receipt of a RETIRE_CONNECTION_ID frame containing a sequence number greater
than any previously sent to the peer MAY be treated as a connection error of
type PROTOCOL_VIOLATION.
The sequence number specified in a RETIRE_CONNECTION_ID frame MUST NOT refer
to the Destination Connection ID field of the packet in which the frame is
contained. The peer MAY treat this as a connection error of type
PROTOCOL_VIOLATION.
An endpoint cannot send this frame if it was provided with a zero-length
connection ID by its peer. An endpoint that provides a zero-length connection
ID MUST treat receipt of a RETIRE_CONNECTION_ID frame as a connection error of
type PROTOCOL_VIOLATION.
## PATH_CHALLENGE Frame {#frame-path-challenge}
Endpoints can use PATH_CHALLENGE frames (type=0x1a) to check reachability to the
peer and for path validation during connection migration.
The PATH_CHALLENGE frames are as follows:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ Data (64) +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
PATH_CHALLENGE frames contain the following fields:
Data:
: This 8-byte field contains arbitrary data.
A PATH_CHALLENGE frame containing 8 bytes that are hard to guess is sufficient
to ensure that it is easier to receive the packet than it is to guess the value
correctly.
The recipient of this frame MUST generate a PATH_RESPONSE frame
({{frame-path-response}}) containing the same Data.
## PATH_RESPONSE Frame {#frame-path-response}
The PATH_RESPONSE frame (type=0x1b) is sent in response to a PATH_CHALLENGE
frame. Its format is identical to the PATH_CHALLENGE frame
({{frame-path-challenge}}).
If the content of a PATH_RESPONSE frame does not match the content of a
PATH_CHALLENGE frame previously sent by the endpoint, the endpoint MAY generate
a connection error of type PROTOCOL_VIOLATION.
## CONNECTION_CLOSE Frames {#frame-connection-close}
An endpoint sends a CONNECTION_CLOSE frame (type=0x1c or 0x1d) to notify its
peer that the connection is being closed. The CONNECTION_CLOSE with a frame
type of 0x1c is used to signal errors at only the QUIC layer, or the absence of
errors (with the NO_ERROR code). The CONNECTION_CLOSE frame with a type of 0x1d
is used to signal an error with the application that uses QUIC.
If there are open streams that haven't been explicitly closed, they are
implicitly closed when the connection is closed.
The CONNECTION_CLOSE frames are as follows:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Error Code (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [ Frame Type (i) ] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Reason Phrase Length (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Reason Phrase (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
CONNECTION_CLOSE frames contain the following fields:
Error Code:
: A variable length integer error code which indicates the reason for
closing this connection. A CONNECTION_CLOSE frame of type 0x1c uses codes
from the space defined in {{error-codes}}. A CONNECTION_CLOSE frame of
type 0x1d uses codes from the application protocol error code space;
see {{app-error-codes}}
Frame Type:
: A variable-length integer encoding the type of frame that triggered the error.
A value of 0 (equivalent to the mention of the PADDING frame) is used when the
frame type is unknown. The application-specific variant of CONNECTION_CLOSE
(type 0x1d) does not include this field.
Reason Phrase Length:
: A variable-length integer specifying the length of the reason phrase in bytes.
Because a CONNECTION_CLOSE frame cannot be split between packets, any limits
on packet size will also limit the space available for a reason phrase.
Reason Phrase:
: A human-readable explanation for why the connection was closed. This can be
zero length if the sender chooses to not give details beyond the Error Code.
This SHOULD be a UTF-8 encoded string {{!RFC3629}}.
## Extension Frames
QUIC frames do not use a self-describing encoding. An endpoint therefore needs
to understand the syntax of all frames before it can successfully process a
packet. This allows for efficient encoding of frames, but it means that an
endpoint cannot send a frame of a type that is unknown to its peer.
An extension to QUIC that wishes to use a new type of frame MUST first ensure
that a peer is able to understand the frame. An endpoint can use a transport
parameter to signal its willingness to receive one or more extension frame types
with the one transport parameter.
Extension frames MUST be congestion controlled and MUST cause an ACK frame to
be sent. The exception is extension frames that replace or supplement the ACK
frame. Extension frames are not included in flow control unless specified
in the extension.
An IANA registry is used to manage the assignment of frame types; see
{{iana-frames}}.
# Transport Error Codes {#error-codes}
QUIC error codes are 62-bit unsigned integers.
This section lists the defined QUIC transport error codes that may be used in a
CONNECTION_CLOSE frame. These errors apply to the entire connection.
NO_ERROR (0x0):
: An endpoint uses this with CONNECTION_CLOSE to signal that the connection is
being closed abruptly in the absence of any error.
INTERNAL_ERROR (0x1):
: The endpoint encountered an internal error and cannot continue with the
connection.
SERVER_BUSY (0x2):
: The server is currently busy and does not accept any new connections.
FLOW_CONTROL_ERROR (0x3):
: An endpoint received more data than it permitted in its advertised data limits
(see {{flow-control}}).
STREAM_LIMIT_ERROR (0x4):
: An endpoint received a frame for a stream identifier that exceeded its
advertised stream limit for the corresponding stream type.
STREAM_STATE_ERROR (0x5):
: An endpoint received a frame for a stream that was not in a state that
permitted that frame (see {{stream-states}}).
FINAL_SIZE_ERROR (0x6):
: An endpoint received a STREAM frame containing data that exceeded the
previously established final size. Or an endpoint received a STREAM frame or
a RESET_STREAM frame containing a final size that was lower than the size of
stream data that was already received. Or an endpoint received a STREAM frame
or a RESET_STREAM frame containing a different final size to the one already
established.
