https://github.com/quicwg/base-drafts
Tip revision: 2ce761915d65dad597821ed4532317b2be32c667 authored by Martin Thomson on 18 December 2018, 23:40:49 UTC
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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: martin.thomson@gmail.com
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
{{QUIC-RECOVERY}} and the use of TLS for key negotiation {{QUIC-TLS}}.
--- 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:
: The smallest 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. An alternative view of QUIC streams is as an elastic "message"
abstraction.
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. Any stream can be
initiated by either endpoint. 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. Stream IDs are unique to a stream. A QUIC endpoint MUST NOT
reuse a stream ID within a connection. Stream IDs are encoded as
variable-length integers (see {{integer-encoding}}).
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 that is
visible to QUIC. STREAM frame boundaries are not expected to be preserved when
data is transmitted, when data is retransmitted after packet loss, or when data
is 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 frames 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.
# 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.
## Send 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 Send Streams"}
The sending part of stream that the endpoint initiates (types 0 and 2 for
clients, 1 and 3 for servers) is opened by the application. 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 send stream to
enter the "Send" state. An implementation might choose to defer allocating a
stream ID to a send stream until it sends the first frame and enters this state,
which can allow for better stream prioritization.
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 send 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 send 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 send 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 on a send stream; this
causes the send stream to open and then immediately transition to the "Reset
Sent" state.
Once a packet containing a RESET_STREAM has been acknowledged, the send stream
enters the "Reset Recvd" state, which is a terminal state.
## Receive 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 receive stream mirror only some of
the states of the send stream at the peer. A receive stream does not track
states on the send stream that cannot be observed, such as the "Ready" state.
Instead, receive streams track 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 Receive Streams"}
The receiving part of a stream initiated by a peer (types 1 and 3 for a client,
or 0 and 2 for a server) 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 a receive stream is "Recv".
The receive stream enters the "Recv" state when the sending part of a
bidirectional stream initiated by the endpoint (type 0 for a client, type 1 for
a server) enters the "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 creating a stream, 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 offset is known (see
{{final-offset}}). The receive stream enters the "Size Known" state. In this
state, the endpoint no longer needs to send MAX_STREAM_DATA frames, it only
receives any retransmissions of stream data.
Once all data for the stream has been received, the receive stream enters the
"Data Recvd" state. This might happen as a result of receiving the same STREAM
frame that causes the transition to "Size Known". In this state, the endpoint
has all stream data. Any STREAM or STREAM_DATA_BLOCKED frames it receives 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 immediately. Alternatively, the RESET_STREAM signal might be
suppressed or withheld if stream data is completely received and is buffered to
be read by the application. In the latter case, the receive stream transitions
from "Reset Recvd" to "Data Recvd".
Once the application has been delivered the signal indicating that the receive
stream was reset, the receive stream transitions to the "Reset Read" state,
which is a terminal state.
## 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 a send stream and a receive stream.
Implementations may represent states of the bidirectional stream as composites
of send and receive stream states. The simplest model presents the stream as
"open" when either send or receive stream is in a non-terminal state and
"closed" when both send and receive streams are in a terminal state.
{{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 send or receive streams
are mapped to the same composite state. Note that this is just one possibility
for such a mapping; this mapping requires that data is acknowledged before the
transition to a "closed" or "half-closed" state.
| Send Stream | Receive Stream | 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) |
| Data 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
receive stream is in the "Recv" state without yet having received any frames.
## Solicited State Transitions
If an endpoint is no longer interested in the data it is receiving on a stream,
it MAY send a STOP_SENDING frame identifying that stream to prompt closure of
the stream in the opposite direction. This typically indicates that the
receiving application is no longer reading data it receives from the stream, but
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 will 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 for that stream. An endpoint SHOULD copy the error code from the
STOP_SENDING frame, but MAY use any application error code. The endpoint
that sends a STOP_SENDING frame can ignore the error code carried in any
RESET_STREAM frame it receives.
If the STOP_SENDING frame is received on a send 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 receive stream that has not been
reset. STOP_SENDING is most useful for streams in the "Recv" or "Size Known"
states.
An endpoint is expected to send another STOP_SENDING frame if a packet
containing a previous STOP_SENDING is lost. However, once either all stream
data or a 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 at any time during the connection. A receiver cannot renege on an
advertisement however. That is, 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 STREAM_DATA_BLOCKED or
DATA_BLOCKED frames to indicate it has data to write but is blocked by flow
control limits. These frames are expected to be sent infrequently in common
cases, but they are considered useful for debugging and monitoring purposes.
A sender sends a single STREAM_DATA_BLOCKED or DATA_BLOCKED frame only once when
it reaches a data limit. 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 MAY 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.
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. If a
RESET_STREAM frame is reordered with stream data for the same stream, the
receiver's estimate of the number of bytes received on that stream can be lower
than the sender's estimate of the number sent. As a result, 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 offset 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 offset 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 Offset {#final-offset}
The final offset is the count of the number of bytes that are transmitted on a
stream. For a stream that is reset, the final offset is carried explicitly in a
RESET_STREAM frame. Otherwise, the final offset is the offset of the end of the
data carried in a STREAM frame marked with a FIN flag, or 0 in the case of
incoming unidirectional streams.
An endpoint will know the final offset for a stream when the receive 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 offset.
Once a final offset for a stream is known, it cannot change. If a RESET_STREAM
or STREAM frame is received indicating a change in the final offset for the
stream, an endpoint SHOULD respond with a FINAL_OFFSET_ERROR error (see
{{error-handling}}). A receiver SHOULD treat receipt of data at or beyond the
final offset as a FINAL_OFFSET_ERROR error, even after a stream is closed.
Generating these errors is not mandatory, but only because requiring that an
endpoint generate these errors also means that the endpoint needs to maintain
the final offset state for closed streams, which could mean a significant state
commitment.
## 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.
Endpoints MUST NOT exceed the limit set by their peer. An endpoint that
receives a STREAM frame with a stream ID exceeding the limit it has sent MUST
treat this as a stream error of type STREAM_LIMIT_ERROR ({{error-handling}}).
A receiver cannot renege on an advertisement. That is, 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 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}}).
Endpoints store received connection IDs for future use. An endpoint that
receives excessive connection IDs MAY discard those it cannot store without
sending a RETIRE_CONNECTION_ID frame. An endpoint that issues connection IDs
cannot expect its peer to store and use all issued connection IDs.
An endpoint SHOULD ensure that its peer has a sufficient number of available and
unused connection IDs. While each endpoint independently chooses how many
connection IDs to issue, endpoints SHOULD provide and maintain at least eight
connection IDs. The endpoint SHOULD do this by always supplying a new
connection ID when a connection ID is retired by its peer or when the endpoint
receives a packet with a previously unused connection ID. Endpoints that
initiate migration and require non-zero-length connection IDs SHOULD provide
their peers with new connection IDs before migration, or risk the peer closing
the connection.
### 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 won't 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.
## 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 MUST drop
packets with zero-length Destination Connection ID fields if they do not
correspond to a single connection.
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, but for which the endpoint
cannot remove packet protection, are discarded.
### 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, clients might receive packets for a connection
that are encrypted with a key it has not yet computed. Clients 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.
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.
Servers MUST drop other packets that contain unsupported versions.
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 forbidden
from sending 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
-->
# 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 first few messages of an exchange between a client attempting to create a
new connection with server is shown in {{fig-vn}}. After version negotiation
completes, connection establishment can proceed, for example as shown in
{{example-handshake-flows}}.
~~~~
Client Server
Packet (v=X) ->
<- Version Negotiation (supported=Y,Z)
Packet (v=Y) ->
<- Packet(s) (v=Y)
~~~~
{: #fig-vn title="Example Version Negotiation Exchange"}
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.
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 the client receives a Version Negotiation packet, it first checks 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 connection using that version. Though the
content of the Initial packet the client sends might not change in response to
version negotiation, a client MUST increase the packet number it uses on every
packet it sends. Packets MUST continue to use long headers ({{long-header}})
and MUST include the new negotiated protocol version.
The client MUST use the long header format and include its selected version on
all packets until it has 1-RTT keys and it has received a packet from the server
which is not a Version Negotiation packet.
A client MUST NOT change the version it uses unless it is in response to a
Version Negotiation packet from the server. Once a client receives a packet
from the server which is not a Version Negotiation packet, it MUST discard other
Version Negotiation packets on the same connection. Similarly, a client MUST
ignore a Version Negotiation packet if it has already received and acted on a
Version Negotiation packet.
A client MUST ignore a Version Negotiation packet that lists the client's chosen
version.
A client MAY attempt 0-RTT after receiving a Version Negotiation packet. A
client that sends additional 0-RTT packets MUST NOT reset the packet number to 0
as a result, see {{retry-0rtt-pn}}.
Version negotiation packets have no cryptographic protection. The result of the
negotiation MUST be revalidated as part of the cryptographic handshake (see
{{version-validation}}).
## Using Reserved Versions
For a server to use a new version in the future, clients must correctly handle
unsupported versions. To help ensure this, a server SHOULD include a reserved
version (see {{versions}}) while generating a Version Negotiation packet.
The design of version negotiation permits a server to avoid maintaining state
for packets that it rejects in this fashion. The validation of version
negotiation (see {{version-validation}}) only validates the result of version
negotiation, which is the same no matter which reserved version was sent.
A server MAY therefore send different reserved version numbers in the Version
Negotiation Packet and in its transport parameters.
A client MAY send a packet using a reserved version number. This can be used to
solicit a list of supported versions from a server.
# 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 the transport parameters of the peer (see
{{transport-parameters}})
* authenticated confirmation of version negotiation (see {{version-validation}})
* 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-verification}}.
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.
## 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 version negotiation and 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[55, "..."], 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, CERT, CV, FIN]
<- 1-RTT[0]: STREAM[1, "..."] ACK[0]
Initial[1]: ACK[0]
Handshake[0]: CRYPTO[FIN], ACK[0]
1-RTT[2]: STREAM[0, "..."] ACK[0] ->
1-RTT[1]: STREAM[55, "..."], ACK[1,2]
<- 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 which has not previously received a
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 final version used for a connection might be different from the version of
the first Initial from the client. To enable consistent routing through the
handshake, a client SHOULD select an initial Destination Connection ID length
long enough to fulfill the minimum size for every QUIC version it supports.
The client populates the Source Connection ID field with a value of its choosing
and sets the SCIL field to match.
The Destination Connection ID field in the server's Initial packet contains a
connection ID that is chosen by the recipient of the packet (i.e., the client);
the Source Connection ID includes the connection ID that the sender of the
packet wishes to use (see {{connection-id}}). The server MUST use consistent
Source Connection IDs during the handshake.
On 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. 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. In
particular, version negotiation MUST be validated (see {{version-validation}})
before the connection establishment is considered properly complete.
Definitions for each of the defined transport parameters are included in
{{transport-parameter-definitions}}. Any given parameter MUST appear at most
once in a given transport parameters extension. An endpoint MUST treat receipt
of duplicate transport parameters as a connection error of type
TRANSPORT_PARAMETER_ERROR.
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}
A client that attempts to send 0-RTT data MUST remember the transport parameters
used by the server. The transport parameters that the server advertises during
connection establishment apply to all connections that are resumed using the
keying material established during that handshake. Remembered transport
parameters apply to the new connection until the handshake completes and new
transport parameters from the server can be provided.
A server can remember the transport parameters that it advertised, or store an
integrity-protected copy of the values in the ticket and recover the information
when accepting 0-RTT data. A server uses the transport parameters in
determining whether to accept 0-RTT data.
A server MAY accept 0-RTT and subsequently provide different values for
transport parameters for use in the new connection. If 0-RTT data is accepted
by the server, the server MUST NOT reduce any limits or alter any values that
might be violated by the client with its 0-RTT data. In particular, a server
that accepts 0-RTT data MUST NOT set values for the following parameters
({{transport-parameter-definitions}}) that are smaller
than the remembered value of those 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.
The value of the server's previous preferred_address MUST NOT be used when
establishing a new connection; rather, the client should wait to observe the
server's new preferred_address value in the handshake.
A server MUST either reject 0-RTT data or abort a handshake if the implied
values for transport parameters cannot be supported.
### New Transport Parameters
New transport parameters can be used to negotiate new protocol behavior. An
endpoint MUST ignore transport parameters that it does not support. Absence of
a transport parameter therefore disables any optional protocol feature that is
negotiated using the parameter.
New transport parameters can be registered according to the rules in
{{iana-transport-parameters}}.
### Version Negotiation Validation {#version-validation}
Though the cryptographic handshake has integrity protection, two forms of QUIC
version downgrade are possible. In the first, an attacker replaces the QUIC
version in the Initial packet. In the second, a fake Version Negotiation packet
is sent by an attacker. To protect against these attacks, the transport
parameters include three fields that encode version information. These
parameters are used to retroactively authenticate the choice of version (see
{{version-negotiation}}).