FRAME_ENCODING_ERROR (0x7):
: An endpoint received a frame that was badly formatted. For instance, a frame
of an unknown type, or an ACK frame that has more acknowledgment ranges than
the remainder of the packet could carry.
TRANSPORT_PARAMETER_ERROR (0x8):
: An endpoint received transport parameters that were badly formatted, included
an invalid value, was absent even though it is mandatory, was present though
it is forbidden, or is otherwise in error.
PROTOCOL_VIOLATION (0xA):
: An endpoint detected an error with protocol compliance that was not covered by
more specific error codes.
CRYPTO_BUFFER_EXCEEDED (0xD):
: An endpoint has received more data in CRYPTO frames than it can buffer.
CRYPTO_ERROR (0x1XX):
: The cryptographic handshake failed. A range of 256 values is reserved for
carrying error codes specific to the cryptographic handshake that is used.
Codes for errors occurring when TLS is used for the crypto handshake are
described in Section 4.8 of {{QUIC-TLS}}.
See {{iana-error-codes}} for details of registering new error codes.
In defining these error codes, several principles are applied. Error conditions
that might require specific action on the part of a recipient are given unique
codes. Errors that represent common conditions are given specific codes.
Absent either of these conditions, error codes are used to identify a general
function of the stack, like flow control or transport parameter handling.
Finally, generic errors are provided for conditions where implementations are
unable or unwilling to use more specific codes.
## Application Protocol Error Codes {#app-error-codes}
Application protocol error codes are 62-bit unsigned integers, but the
management of application error codes is left to application protocols.
Application protocol error codes are used for the RESET_STREAM frame
({{frame-reset-stream}}), the STOP_SENDING frame ({{frame-stop-sending}}), and
the CONNECTION_CLOSE frame with a type of 0x1d ({{frame-connection-close}}).
# Security Considerations
## Handshake Denial of Service
As an encrypted and authenticated transport QUIC provides a range of protections
against denial of service. Once the cryptographic handshake is complete, QUIC
endpoints discard most packets that are not authenticated, greatly limiting the
ability of an attacker to interfere with existing connections.
Once a connection is established QUIC endpoints might accept some
unauthenticated ICMP packets (see {{icmp-pmtud}}), but the use of these packets
is extremely limited. The only other type of packet that an endpoint might
accept is a stateless reset ({{stateless-reset}}) which relies on the token
being kept secret until it is used.
During the creation of a connection, QUIC only provides protection against
attack from off the network path. All QUIC packets contain proof that the
recipient saw a preceding packet from its peer.
The first mechanism used is the source and destination connection IDs, which are
required to match those set by a peer. Except for an Initial and stateless
reset packets, an endpoint only accepts packets that include a destination
connection that matches a connection ID the endpoint previously chose. This is
the only protection offered for Version Negotiation packets.
The destination connection ID in an Initial packet is selected by a client to be
unpredictable, which serves an additional purpose. The packets that carry the
cryptographic handshake are protected with a key that is derived from this
connection ID and salt specific to the QUIC version. This allows endpoints to
use the same process for authenticating packets that they receive as they use
after the cryptographic handshake completes. Packets that cannot be
authenticated are discarded. Protecting packets in this fashion provides a
strong assurance that the sender of the packet saw the Initial packet and
understood it.
These protections are not intended to be effective against an attacker that is
able to receive QUIC packets prior to the connection being established. Such an
attacker can potentially send packets that will be accepted by QUIC endpoints.
This version of QUIC attempts to detect this sort of attack, but it expects that
endpoints will fail to establish a connection rather than recovering. For the
most part, the cryptographic handshake protocol {{QUIC-TLS}} is responsible for
detecting tampering during the handshake.
Endpoints are permitted to use other methods to detect and attempt to recover
from interference with the handshake. Invalid packets may be identified and
discarded using other methods, but no specific method is mandated in this
document.
## Amplification Attack
An attacker might be able to receive an address validation token
({{address-validation}}) from a server and then release the IP address it used
to acquire that token. At a later time, the attacker may initiate a 0-RTT
connection with a server by spoofing this same address, which might now address
a different (victim) endpoint. The attacker can thus potentially cause the
server to send an initial congestion window's worth of data towards the victim.
Servers SHOULD provide mitigations for this attack by limiting the usage and
lifetime of address validation tokens (see {{validate-future}}).
## Optimistic ACK Attack
An endpoint that acknowledges packets it has not received might cause a
congestion controller to permit sending at rates beyond what the network
supports. An endpoint MAY skip packet numbers when sending packets to detect
this behavior. An endpoint can then immediately close the connection with a
connection error of type PROTOCOL_VIOLATION (see {{immediate-close}}).
## Slowloris Attacks
The attacks commonly known as Slowloris {{SLOWLORIS}} try to keep many
connections to the target endpoint open and hold them open as long as possible.
These attacks can be executed against a QUIC endpoint by generating the minimum
amount of activity necessary to avoid being closed for inactivity. This might
involve sending small amounts of data, gradually opening flow control windows in
order to control the sender rate, or manufacturing ACK frames that simulate a
high loss rate.
QUIC deployments SHOULD provide mitigations for the Slowloris attacks, such as
increasing the maximum number of clients the server will allow, limiting the
number of connections a single IP address is allowed to make, imposing
restrictions on the minimum transfer speed a connection is allowed to have, and
restricting the length of time an endpoint is allowed to stay connected.
## Stream Fragmentation and Reassembly Attacks
An adversarial sender might intentionally send fragments of stream data in
order to cause disproportionate receive buffer memory commitment and/or
creation of a large and inefficient data structure.
An adversarial receiver might intentionally not acknowledge packets
containing stream data in order to force the sender to store the
unacknowledged stream data for retransmission.
The attack on receivers is mitigated if flow control windows correspond to
available memory. However, some receivers will over-commit memory and
advertise flow control offsets in the aggregate that exceed actual available
memory. The over-commitment strategy can lead to better performance when
endpoints are well behaved, but renders endpoints vulnerable to the stream
fragmentation attack.
QUIC deployments SHOULD provide mitigations against stream fragmentation
attacks. Mitigations could consist of avoiding over-committing memory,
limiting the size of tracking data structures, delaying reassembly
of STREAM frames, implementing heuristics based on the age and
duration of reassembly holes, or some combination.
## Stream Commitment Attack
An adversarial endpoint can open lots of streams, exhausting state on an
endpoint. The adversarial endpoint could repeat the process on a large number
of connections, in a manner similar to SYN flooding attacks in TCP.
Normally, clients will open streams sequentially, as explained in {{stream-id}}.
However, when several streams are initiated at short intervals, loss or
reordering may cause STREAM frames that open streams to be received out of
sequence. On receiving a higher-numbered stream ID, a receiver is required to
open all intervening streams of the same type (see {{stream-recv-states}}).
Thus, on a new connection, opening stream 4000000 opens 1 million and 1
client-initiated bidirectional streams.
The number of active streams is limited by the initial_max_streams_bidi and
initial_max_streams_uni transport parameters, as explained in
{{controlling-concurrency}}. If chosen judiciously, these limits mitigate the
effect of the stream commitment attack. However, setting the limit too low
could affect performance when applications expect to open large number of
streams.
## Peer Denial of Service {#useless}
QUIC and TLS both contain messages that have legitimate uses in some contexts,
but that can be abused to cause a peer to expend processing resources without
having any observable impact on the state of the connection.
Messages can also be used to change and revert state in small or inconsequential
ways, such as by sending small increments to flow control limits.
If processing costs are disproportionately large in comparison to bandwidth
consumption or effect on state, then this could allow a malicious peer to
exhaust processing capacity.
While there are legitimate uses for all messages, implementations SHOULD track
cost of processing relative to progress and treat excessive quantities of any
non-productive packets as indicative of an attack. Endpoints MAY respond to
this condition with a connection error, or by dropping packets.
## Explicit Congestion Notification Attacks {#security-ecn}
An on-path attacker could manipulate the value of ECN codepoints in the IP
header to influence the sender's rate. {{!RFC3168}} discusses manipulations and
their effects in more detail.
An on-the-side attacker can duplicate and send packets with modified ECN
codepoints to affect the sender's rate. If duplicate packets are discarded by a
receiver, an off-path attacker will need to race the duplicate packet against
the original to be successful in this attack. Therefore, QUIC endpoints ignore
the ECN codepoint field on an IP packet unless at least one QUIC packet in that
IP packet is successfully processed; see {{ecn}}.
## Stateless Reset Oracle {#reset-oracle}
Stateless resets create a possible denial of service attack analogous to a TCP
reset injection. This attack is possible if an attacker is able to cause a
stateless reset token to be generated for a connection with a selected
connection ID. An attacker that can cause this token to be generated can reset
an active connection with the same connection ID.
If a packet can be routed to different instances that share a static key, for
example by changing an IP address or port, then an attacker can cause the server
to send a stateless reset. To defend against this style of denial service,
endpoints that share a static key for stateless reset (see {{reset-token}}) MUST
be arranged so that packets with a given connection ID always arrive at an
instance that has connection state, unless that connection is no longer active.
In the case of a cluster that uses dynamic load balancing, it's possible that a
change in load balancer configuration could happen while an active instance
retains connection state; even if an instance retains connection state, the
change in routing and resulting stateless reset will result in the connection
being terminated. If there is no chance in the packet being routed to the
correct instance, it is better to send a stateless reset than wait for
connections to time out. However, this is acceptable only if the routing cannot
be influenced by an attacker.
## Version Downgrade {#version-downgrade}
This document defines QUIC Version Negotiation packets {{version-negotiation}},
which can be used to negotiate the QUIC version used between two endpoints.
However, this document does not specify how this negotiation will be performed
between this version and subsequent future versions. In particular, Version
Negotiation packets do not contain any mechanism to prevent version downgrade
attacks. Future versions of QUIC that use Version Negotiation packets MUST
define a mechanism that is robust against version downgrade attacks.
## Targeted Attacks by Routing
Deployments should limit the ability of an attacker to target a new connection
to a particular server instance. This means that client-controlled fields, such
as the initial Destination Connection ID used on Initial and 0-RTT packets
SHOULD NOT be used by themselves to make routing decisions. Ideally, routing
decisions are made independently of client-selected values; a Source Connection
ID can be selected to route later packets to the same server.
# IANA Considerations
## QUIC Transport Parameter Registry {#iana-transport-parameters}
IANA \[SHALL add/has added] a registry for "QUIC Transport Parameters" under a
"QUIC Protocol" heading.
The "QUIC Transport Parameters" registry governs a 16-bit space. This space is
split into two spaces that are governed by different policies. Values with the
first byte in the range 0x00 to 0xfe (in hexadecimal) are assigned via the
Specification Required policy {{!RFC8126}}. Values with the first byte 0xff are
reserved for Private Use {{!RFC8126}}.
Registrations MUST include the following fields:
Value:
: The numeric value of the assignment (registrations will be between 0x0000 and
0xfeff).
Parameter Name:
: A short mnemonic for the parameter.
Specification:
: A reference to a publicly available specification for the value.
The nominated expert(s) verify that a specification exists and is readily
accessible. Expert(s) are encouraged to be biased towards approving
registrations unless they are abusive, frivolous, or actively harmful (not
merely aesthetically displeasing, or architecturally dubious).