The cryptographic handshake provides integrity protection for the negotiated
version as part of the transport parameters (see
{{transport-parameter-definitions}}). As a result, attacks on version
negotiation by an attacker can be detected.
The client includes the initial_version field in its transport parameters. The
initial_version is the version that the client initially attempted to use. If
the server did not send a Version Negotiation packet {{packet-version}}, this
will be identical to the negotiated_version field in the server transport
parameters.
A server that processes all packets in a stateful fashion can remember how
version negotiation was performed and validate the initial_version value.
A server that does not maintain state for every packet it receives (i.e., a
stateless server) uses a different process. If the initial_version matches the
version of QUIC that is in use, a stateless server can accept the value.
If the initial_version is different from the version of QUIC that is in use, a
stateless server MUST check that it would have sent a Version Negotiation packet
if it had received a packet with the indicated initial_version. If a server
would have accepted the version included in the initial_version and the value
differs from the QUIC version that is in use, the server MUST terminate the
connection with a VERSION_NEGOTIATION_ERROR error.
The server includes both the version of QUIC that is in use and a list of the
QUIC versions that the server supports (see
{{transport-parameter-definitions}}).
The negotiated_version field is the version that is in use. This MUST be set by
the server to the value that is on the Initial packet that it accepts (not an
Initial packet that triggers a Retry or Version Negotiation packet). A client
that receives a negotiated_version that does not match the version of QUIC that
is in use MUST terminate the connection with a VERSION_NEGOTIATION_ERROR error
code.
The server includes a list of versions that it would send in any version
negotiation packet ({{packet-version}}) in the supported_versions field. The
server populates this field even if it did not send a version negotiation
packet.
The client validates that the negotiated_version is included in the
supported_versions list and - if version negotiation was performed - that it
would have selected the negotiated version. A client MUST terminate the
connection with a VERSION_NEGOTIATION_ERROR error code if the current QUIC
version is not listed in the supported_versions list. A client MUST terminate
with a VERSION_NEGOTIATION_ERROR error code if version negotiation occurred but
it would have selected a different version based on the value of the
supported_versions list.
When an endpoint accepts multiple QUIC versions, it can potentially interpret
transport parameters as they are defined by any of the QUIC versions it
supports. The version field in the QUIC packet header is authenticated using
transport parameters. The position and the format of the version fields in
transport parameters MUST either be identical across different QUIC versions, or
be unambiguously different to ensure no confusion about their interpretation.
One way that a new format could be introduced is to define a TLS extension with
a different codepoint.
# 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 pad UDP datagrams that contain only Initial packets to at least
1200 bytes. Once a client has received an acknowledgment for a Handshake packet
it MAY send smaller datagrams. Sending padded datagrams ensures that the server
is not overly constrained by the amplification restriction.
In order to prevent a handshake deadlock as a result of the server being unable
to send, clients SHOULD send a packet upon a handshake 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}}).
### 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 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[0]: CRYPTO[CH] ->
Initial[0]: CRYPTO[SH] ACK[0]
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 token. The client MUST NOT use the token provided in a Retry
for future connections. Servers MAY discard any Initial packet that does not
carry the expected token.
Unlike the token that is created for a Retry packet, there might be some time
between when the token is created and when the token is subsequently used.
Thus, a resumption token SHOULD include an expiration time. The server MAY
include either an explicit expiration time or an issued timestamp and
dynamically calculate the expiration time. It is also unlikely that the client
port number is the same on two different connections; validating the port is
therefore unlikely to be successful.
A resumption token SHOULD be constructed to be easily distinguishable from
tokens that are sent in Retry packets as they are carried in the same field.
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 can include that value in the
Token field of its Initial packet.
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. Tokens obtained in Retry packets MUST NOT be discarded.
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 it migrates
prior to completing the handshake.
When a server receives an Initial packet with an address validation token, it
SHOULD attempt to validate it, 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 MAY 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 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 can be both sent to and received 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. 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 fresh random data in every PATH_CHALLENGE frame so that it
can associate the peer's response with the causative 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.
However, because a PATH_CHALLENGE might be sent from a spoofed address, an
endpoint MUST limit the rate at which it sends PATH_RESPONSE frames and MAY
silently discard PATH_CHALLENGE frames that would cause it to respond at a
higher rate.
To ensure that packets can be both sent to and received from the peer, the
PATH_RESPONSE MUST be sent on the same path as the triggering PATH_CHALLENGE.
That is, from the same local address on which the PATH_CHALLENGE was received,
to the same remote address from which the PATH_CHALLENGE was received.
## Successful Path Validation
A new address is considered valid when a PATH_RESPONSE frame is received
containing 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.
For path validation to be successful, a PATH_RESPONSE frame MUST be received
from the same remote address to which the corresponding PATH_CHALLENGE was
sent. If a PATH_RESPONSE frame is received from a different remote address than
the one to which the PATH_CHALLENGE was sent, path validation is considered to
have failed, even if the data matches that sent in the PATH_CHALLENGE.
Additionally, the PATH_RESPONSE frame MUST be received on the same local address
from which the corresponding PATH_CHALLENGE was sent. An endpoint considers the
path to be valid when a PATH_RESPONSE frame is received on the same path with
the same payload as the PATH_CHALLENGE frame.
If a PATH_RESPONSE frame is received on a different local address than the one
from which the PATH_CHALLENGE was sent, path validation is not considered to be
successful, even if the data matches the PATH_CHALLENGE. This doesn't result in
path validation failure, as it might be a result of a forwarded packet (see
{{off-path-forward}}) or misrouting.
## 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 current
Retransmittion Timeout (RTO) as defined in {{QUIC-RECOVERY}} is RECOMMENDED.
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 (that is, IP address and/or 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.
An endpoint MUST NOT initiate connection migration before the handshake is
finished and the endpoint has 1-RTT keys. The design of QUIC relies on
endpoints retaining a stable address for the duration of the handshake.
An endpoint also MUST NOT initiate connection migration if the peer sent the
`disable_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 MAY treat this as a connection error of type
INVALID_MIGRATION.
Not all changes of peer address are intentional 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.
NAT rebinding is not connection migration as defined in this section, though an
endpoint SHOULD perform path validation ({{migrate-validate}}) if it detects a
change in the IP address of its peer.
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 SHOULD
immediately reset the congestion controller and round-trip time estimator for
the new path.