The initial contents of this registry are shown in {{iana-tp-table}}.
| Value | Parameter Name | Specification |
|:-------|:----------------------------|:------------------------------------|
| 0x0000 | original_connection_id | {{transport-parameter-definitions}} |
| 0x0001 | idle_timeout | {{transport-parameter-definitions}} |
| 0x0002 | stateless_reset_token | {{transport-parameter-definitions}} |
| 0x0003 | max_packet_size | {{transport-parameter-definitions}} |
| 0x0004 | initial_max_data | {{transport-parameter-definitions}} |
| 0x0005 | initial_max_stream_data_bidi_local | {{transport-parameter-definitions}} |
| 0x0006 | initial_max_stream_data_bidi_remote | {{transport-parameter-definitions}} |
| 0x0007 | initial_max_stream_data_uni | {{transport-parameter-definitions}} |
| 0x0008 | initial_max_streams_bidi | {{transport-parameter-definitions}} |
| 0x0009 | initial_max_streams_uni | {{transport-parameter-definitions}} |
| 0x000a | ack_delay_exponent | {{transport-parameter-definitions}} |
| 0x000b | max_ack_delay | {{transport-parameter-definitions}} |
| 0x000c | disable_active_migration | {{transport-parameter-definitions}} |
| 0x000d | preferred_address | {{transport-parameter-definitions}} |
| 0x000e | active_connection_id_limit | {{transport-parameter-definitions}} |
{: #iana-tp-table title="Initial QUIC Transport Parameters Entries"}
Additionally, each value of the format `31 * N + 27` for integer values of N
(that is, `27`, `58`, `89`, ...) MUST NOT be assigned by IANA.
## QUIC Frame Type Registry {#iana-frames}
IANA \[SHALL add/has added] a registry for "QUIC Frame Types" under a
"QUIC Protocol" heading.
The "QUIC Frame Types" registry governs a 62-bit space. This space is split
into three spaces that are governed by different policies. Values between 0x00
and 0x3f (in hexadecimal) are assigned via the Standards Action or IESG Review
policies {{!RFC8126}}. Values from 0x40 to 0x3fff operate on the Specification
Required policy {{!RFC8126}}. All other values are assigned to Private Use
{{!RFC8126}}.
Registrations MUST include the following fields:
Value:
: The numeric value of the assignment (registrations will be between 0x00 and
0x3fff). A range of values MAY be assigned.
Frame Name:
: A short mnemonic for the frame type.
Specification:
: A reference to a publicly available specification for the value.
The nominated expert(s) verify that a specification exists and is readily
accessible. Specifications for new registrations need to describe the means by
which an endpoint might determine that it can send the identified type of frame.
An accompanying transport parameter registration (see
{{iana-transport-parameters}}) is expected for most registrations. The
specification needs to describe the format and assigned semantics of any fields
in the frame.
Expert(s) are encouraged to be biased towards approving registrations unless
they are abusive, frivolous, or actively harmful (not merely aesthetically
displeasing, or architecturally dubious).
The initial contents of this registry are tabulated in {{frame-types}}.
## QUIC Transport Error Codes Registry {#iana-error-codes}
IANA \[SHALL add/has added] a registry for "QUIC Transport Error Codes" under a
"QUIC Protocol" heading.
The "QUIC Transport Error Codes" registry governs a 62-bit space. This space is
split into three spaces that are governed by different policies. Values between
0x00 and 0x3f (in hexadecimal) are assigned via the Standards Action or IESG
Review policies {{!RFC8126}}. Values from 0x40 to 0x3fff operate on the
Specification Required policy {{!RFC8126}}. All other values are assigned to
Private Use {{!RFC8126}}.
Registrations MUST include the following fields:
Value:
: The numeric value of the assignment (registrations will be between 0x0000 and
0x3fff).
Code:
: A short mnemonic for the parameter.
Description:
: A brief description of the error code semantics, which MAY be a summary if a
specification reference is provided.
Specification:
: A reference to a publicly available specification for the value.
The nominated expert(s) verify that a specification exists and is readily
accessible. Expert(s) are encouraged to be biased towards approving
registrations unless they are abusive, frivolous, or actively harmful (not
merely aesthetically displeasing, or architecturally dubious).
The initial contents of this registry are shown in {{iana-error-table}}.
| Value | Error | Description | Specification |
|:------|:--------------------------|:------------------------------|:----------------|
| 0x0 | NO_ERROR | No error | {{error-codes}} |
| 0x1 | INTERNAL_ERROR | Implementation error | {{error-codes}} |
| 0x2 | SERVER_BUSY | Server currently busy | {{error-codes}} |
| 0x3 | FLOW_CONTROL_ERROR | Flow control error | {{error-codes}} |
| 0x4 | STREAM_LIMIT_ERROR | Too many streams opened | {{error-codes}} |
| 0x5 | STREAM_STATE_ERROR | Frame received in invalid stream state | {{error-codes}} |
| 0x6 | FINAL_SIZE_ERROR | Change to final size | {{error-codes}} |
| 0x7 | FRAME_ENCODING_ERROR | Frame encoding error | {{error-codes}} |
| 0x8 | TRANSPORT_PARAMETER_ERROR | Error in transport parameters | {{error-codes}} |
| 0xA | PROTOCOL_VIOLATION | Generic protocol violation | {{error-codes}} |
| 0xD | CRYPTO_BUFFER_EXCEEDED | CRYPTO data buffer overflowed | {{error-codes}} |
{: #iana-error-table title="Initial QUIC Transport Error Codes Entries"}
--- back
# Sample Packet Number Decoding Algorithm {#sample-packet-number-decoding}
The following pseudo-code shows how an implementation can decode packet
numbers after header protection has been removed.