An endpoint MUST NOT return to the send rate used for the previous path unless
it is reasonably sure that the previous send rate is valid for the new path.
For instance, a change in the client's port number is likely indicative of a
rebinding in a middlebox and not a complete change in path. This determination
likely depends on heuristics, which could be imperfect; if the new path capacity
is significantly reduced, ultimately this relies on the congestion controller
responding to congestion signals and reducing send rates appropriately.
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.
This eliminates the use of the connection ID for linking activity from
the same connection on different networks. Protection of packet numbers 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.
Clients MAY move to a new connection ID at any time based on
implementation-specific concerns. For example, after a period of network
inactivity NAT rebinding might occur when the client begins sending data again.
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.
Endpoints that use connection IDs with length greater than zero could have their
activity correlated if their peers keep using the same destination connection ID
after migration. Endpoints that receive packets with a previously unused
Destination Connection ID SHOULD change to sending packets with a connection ID
that has not been used on any other network path. The goal here is to ensure
that packets sent on different paths cannot be correlated. To fulfill this
privacy requirement, endpoints that initiate migration and use connection IDs
with length greater than zero SHOULD provide their peers with new connection IDs
before migration.
Caution:
: If both endpoints change connection ID in response to seeing a change in
connection ID from their peer, then this can trigger an infinite sequence of
changes.
## 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.
Once the handshake is finished, the client SHOULD initiate path validation (see
{{migrate-validate}}) of the server's preferred 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 MAY send other non-probing frames from its preferred address, but MUST
continue sending all probing packets from its original IP address.
The server SHOULD also initiate path validation of the client using its
preferred address and the address from which it received the client probe. 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.
# Connection Termination {#termination}
Connections should remain open until they become idle for a pre-negotiated
period of time. A QUIC connection, once established, can be terminated in one
of three ways:
* idle timeout ({{idle-timeout}})
* immediate close ({{immediate-close}})
* stateless reset ({{stateless-reset}})
## 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 three times the current Retransmission Timeout
(RTO) 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 connection ID and QUIC version is sufficient information to identify packets
for a closing connection; an endpoint can discard all other connection state.
An endpoint MAY retain packet protection keys for incoming packets to allow it
to read and process a 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
can confirm that its peer is also closing or draining. Receiving a
CONNECTION_CLOSE frame is sufficient confirmation, as is receiving a stateless
reset. The draining period SHOULD end when the closing period would have ended.
In other words, the endpoint can use the same end time, but cease retransmission
of the closing packet.
Disposing of connection state prior to the end of the closing or draining period
could cause delayed or reordered packets to be handled poorly. Endpoints that
have some alternative means to ensure that late-arriving packets on the
connection do not create QUIC state, such as those that are able to close the
UDP socket, MAY use an abbreviated draining period which can allow for faster
resource recovery. Servers that retain an open socket for accepting new
connections SHOULD NOT exit the closing or draining period early.
Once the closing or draining period has ended, an endpoint SHOULD discard all
connection state. This results in new packets on the connection being handled
generically. For instance, an endpoint MAY send a stateless reset in response
to any further incoming packets.
The draining and closing periods do not apply when a stateless reset
({{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.
An endpoint could receive packets from a new source address, indicating a client
connection migration ({{migration}}), while in the closing period. 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
If the idle timeout is enabled, a connection that remains idle for longer than
the advertised idle timeout (see {{transport-parameter-definitions}}) is closed.
A connection enters the draining state when the idle timeout expires.
Each endpoint advertises its own idle timeout to its peer. The idle timeout
starts from the last packet received. In order to ensure that initiating new
activity postpones an idle timeout, an endpoint restarts this timer when sending
a packet. An endpoint does not postpone the idle timeout if another packet has
been sent containing frames other than ACK or PADDING, and that other packet has
not been acknowledged or declared lost. Packets that contain only ACK or
PADDING frames are not acknowledged until an endpoint has other frames to send,
so they could prevent the timeout from being refreshed.
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 an RTO
(see Section 5.3.3 of {{QUIC-RECOVERY}}), it is advisable to test for liveness
before sending any data that cannot be retried safely.
## 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 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 an CONNECTION_CLOSE frame with an appropriate error code to
signal closure.
If the connection has been successfully established, endpoints MUST send any
CONNECTION_CLOSE frames in a 1-RTT packet. Prior to connection establishment a
peer might not have 1-RTT keys, so endpoints SHOULD send CONNECTION_CLOSE frames
in a Handshake packet. If the endpoint does not have Handshake keys, or it is
not certain that the peer has Handshake keys, it MAY send CONNECTION_CLOSE
frames in an Initial packet. If multiple packets are sent, they can be
coalesced (see {{packet-coalesce}}) to facilitate retransmission.
## 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. 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| Random Bits (182..) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ +
| |
+ 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 random bytes following it are set to unpredictable values. The last
16 bytes of the datagram contain a Stateless Reset Token.
A stateless reset will be interpreted by a recipient as a packet with a short
header. For the packet to appear as valid, the Random Bits field needs to
include at least 182 bits of random or unpredictable values (or 24 bytes, less
the two fixed bits). This is intended to allow for a destination connection ID
of the maximum length permitted, with a minimal packet number, and payload. The
Stateless Reset Token corresponds to the minimum expansion of the packet
protection AEAD. More random bytes might be necessary if the endpoint could
have negotiated a packet protection scheme with a larger minimum AEAD expansion.
An endpoint SHOULD NOT send a stateless reset that is significantly larger than
the packet it receives. 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.
An endpoint MAY send a stateless reset in response to a packet with a long
header. This would not be effective if the stateless reset token was not yet
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 more effective.
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.
Finally, the last 16 bytes of the packet are set to the value of the Stateless
Reset Token.
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 a packet with a short
header either cannot be decrypted or is marked as a duplicate packet. The
endpoint then compares the last 16 bytes of the packet with the Stateless Reset
Token provided by its peer, either in a NEW_CONNECTION_ID frame or the server's
transport parameters. If these values are identical, 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
cannot occur 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 cannot be reused for new
connections at nodes that share a static key.
Note that Stateless Reset packets do not have any cryptographic protection.
### Looping {#reset-looping}
The design of a Stateless Reset is such that it is indistinguishable from a
valid packet. This means that a Stateless Reset might trigger the sending of a
Stateless Reset in response, which could lead to infinite exchanges.
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 the recommended minimum size of 37
bytes could mean that the packet could reveal to an observer that it is a
Stateless Reset. 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.
An endpoint can increase the odds that a packet will trigger a Stateless Reset
if it cannot be processed by padding it to at least 38 bytes.
# 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.
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 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 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, either directly or
through the receipt of a STOP_SENDING frame from a peer. RESET_STREAM carries
an application error code. Resetting a stream without knowledge of the
application protocol could cause the protocol to enter an unrecoverable state.
Application protocols might require certain streams to be reliably delivered in
order to guarantee consistent state between endpoints.
# Packets and Frames {#packets-frames}
QUIC endpoints communicate by exchanging packets. Packets are carried in UDP
datagrams (see {{packet-coalesce}}) and have confidentiality and integrity
protection (see {{packet-protected}}).
This version of QUIC uses the long packet header (see {{long-header}}) during
connection establishment and the short header (see {{short-header}}) once 1-RTT
keys have been established.
Packets that carry the long header are Initial {{packet-initial}}, Retry
{{packet-retry}}, Handshake {{packet-handshake}}, and 0-RTT Protected packets
{{packet-protected}}.
Packets with the short header are designed for minimal overhead and are used
after a connection is established.
Version negotiation uses a packet with a special format (see
{{packet-version}}).
## Protected Packets {#packet-protected}
All QUIC packets except Version Negotiation and Retry packets use authenticated
encryption with additional data (AEAD) {{!RFC5119}} 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, it 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, see {{packet-numbers}} for details.
## Coalescing Packets {#packet-coalesce}
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 starting sending data. 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 will always be the last packet included in a UDP datagram.
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) or the
packet is of an unknown type, 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 cannot be followed by
other packets in the same UDP datagram.
## Packet Numbers {#packet-numbers}
The packet number is an integer in the range 0 to 2^62-1. Where present, packet
numbers are encoded as a variable-length integer (see {{integer-encoding}}).
This number is used in determining the cryptographic nonce for packet
protection. Each endpoint maintains a separate packet number for sending and
receiving.
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. Each packet number space starts 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 (that is, under the same cryptographic keys). If the
packet number for sending reaches 2^62 - 1, the sender MUST close the connection
without sending a CONNECTION_CLOSE frame or any further packets; 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, commonly consists
of a sequence of frames, as shown in {{packet-frames}}. Version Negotiation,
Stateless Reset, and Retry packets do not contain frames.
<!-- TODO: Editorial work needed in this section. Not all packets 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"}
QUIC payloads MUST contain at least one frame, and MAY contain multiple frames
and multiple frame types.
Frames MUST fit within a single QUIC packet and MUST NOT span a QUIC packet
boundary. Each frame begins with a Frame Type, 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 STREAM 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"}
All QUIC frames are idempotent. 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 0x4007 is a legitimate two-byte
encoding for a variable-length integer with a value of 7, PING frames are always
encoded as a single byte with the value 0x07. 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 and Acknowledgment {#processing-and-ack}
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? -->
### Sending ACK Frames
<!-- TODO: Re-read this section for flow and redundancy. -->
To avoid creating an indefinite feedback loop, an endpoint MUST NOT send an ACK
frame in response to a packet containing only ACK or PADDING frames, even if
there are packet gaps which precede the received packet. The endpoint MUST
however acknowledge packets containing only ACK or PADDING frames when sending
ACK frames in response to other packets.
Packets containing PADDING frames are considered
to be in flight for congestion control purposes {{QUIC-RECOVERY}}. Sending only
PADDING frames might cause the sender to become limited by the congestion
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 MUST NOT send more than one packet containing only an ACK frame per
received packet that contains frames other than ACK and PADDING frames.
The 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.
Strategies and implications of the frequency of generating acknowledgments are
discussed in more detail in {{QUIC-RECOVERY}}.
To limit ACK Blocks to those that have not yet been received by the sender, the
receiver SHOULD track which ACK frames have been acknowledged by its peer. Once
an ACK frame has been acknowledged, the packets it acknowledges SHOULD NOT be
acknowledged again.
Because ACK frames are not sent in response to ACK-only packets, a receiver that
is only sending ACK frames will only receive acknowledgements for its packets
if the sender includes them in packets with non-ACK frames. A sender SHOULD
bundle ACK frames with other frames when possible.
To limit receiver state or the size of ACK frames, a receiver MAY limit the
number of ACK Blocks it sends. A receiver can do this even without receiving
acknowledgment of its ACK frames, with the knowledge this could cause the sender
to unnecessarily retransmit some data. Standard QUIC {{QUIC-RECOVERY}}
algorithms declare packets lost after sufficiently newer packets are
acknowledged. Therefore, the receiver SHOULD repeatedly acknowledge newly
received packets in preference to packets received in the past.
### 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.
Endpoints SHOULD send acknowledgments for packets containing CRYPTO frames with
a reduced delay; see Section 5.3.1 of {{QUIC-RECOVERY}}.
## 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
send 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 receive 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 receive 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 streams 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.
A new PATH_CHALLENGE frame will be sent if another PATH_RESPONSE frame is
needed.
* 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.
* 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}}).
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 verifies the path, both during connection
establishment and when migrating to a new path (see {{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 {{processing-and-ack}} 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 {{processing-and-ack}} 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 Verification {#ecn-verification}
Each endpoint independently verifies and enables use of ECN by setting the IP
header ECN codepoint to ECN Capable Transport (ECT) for the path from it to the
other peer. Even if not setting ECN codepoints on packets it transmits, the
endpoint SHOULD provide feedback about ECN markings received (if accessible).
To verify both that a path supports ECN and the peer can provide ECN feedback,
an endpoint sets the ECT(0) codepoint in the IP header of all outgoing
packets {{!RFC8311}}.
If an ECT codepoint set in the IP header is not corrupted by a network device,
then a received packet contains either the codepoint sent by the peer or the
Congestion Experienced (CE) codepoint set by a network device that is
experiencing congestion.
If a QUIC packet sent with an ECT codepoint is newly acknowledged by the peer in
an ACK frame without ECN feedback, the endpoint stops setting ECT codepoints in
subsequent IP packets, with the expectation that either the network path or the
peer no longer supports ECN.
Network devices that corrupt or apply non-standard ECN markings might result in
reduced throughput or other undesirable side-effects. To reduce this risk, an
endpoint uses the following steps to verify the counts it receives in an ACK
frame. Counts MUST NOT be verified if the ACK frame does not increase the
largest received packet number at the endpoint.
* 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).
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 verification succeeds, the local reference counts MUST be
increased to match the counts in the ACK frame.
Upon successful verification, an endpoint continues to set ECT codepoints in
subsequent packets with the expectation that the path is ECN-capable.