~~~
DecodePacketNumber(largest_pn, truncated_pn, pn_nbits):
expected_pn = largest_pn + 1
pn_win = 1 << pn_nbits
pn_hwin = pn_win / 2
pn_mask = pn_win - 1
// The incoming packet number should be greater than
// expected_pn - pn_hwin and less than or equal to
// expected_pn + pn_hwin
//
// This means we can't just strip the trailing bits from
// expected_pn and add the truncated_pn because that might
// yield a value outside the window.
//
// The following code calculates a candidate value and
// makes sure it's within the packet number window.
candidate_pn = (expected_pn & ~pn_mask) | truncated_pn
if candidate_pn <= expected_pn - pn_hwin:
return candidate_pn + pn_win
// Note the extra check for underflow when candidate_pn
// is near zero.
if candidate_pn > expected_pn + pn_hwin and
candidate_pn > pn_win:
return candidate_pn - pn_win
return candidate_pn
~~~
# Change Log
> **RFC Editor's Note:** Please remove this section prior to publication of a
> final version of this document.
Issue and pull request numbers are listed with a leading octothorp.
## Since draft-ietf-quic-transport-22
- Rules for preventing correlation by connection ID tightened (#2084, #2929)
- Clarified use of CONNECTION_CLOSE in Handshake packets (#2151, #2541, #2688)
- Discourage regressions of largest acknowledged in ACK (#2205, #2752)
- Improved robusness of validation process for ECN counts (#2534, #2752)
- Require endpoints to ignore spurious migration attempts (#2342, #2893)
- Transport parameter for disabling migration clarified to allow NAT rebinding
(#2389, #2893)
- Document principles for defining new error codes (#2388, #2880)
- Reserve transport parameters for greasing (#2550, #2873)
- A maximum ACK delay of 0 is used for handshake packet number spaces (#2646,
#2638)
- Improved rules for use of congestion control state on new paths (#2685, #2918)
- Removed recommendation to coordinate spin for multiple connections that share
a path (#2763, #2882)
- Allow smaller stateless resets and recommend a smaller minimum on packets
that might trigger a stateless reset (#2770, #2869, #2927)
- Provide guidance around the interface to QUIC as used by application protocols
(#2805, #2857)
- Frames other than STREAM can cause STREAM_LIMIT_ERROR (#2825, #2826)
- Tighter rules about processing of rejected 0-RTT packets (#2829, #2840, #2841)
- Explanation of the effect of Retry on 0-RTT packets (#2842, #2852)
- Cryptographic handshake needs to provide server transport parameter encryption
(#2920, #2921)
- Moved ACK generation guidance from recovery draft to transport draft (#1860,
#2916).
## Since draft-ietf-quic-transport-21
- Connection ID lengths are now one octet, but limited in version 1 to 20 octets
of length (#2736, #2749)
## Since draft-ietf-quic-transport-20
- Error codes are encoded as variable-length integers (#2672, #2680)
- NEW_CONNECTION_ID includes a request to retire old connection IDs (#2645,
#2769)
- Tighter rules for generating and explicitly eliciting ACK frames (#2546,
#2794)
- Recommend having only one packet per encryption level in a datagram (#2308,
#2747)
- More normative language about use of stateless reset (#2471, #2574)
- Allow reuse of stateless reset tokens (#2732, #2733)
- Allow, but not require, enforcing non-duplicate transport parameters (#2689,
#2691)
- Added an active_connection_id_limit transport parameter (#1994, #1998)
- max_ack_delay transport parameter defaults to 0 (#2638, #2646)
- When sending 0-RTT, only remembered transport parameters apply (#2458, #2360,
#2466, #2461)
- Define handshake completion and confirmation; define clearer rules when it
encryption keys should be discarded (#2214, #2267, #2673)
- Prohibit path migration prior to handshake confirmation (#2309, #2370)
- PATH_RESPONSE no longer needs to be received on the validated path (#2582,
#2580, #2579, #2637)
- PATH_RESPONSE frames are not stored and retransmitted (#2724, #2729)
- Document hack for enabling routing of ICMP when doing PMTU probing (#1243,
#2402)
## Since draft-ietf-quic-transport-19
- Refine discussion of 0-RTT transport parameters (#2467, #2464)
- Fewer transport parameters need to be remembered for 0-RTT (#2624, #2467)
- Spin bit text incorporated (#2564)
- Close the connection when maximum stream ID in MAX_STREAMS exceeds 2^62 - 1
(#2499, #2487)
- New connection ID required for intentional migration (#2414, #2413)
- Connection ID issuance can be rate-limited (#2436, #2428)
- The "QUIC bit" is ignored in Version Negotiation (#2400, #2561)
- Initial packets from clients need to be padded to 1200 unless a Handshake
packet is sent as well (#2522, #2523)
- CRYPTO frames can be discarded if too much data is buffered (#1834, #2524)
- Stateless reset uses a short header packet (#2599, #2600)
## Since draft-ietf-quic-transport-18
- Removed version negotiation; version negotiation, including authentication of
the result, will be addressed in the next version of QUIC (#1773, #2313)
- Added discussion of the use of IPv6 flow labels (#2348, #2399)
- A connection ID can't be retired in a packet that uses that connection ID
(#2101, #2420)
- Idle timeout transport parameter is in milliseconds (from seconds) (#2453,
#2454)
- Endpoints are required to use new connection IDs when they use new network
paths (#2413, #2414)
- Increased the set of permissible frames in 0-RTT (#2344, #2355)
## Since draft-ietf-quic-transport-17
- Stream-related errors now use STREAM_STATE_ERROR (#2305)
- Endpoints discard initial keys as soon as handshake keys are available (#1951,
#2045)