If verification fails, then the endpoint ceases setting ECT codepoints in
subsequent IP packets with the expectation that either the network path or the
peer does not support ECN.
If an endpoint sets ECT codepoints on outgoing IP packets and encounters a
retransmission timeout due to the absence of acknowledgments from the peer (see
{{QUIC-RECOVERY}}), or if an endpoint has reason to believe that an element on
the network path might be corrupting ECN codepoints, the endpoint MAY cease
setting ECT codepoints in subsequent packets. Doing so allows the connection to
be resilient to network elements that corrupt ECN codepoints in the IP header or
drop packets with ECT or CE codepoints in the IP header.
# 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 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
combining the Initial packet with a 0-RTT 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.
# 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 in long and short packet headers 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 Packet {#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|R R|P P|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Version (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|DCIL(4)|SCIL(4)|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Connection ID (0/32..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Source Connection ID (0/32..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Length (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Packet Number (8/16/24/32) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Payload (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~~~
{: #fig-long-header title="Long Header Packet Format"}
Long headers are used for packets that are sent prior to the completion of
version negotiation and establishment of 1-RTT keys. Once both conditions are
met, a sender switches to sending packets using the short header
({{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}}.
Reserved Bits (R):
: The next two bits (those with a mask of 0x0c) 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 protection as a connection error of type PROTOCOL_VIOLATION.
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}}).
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.
DCIL and SCIL:
: The byte following the version contains the lengths of the two connection ID
fields that follow it. These lengths are encoded as two 4-bit unsigned
integers. The Destination Connection ID Length (DCIL) field occupies the 4
high bits of the byte and the Source Connection ID Length (SCIL) field
occupies the 4 low bits of the byte. An encoded length of 0 indicates that
the connection ID is also 0 bytes in length. Non-zero encoded lengths are
increased by 3 to get the full length of the connection ID, producing a length
between 4 and 18 bytes inclusive. For example, an byte with the value 0x50
describes an 8-byte Destination Connection ID and a zero-length Source
Connection ID.
Destination Connection ID:
: The Destination Connection ID field follows the connection ID lengths and is
either 0 bytes in length or between 4 and 18 bytes.
{{negotiating-connection-ids}} describes the use of this field in more detail.
Source Connection ID:
: The Source Connection ID field follows the Destination Connection ID and is
either 0 bytes in length or between 4 and 18 bytes.
{{negotiating-connection-ids}} describes the use of this field in more detail.
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 plaintext packet number. See {{packet-encoding}} for details.
Payload:
: The payload of the packet.
The following packet types are defined:
| Type | Name | Section |
|-----:|:------------------------------|:----------------------------|
| 0x0 | Initial | {{packet-initial}} |
| 0x1 | 0-RTT Protected | {{packet-protected}} |
| 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, plus the Length and
Packet Number fields 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. Type-specific semantics for this version are described in the
following sections.
The end of the packet is determined by the Length field. The Length field
covers 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. The Length field
enables packet coalescing ({{packet-coalesce}}).
## Short Header Packet {#short-header}
~~~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+
|0|1|S|R|R|K|P P|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Connection ID (0..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 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 sixth bit (0x20) of byte 0 is the Latency Spin Bit, set as described in
{{!SPIN=I-D.ietf-quic-spin-exp}}.
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 protection as a connection error of type
PROTOCOL_VIOLATION.
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.
## Version Negotiation Packet {#packet-version}
A Version Negotiation packet is inherently not version-specific, and does not
use the long packet header (see {{long-header}}). Upon receipt by a client, it
will appear to be a packet using the long header, but will be identified as a
Version Negotiation packet based on the Version field 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) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|DCIL(4)|SCIL(4)|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Connection ID (0/32..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Source Connection ID (0/32..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 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.
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.
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.
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.
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.
An Initial packet (shown in {{initial-format}}) has two additional header fields
that are added to the Long Header before the Length field.
~~~
+-+-+-+-+-+-+-+-+
|1|1| 0 |R R|P P|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Version (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|DCIL(4)|SCIL(4)|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Connection ID (0/32..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Source Connection ID (0/32..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Token Length (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Token (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Length (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Packet Number (8/16/24/32) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Payload (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #initial-format title="Initial Packet"}
These fields include the token that was previously provided in a Retry packet or
NEW_TOKEN frame:
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.
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 Version Negotiation ({{packet-version}}) or
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 4.10 of {{QUIC-TLS}}) along with any loss recovery and congestion
control state (see Sections 5.3.1.2 and 6.9 of {{QUIC-RECOVERY}}).
Any data in CRYPTO frames is discarded - and no longer retransmitted - when
Initial keys are discarded.
### Starting Packet Numbers
<!-- TODO: delete this section after confirming that it is redundant -->
The first Initial packet sent by either endpoint contains a packet number of
0. The packet number MUST increase monotonically thereafter. Initial packets
are in a different packet number space to other packets (see
{{packet-numbers}}).
### 0-RTT Packet Numbers {#retry-0rtt-pn}
<!-- TODO: Thus far, we have not really addressed 0-RTT at all. We don't even
really say much about what it is for. This really belongs in a section for that,
probably right after the example handshake exchanges. -->
Packet numbers for 0-RTT protected packets use the same space as 1-RTT protected
packets.
After a client receives a Retry or Version Negotiation packet, 0-RTT packets are
likely to have been lost or discarded by the server. A client MAY 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. The keys
used to protect 0-RTT packets will not change as a result of responding to a
Retry or Version Negotiation packet unless the client also regenerates the
cryptographic handshake message. 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.
Receiving a Retry or Version Negotiation packet, especially a Retry that changes
the connection ID used for subsequent packets, indicates a strong possibility
that 0-RTT packets could be lost. 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 SHOULD instead generate a fresh cryptographic handshake message and
start packet numbers from 0. This ensures that new 0-RTT packets will not use
the same keys, avoiding any risk of key and nonce reuse; this also prevents
0-RTT packets from previous handshake attempts from being accepted as part of
the connection.
## Handshake Packet {#packet-handshake}
A Handshake packet uses long headers with a type value of 0x2. It is
used to carry acknowledgments and cryptographic handshake messages from the
server and client.
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}}).
The first Handshake packet sent by a server contains a packet number of 0.
Handshake packets are their own packet number space. Packet numbers are
incremented normally for other Handshake packets.
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 stateless 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 |ODCIL(4|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Version (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|DCIL(4)|SCIL(4)|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Connection ID (0/32..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Source Connection ID (0/32..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Original Destination Connection ID (0/32..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Retry Token (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #retry-format title="Retry Packet"}
A Retry packet (shown in {{retry-format}}) only uses the invariant portion of
the long packet header {{QUIC-INVARIANTS}}; that is, the fields up to and
including the Destination and Source Connection ID fields. A Retry packet does
not contain any protected fields. Like Version Negotiation, a Retry packet
contains the long header including the connection IDs, but omits the Length,
Packet Number, and Payload fields. These are replaced with:
ODCIL:
: The four least-significant bits of the first byte of a Retry packet are not
protected as they are for other packets with the long header, because Retry
packets don't contain a protected payload. These bits instead encode the
length of the Original Destination Connection ID field. The length uses the
same encoding as the DCIL and SCIL fields.
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 ODCIL.
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 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
{{validate-future}}).
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 can either reuse the cryptographic handshake
message or construct a new one at its discretion.
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 that sends
additional 0-RTT packets without constructing a new cryptographic handshake
message MUST NOT reset the packet number to 0 after a Retry packet, see
{{retry-0rtt-pn}}.
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 validates that the
original_connection_id transport parameter is present and correct; otherwise, it
validates 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.
# 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}}.
~~~
uint32 QuicVersion;
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_migration(12),
preferred_address(13),
(65535)
} TransportParameterId;
struct {
TransportParameterId parameter;
opaque value<0..2^16-1>;
} TransportParameter;
struct {
select (Handshake.msg_type) {
case client_hello:
QuicVersion initial_version;
case encrypted_extensions:
QuicVersion negotiated_version;
QuicVersion supported_versions<4..2^8-4>;
};
TransportParameter parameters<0..2^16-1>;
} TransportParameters;
~~~
{: #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.
## 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 seconds that is encoded as an integer. 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 is only sent by a server.
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 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 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). The default value is also used for ACK frames
that are sent in Initial and Handshake packets. 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.
disable_migration (0x000c):
: The disable migration transport parameter is included if the endpoint does not
support 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 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-preffered-address}}. This transport parameter is only sent by a server.
~~~
struct {
enum { IPv4(4), IPv6(6), (15) } ipVersion;
opaque ipAddress<4..2^8-1>;
uint16 port;
opaque connectionId<0..18>;
opaque statelessResetToken[16];
} PreferredAddress;
~~~
{: #fig-preffered-address title="Preferred Address format"}
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 Blocks.
ACK Blocks are ranges of 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 Block Count (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ACK Blocks (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [ECN Section] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #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 including the time in microseconds that the largest
acknowledged packet, as indicated in the Largest Acknowledged field, was
received by this peer to when this ACK was sent. The value of the ACK Delay
field is scaled by multiplying the encoded value by 2 to the power of the
value of the `ack_delay_exponent` transport parameter set by the sender of the
ACK frame. The `ack_delay_exponent` defaults to 3, or a multiplier of 8 (see
{{transport-parameter-definitions}}). Scaling in this fashion allows for a
larger range of values with a shorter encoding at the cost of lower
resolution.
ACK Block Count:
: A variable-length integer specifying the number of Additional ACK Block (and
Gap) fields after the First ACK Block.
ACK Blocks:
: Contains one or more blocks of packet numbers which have been successfully
received, see {{ack-block-section}}.
### ACK Block Section {#ack-block-section}
The ACK Block Section consists of alternating Gap and ACK Block fields in
descending packet number order. A First Ack Block field is followed by a
variable number of alternating Gap and Additional ACK Blocks. The number of
Gap and Additional ACK Block fields is determined by the ACK Block Count field.
Gap and ACK Block fields use a relative integer encoding for efficiency. Though
each encoded value is positive, the values are subtracted, so that each ACK
Block describes progressively lower-numbered packets. As long as contiguous
ranges of packets are small, the variable-length integer encoding ensures that
each range can be expressed in a small number of bytes.
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.
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| First ACK Block (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Gap (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Additional ACK Block (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Gap (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Additional ACK Block (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Gap (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Additional ACK Block (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #ack-block-format title="ACK Block Section"}
Each ACK Block acknowledges a contiguous range of packets by indicating the
number of acknowledged packets that precede the largest packet number in that
block. A value of zero indicates that only the largest packet number is
acknowledged. Larger ACK Block values indicate a larger range, with
corresponding lower values for the smallest packet number in the range. Thus,
given a largest packet number for the ACK, the smallest value is determined by
the formula:
~~~
smallest = largest - ack_block
~~~
The range of packets that are acknowledged by the ACK Block include the range
from the smallest packet number to the largest, inclusive.
The largest value for the First ACK Block is determined by the Largest
Acknowledged field; the largest for Additional ACK Blocks is determined by
cumulatively subtracting the size of all preceding ACK Blocks and Gaps.
Each Gap indicates a range of packets that are not being acknowledged. The
number of packets in the gap is one higher than the encoded value of the Gap
Field.
The value of the Gap field establishes the largest packet number value for the
ACK Block that follows the gap using the following formula:
~~~
largest = previous_smallest - gap - 2
~~~
If the calculated value for largest or smallest packet number for any ACK Block
is negative, an endpoint MUST generate a connection error of type
FRAME_ENCODING_ERROR indicating an error in an ACK frame.
The fields in the ACK Block Section are:
First ACK Block:
: A variable-length integer indicating the number of contiguous packets
preceding the Largest Acknowledged that are being acknowledged.
Gap (repeated):
: A variable-length integer indicating the number of contiguous unacknowledged
packets preceding the packet number one lower than the smallest in the
preceding ACK Block.
Additional ACK Block (repeated):
: A variable-length integer indicating the number of contiguous acknowledged
packets preceding the largest packet number, as determined by the
preceding Gap.
### ECN section
The ECN section should only be parsed when the ACK frame type is 0x03. The ECN
section consists of 3 ECN counts as shown below.
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ECT(0) Count (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ECT(1) Count (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ECN-CE Count (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
ECT(0) Count:
: A variable-length integer representing the total number packets received with
the ECT(0) codepoint.
ECT(1) Count:
: A variable-length integer representing the total number packets received with
the ECT(1) codepoint.
CE Count:
: A variable-length integer representing the total number packets received with
the CE codepoint.
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 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 PROTOCOL_VIOLATION.
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 (16) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Final Offset (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 16-bit application protocol error code (see {{app-error-codes}}) which
indicates why the stream is being closed.
Final Offset:
: A variable-length integer indicating the absolute byte offset of the end of
data written on this stream by the RESET_STREAM sender.