- Expanded conditions for ignoring ICMP packet too big messages (#2108, #2161)
- Remove rate control from PATH_CHALLENGE/PATH_RESPONSE (#2129, #2241)
- Endpoints are permitted to discard malformed initial packets (#2141)
- Clarified ECN implementation and usage requirements (#2156, #2201)
- Disable ECN count verification for packets that arrive out of order (#2198,
#2215)
- Use Probe Timeout (PTO) instead of RTO (#2206, #2238)
- Loosen constraints on retransmission of ACK ranges (#2199, #2245)
- Limit Retry and Version Negotiation to once per datagram (#2259, #2303)
- Set a maximum value for max_ack_delay transport parameter (#2282, #2301)
- Allow server preferred address for both IPv4 and IPv6 (#2122, #2296)
- Corrected requirements for migration to a preferred address (#2146, #2349)
- ACK of non-existent packet is illegal (#2298, #2302)
## Since draft-ietf-quic-transport-16
- Stream limits are defined as counts, not maximums (#1850, #1906)
- Require amplification attack defense after closing (#1905, #1911)
- Remove reservation of application error code 0 for STOPPING (#1804, #1922)
- Renumbered frames (#1945)
- Renumbered transport parameters (#1946)
- Numeric transport parameters are expressed as varints (#1608, #1947, #1955)
- Reorder the NEW_CONNECTION_ID frame (#1952, #1963)
- Rework the first byte (#2006)
- Fix the 0x40 bit
- Change type values for long header
- Add spin bit to short header (#631, #1988)
- Encrypt the remainder of the first byte (#1322)
- Move packet number length to first byte
- Move ODCIL to first byte of retry packets
- Simplify packet number protection (#1575)
- Allow STOP_SENDING to open a remote bidirectional stream (#1797, #2013)
- Added mitigation for off-path migration attacks (#1278, #1749, #2033)
- Don't let the PMTU to drop below 1280 (#2063, #2069)
- Require peers to replace retired connection IDs (#2085)
- Servers are required to ignore Version Negotiation packets (#2088)
- Tokens are repeated in all Initial packets (#2089)
- Clarified how PING frames are sent after loss (#2094)
- Initial keys are discarded once Handshake are available (#1951, #2045)
- ICMP PTB validation clarifications (#2161, #2109, #2108)
## Since draft-ietf-quic-transport-15
Substantial editorial reorganization; no technical changes.
## Since draft-ietf-quic-transport-14
- Merge ACK and ACK_ECN (#1778, #1801)
- Explicitly communicate max_ack_delay (#981, #1781)
- Validate original connection ID after Retry packets (#1710, #1486, #1793)
- Idle timeout is optional and has no specified maximum (#1765)
- Update connection ID handling; add RETIRE_CONNECTION_ID type (#1464, #1468,
#1483, #1484, #1486, #1495, #1729, #1742, #1799, #1821)
- Include a Token in all Initial packets (#1649, #1794)
- Prevent handshake deadlock (#1764, #1824)
## Since draft-ietf-quic-transport-13
- Streams open when higher-numbered streams of the same type open (#1342, #1549)
- Split initial stream flow control limit into 3 transport parameters (#1016,
#1542)
- All flow control transport parameters are optional (#1610)
- Removed UNSOLICITED_PATH_RESPONSE error code (#1265, #1539)
- Permit stateless reset in response to any packet (#1348, #1553)
- Recommended defense against stateless reset spoofing (#1386, #1554)
- Prevent infinite stateless reset exchanges (#1443, #1627)
- Forbid processing of the same packet number twice (#1405, #1624)
- Added a packet number decoding example (#1493)
- More precisely define idle timeout (#1429, #1614, #1652)
- Corrected format of Retry packet and prevented looping (#1492, #1451, #1448,
#1498)
- Permit 0-RTT after receiving Version Negotiation or Retry (#1507, #1514,
#1621)
- Permit Retry in response to 0-RTT (#1547, #1552)
- Looser verification of ECN counters to account for ACK loss (#1555, #1481,
#1565)
- Remove frame type field from APPLICATION_CLOSE (#1508, #1528)
## Since draft-ietf-quic-transport-12
- Changes to integration of the TLS handshake (#829, #1018, #1094, #1165, #1190,
#1233, #1242, #1252, #1450, #1458)
- The cryptographic handshake uses CRYPTO frames, not stream 0
- QUIC packet protection is used in place of TLS record protection
- Separate QUIC packet number spaces are used for the handshake
- Changed Retry to be independent of the cryptographic handshake
- Added NEW_TOKEN frame and Token fields to Initial packet
- Limit the use of HelloRetryRequest to address TLS needs (like key shares)
- Enable server to transition connections to a preferred address (#560, #1251,
#1373)
- Added ECN feedback mechanisms and handling; new ACK_ECN frame (#804, #805,
#1372)
- Changed rules and recommendations for use of new connection IDs (#1258, #1264,
#1276, #1280, #1419, #1452, #1453, #1465)
- Added a transport parameter to disable intentional connection migration
(#1271, #1447)
- Packets from different connection ID can't be coalesced (#1287, #1423)
- Fixed sampling method for packet number encryption; the length field in long
headers includes the packet number field in addition to the packet payload
(#1387, #1389)
- Stateless Reset is now symmetric and subject to size constraints (#466, #1346)
- Added frame type extension mechanism (#58, #1473)
## Since draft-ietf-quic-transport-11
- Enable server to transition connections to a preferred address (#560, #1251)
- Packet numbers are encrypted (#1174, #1043, #1048, #1034, #850, #990, #734,
#1317, #1267, #1079)
- Packet numbers use a variable-length encoding (#989, #1334)
- STREAM frames can now be empty (#1350)
## Since draft-ietf-quic-transport-10
- Swap payload length and packed number fields in long header (#1294)