## 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. This signals a peer
to abruptly terminate transmission on a stream.
Receipt of a STOP_SENDING frame is invalid for a locally-initiated stream that
has not yet been created or is in the "Ready" state (see
{{stream-send-states}}). Receiving a STOP_SENDING frame for a locally-initiated
send stream that is "Ready" or not yet created MUST be treated as a connection
error of type PROTOCOL_VIOLATION. An endpoint that receives a STOP_SENDING
frame for a receive-only stream MUST terminate the connection with error
PROTOCOL_VIOLATION.
The STOP_SENDING frame is as follows:
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Application Error Code (16) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
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 16-bit, 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. 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 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.
## 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 determine the fields that
are present in the frame.
* The OFF bit (0x04) in the frame type is set to indicate that there is an
Offset field present. When set to 1, the Offset field is present. When set
to 0, the Offset field is absent and the Stream Data starts at an offset of 0
(that is, the frame contains the first bytes of the stream, or the end of a
stream that includes no data).
* The LEN bit (0x02) in the frame type is set to indicate that there is a Length
field present. If this bit is set to 0, the Length field is absent and the
Stream Data field extends to the end of the packet. If this bit is set to 1,
the Length field is present.
* The FIN bit (0x01) of the frame type is set only on frames that contain the
final offset of the stream. Setting this bit indicates that the frame
marks the end of the stream.
An endpoint that receives a STREAM frame for a send-only stream MUST terminate
the connection with error PROTOCOL_VIOLATION.
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 re-constructed offset and data length - MUST be
less than 2^62.
## 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.
An endpoint that receives a MAX_STREAM_DATA frame for a receive-only stream
MUST terminate the connection with error PROTOCOL_VIOLATION.
An endpoint that receives a MAX_STREAM_DATA frame for a send-only stream
it has not opened MUST terminate the connection with error PROTOCOL_VIOLATION.
Note that an endpoint may legally receive a MAX_STREAM_DATA frame on a
bidirectional stream it has not opened.
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 PROTOCOL_VIOLATION.
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) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Length (8) | |
+-+-+-+-+-+-+-+-+ Connection ID (32..144) +
| ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ +
| |
+ 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}}.
Length:
: An 8-bit unsigned integer containing the length of the connection ID. Values
less than 4 and greater than 18 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.
## 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.
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 (8) +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
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 (16) | [ Frame Type (i) ] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Reason Phrase Length (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Reason Phrase (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
CONNECTION_CLOSE frames contain the following fields:
Error Code:
: A 16-bit 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.
Note that a CONNECTION_CLOSE frame cannot be split between packets, so in
practice any limits on packet size will also limit the space available for a
reason phrase.
Reason Phrase:
: A human-readable explanation for why the connection was closed. This can be
zero length if the sender chooses to not give details beyond the Error Code.
This SHOULD be a UTF-8 encoded string {{!RFC3629}}.
## 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 16-bit unsigned integers.
This section lists the defined QUIC transport error codes that may be used in a
CONNECTION_CLOSE frame. These errors apply to the entire connection.
NO_ERROR (0x0):
: An endpoint uses this with CONNECTION_CLOSE to signal that the connection is
being closed abruptly in the absence of any error.
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_OFFSET_ERROR (0x6):
: An endpoint received a STREAM frame containing data that exceeded the
previously established final offset. Or an endpoint received a RESET_STREAM
frame containing a final offset that was lower than the maximum offset of data
that was already received. Or an endpoint received a RESET_STREAM frame
containing a different final offset 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.
VERSION_NEGOTIATION_ERROR (0x9):
: An endpoint received transport parameters that contained version negotiation
parameters that disagreed with the version negotiation that it performed.
This error code indicates a potential version downgrade attack.
PROTOCOL_VIOLATION (0xA):
: An endpoint detected an error with protocol compliance that was not covered by
more specific error codes.
INVALID_MIGRATION (0xC):
: A peer has migrated to a different network when the endpoint had disabled
migration.
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.
## Application Protocol Error Codes {#app-error-codes}
Application protocol error codes are 16-bit unsigned integers, but the
management of application error codes are left to application protocols.
Application protocol error codes are used for the RESET_STREAM frame
({{frame-reset-stream}}) 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, though additional validation is
required for version negotiation (see {{version-validation}}).
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, transmission
error may cause STREAM DATA frames opening streams to be received out of
sequence. A receiver is obligated to open intervening streams if a
higher-numbered stream ID is received. Thus, on a new connection, opening
stream 2000001 opens 1 million streams, as required by the specification.
The number of active streams is limited by the concurrent stream limit transport
parameter, as explained in {{controlling-concurrency}}. If chosen judiciously,
this limit mitigates the effect of the stream commitment attack. However,
setting the limit too low could affect performance when applications expect to
open large number of streams.
## 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 receivers ignore
ECN codepoints set in duplicate packets (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.
# 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_migration | {{transport-parameter-definitions}} |
| 0x000d | preferred_address | {{transport-parameter-definitions}} |
{: #iana-tp-table title="Initial QUIC Transport Parameters Entries"}
## 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 16-bit space. This space is
split into two spaces that are governed by different policies. Values with the
first byte in the range 0x00 to 0xfe (in hexadecimal) are assigned via the
Specification Required policy {{!RFC8126}}. Values with the first byte 0xff are
reserved for Private Use {{!RFC8126}}.
Registrations MUST include the following fields:
Value:
: The numeric value of the assignment (registrations will be between 0x0000 and
0xfeff).
Code:
: A short mnemonic for the parameter.
Description:
: A brief description of the error code semantics, which MAY be a summary if a
specification reference is provided.
Specification:
: A reference to a publicly available specification for the value.
The initial contents of this registry are shown in {{iana-error-table}}. Values
from 0xFF00 to 0xFFFF are reserved for Private Use {{!RFC8126}}.
| 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_OFFSET_ERROR | Change to final stream offset | {{error-codes}} |
| 0x7 | FRAME_ENCODING_ERROR | Frame encoding error | {{error-codes}} |
| 0x8 | TRANSPORT_PARAMETER_ERROR | Error in transport parameters | {{error-codes}} |
| 0x9 | VERSION_NEGOTIATION_ERROR | Version negotiation failure | {{error-codes}} |
| 0xA | PROTOCOL_VIOLATION | Generic protocol violation | {{error-codes}} |
| 0xC | INVALID_MIGRATION | Violated disabled migration | {{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-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
- 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 avaialble (#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.