- Clarified that CONNECTION_CLOSE is allowed in Handshake packet (#1274)
- Spin bit reserved (#1283)
- Coalescing multiple QUIC packets in a UDP datagram (#1262, #1285)
- A more complete connection migration (#1249)
- Refine opportunistic ACK defense text (#305, #1030, #1185)
- A Stateless Reset Token isn't mandatory (#818, #1191)
- Removed implicit stream opening (#896, #1193)
- An empty STREAM frame can be used to open a stream without sending data (#901,
#1194)
- Define stream counts in transport parameters rather than a maximum stream ID
(#1023, #1065)
- STOP_SENDING is now prohibited before streams are used (#1050)
- Recommend including ACK in Retry packets and allow PADDING (#1067, #882)
- Endpoints now become closing after an idle timeout (#1178, #1179)
- Remove implication that Version Negotiation is sent when a packet of the wrong
version is received (#1197)
## Since draft-ietf-quic-transport-09
- Added PATH_CHALLENGE and PATH_RESPONSE frames to replace PING with Data and
PONG frame. Changed ACK frame type from 0x0e to 0x0d. (#1091, #725, #1086)
- A server can now only send 3 packets without validating the client address
(#38, #1090)
- Delivery order of stream data is no longer strongly specified (#252, #1070)
- Rework of packet handling and version negotiation (#1038)
- Stream 0 is now exempt from flow control until the handshake completes (#1074,
#725, #825, #1082)
- Improved retransmission rules for all frame types: information is
retransmitted, not packets or frames (#463, #765, #1095, #1053)
- Added an error code for server busy signals (#1137)
- Endpoints now set the connection ID that their peer uses. Connection IDs are
variable length. Removed the omit_connection_id transport parameter and the
corresponding short header flag. (#1089, #1052, #1146, #821, #745, #821,
#1166, #1151)
## Since draft-ietf-quic-transport-08
- Clarified requirements for BLOCKED usage (#65, #924)
- BLOCKED frame now includes reason for blocking (#452, #924, #927, #928)
- GAP limitation in ACK Frame (#613)
- Improved PMTUD description (#614, #1036)
- Clarified stream state machine (#634, #662, #743, #894)
- Reserved versions don't need to be generated deterministically (#831, #931)
- You don't always need the draining period (#871)
- Stateless reset clarified as version-specific (#930, #986)
- initial_max_stream_id_x transport parameters are optional (#970, #971)
- Ack Delay assumes a default value during the handshake (#1007, #1009)
- Removed transport parameters from NewSessionTicket (#1015)
## Since draft-ietf-quic-transport-07
- The long header now has version before packet number (#926, #939)
- Rename and consolidate packet types (#846, #822, #847)
- Packet types are assigned new codepoints and the Connection ID Flag is
inverted (#426, #956)
- Removed type for Version Negotiation and use Version 0 (#963, #968)
- Streams are split into unidirectional and bidirectional (#643, #656, #720,
#872, #175, #885)
* Stream limits now have separate uni- and bi-directional transport parameters
(#909, #958)
* Stream limit transport parameters are now optional and default to 0 (#970,
#971)
- The stream state machine has been split into read and write (#634, #894)
- Employ variable-length integer encodings throughout (#595)
- Improvements to connection close
* Added distinct closing and draining states (#899, #871)
* Draining period can terminate early (#869, #870)
* Clarifications about stateless reset (#889, #890)
- Address validation for connection migration (#161, #732, #878)
- Clearly defined retransmission rules for BLOCKED (#452, #65, #924)
- negotiated_version is sent in server transport parameters (#710, #959)
- Increased the range over which packet numbers are randomized (#864, #850,
#964)
## Since draft-ietf-quic-transport-06
- Replaced FNV-1a with AES-GCM for all "Cleartext" packets (#554)
- Split error code space between application and transport (#485)
- Stateless reset token moved to end (#820)
- 1-RTT-protected long header types removed (#848)
- No acknowledgments during draining period (#852)
- Remove "application close" as a separate close type (#854)
- Remove timestamps from the ACK frame (#841)
- Require transport parameters to only appear once (#792)
## Since draft-ietf-quic-transport-05
- Stateless token is server-only (#726)
- Refactor section on connection termination (#733, #748, #328, #177)
- Limit size of Version Negotiation packet (#585)
- Clarify when and what to ack (#736)
- Renamed STREAM_ID_NEEDED to STREAM_ID_BLOCKED
- Clarify Keep-alive requirements (#729)
## Since draft-ietf-quic-transport-04
- Introduce STOP_SENDING frame, RESET_STREAM only resets in one direction (#165)
- Removed GOAWAY; application protocols are responsible for graceful shutdown
(#696)
- Reduced the number of error codes (#96, #177, #184, #211)
- Version validation fields can't move or change (#121)
- Removed versions from the transport parameters in a NewSessionTicket message
(#547)
- Clarify the meaning of "bytes in flight" (#550)
- Public reset is now stateless reset and not visible to the path (#215)
- Reordered bits and fields in STREAM frame (#620)
- Clarifications to the stream state machine (#572, #571)
- Increased the maximum length of the Largest Acknowledged field in ACK frames
to 64 bits (#629)
- truncate_connection_id is renamed to omit_connection_id (#659)
- CONNECTION_CLOSE terminates the connection like TCP RST (#330, #328)
- Update labels used in HKDF-Expand-Label to match TLS 1.3 (#642)
## Since draft-ietf-quic-transport-03
- Change STREAM and RESET_STREAM layout
- Add MAX_STREAM_ID settings
## Since draft-ietf-quic-transport-02
- The size of the initial packet payload has a fixed minimum (#267, #472)
- Define when Version Negotiation packets are ignored (#284, #294, #241, #143,
#474)
- The 64-bit FNV-1a algorithm is used for integrity protection of unprotected
packets (#167, #480, #481, #517)
- Rework initial packet types to change how the connection ID is chosen (#482,
#442, #493)
- No timestamps are forbidden in unprotected packets (#542, #429)
- Cryptographic handshake is now on stream 0 (#456)
- Remove congestion control exemption for cryptographic handshake (#248, #476)
- Version 1 of QUIC uses TLS; a new version is needed to use a different
handshake protocol (#516)
- STREAM frames have a reduced number of offset lengths (#543, #430)
- Split some frames into separate connection- and stream- level frames
(#443)
- WINDOW_UPDATE split into MAX_DATA and MAX_STREAM_DATA (#450)
- BLOCKED split to match WINDOW_UPDATE split (#454)
- Define STREAM_ID_NEEDED frame (#455)
- A NEW_CONNECTION_ID frame supports connection migration without linkability
(#232, #491, #496)
- Transport parameters for 0-RTT are retained from a previous connection (#405,
#513, #512)
- A client in 0-RTT no longer required to reset excess streams (#425, #479)
- Expanded security considerations (#440, #444, #445, #448)
## Since draft-ietf-quic-transport-01
- Defined short and long packet headers (#40, #148, #361)
- Defined a versioning scheme and stable fields (#51, #361)
- Define reserved version values for "greasing" negotiation (#112, #278)
- The initial packet number is randomized (#35, #283)
- Narrow the packet number encoding range requirement (#67, #286, #299, #323,
#356)
- Defined client address validation (#52, #118, #120, #275)
- Define transport parameters as a TLS extension (#49, #122)
- SCUP and COPT parameters are no longer valid (#116, #117)
- Transport parameters for 0-RTT are either remembered from before, or assume
default values (#126)
- The server chooses connection IDs in its final flight (#119, #349, #361)
- The server echoes the Connection ID and packet number fields when sending a
Version Negotiation packet (#133, #295, #244)
- Defined a minimum packet size for the initial handshake packet from the client
(#69, #136, #139, #164)
- Path MTU Discovery (#64, #106)
- The initial handshake packet from the client needs to fit in a single packet
(#338)
- Forbid acknowledgment of packets containing only ACK and PADDING (#291)
- Require that frames are processed when packets are acknowledged (#381, #341)
- Removed the STOP_WAITING frame (#66)
- Don't require retransmission of old timestamps for lost ACK frames (#308)
- Clarified that frames are not retransmitted, but the information in them can
be (#157, #298)
- Error handling definitions (#335)
- Split error codes into four sections (#74)
- Forbid the use of Public Reset where CONNECTION_CLOSE is possible (#289)
- Define packet protection rules (#336)
- Require that stream be entirely delivered or reset, including acknowledgment
of all STREAM frames or the RESET_STREAM, before it closes (#381)
- Remove stream reservation from state machine (#174, #280)
- Only stream 1 does not contribute to connection-level flow control (#204)
- Stream 1 counts towards the maximum concurrent stream limit (#201, #282)
- Remove connection-level flow control exclusion for some streams (except 1)
(#246)
- RESET_STREAM affects connection-level flow control (#162, #163)
- Flow control accounting uses the maximum data offset on each stream, rather
than bytes received (#378)
- Moved length-determining fields to the start of STREAM and ACK (#168, #277)
- Added the ability to pad between frames (#158, #276)
- Remove error code and reason phrase from GOAWAY (#352, #355)
- GOAWAY includes a final stream number for both directions (#347)
- Error codes for RESET_STREAM and CONNECTION_CLOSE are now at a consistent
offset (#249)
- Defined priority as the responsibility of the application protocol (#104,
#303)
## Since draft-ietf-quic-transport-00
- Replaced DIVERSIFICATION_NONCE flag with KEY_PHASE flag
- Defined versioning
- Reworked description of packet and frame layout
- Error code space is divided into regions for each component
- Use big endian for all numeric values
## Since draft-hamilton-quic-transport-protocol-01
- Adopted as base for draft-ietf-quic-tls
- Updated authors/editors list
- Added IANA Considerations section
- Moved Contributors and Acknowledgments to appendices
# Acknowledgments
{:numbered="false"}
Special thanks are due to the following for helping shape pre-IETF QUIC and its
deployment: Chris Bentzel, Misha Efimov, Roberto Peon, Alistair Riddoch,
Siddharth Vijayakrishnan, and Assar Westerlund.
This document has benefited immensely from various private discussions and
public ones on the quic@ietf.org and proto-quic@chromium.org mailing lists. Our
thanks to all.
# Contributors
{:numbered="false"}
The original authors of this specification were Ryan Hamilton, Jana Iyengar, Ian
Swett, and Alyssa Wilk.
The original design and rationale behind this protocol draw significantly from
work by Jim Roskind {{EARLY-DESIGN}}. In alphabetical order, the contributors to
the pre-IETF QUIC project at Google are: Britt Cyr, Jeremy Dorfman, Ryan
Hamilton, Jana Iyengar, Fedor Kouranov, Charles Krasic, Jo Kulik, Adam Langley,
Jim Roskind, Robbie Shade, Satyam Shekhar, Cherie Shi, Ian Swett, Raman Tenneti,
Victor Vasiliev, Antonio Vicente, Patrik Westin, Alyssa Wilk, Dale Worley, Fan
Yang, Dan Zhang, Daniel Ziegler.
