https://github.com/quicwg/base-drafts
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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. This document
describes connection establishment, packet format, multiplexing and reliability.
Accompanying documents describe the cryptographic handshake and loss detection.
--- 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 transport protocol that runs on top of UDP.
QUIC aims to provide a flexible set of features that allow it to be a
general-purpose secure transport for multiple applications.
* Version negotiation
* Low-latency connection establishment
* Authenticated and encrypted header and payload
* Stream multiplexing
* Stream and connection-level flow control
* Connection migration and resilience to NAT rebinding
QUIC implements techniques learned from experience with TCP, SCTP and other
transport protocols. QUIC uses UDP as substrate so as to not require changes to
legacy client operating systems and middleboxes to be deployable. QUIC
authenticates all of its headers and encrypts most of the data it exchanges,
including its signaling. This allows the protocol to evolve without incurring a
dependency on upgrades to middleboxes. This document describes the core QUIC
protocol, including the conceptual design, wire format, and mechanisms of the
QUIC protocol for connection establishment, stream multiplexing, stream and
connection-level flow control, connection migration, and data reliability.
Accompanying documents describe QUIC's loss detection and congestion control
{{QUIC-RECOVERY}}, and the use of TLS 1.3 for key negotiation {{QUIC-TLS}}.
QUIC version 1 conforms to the protocol invariants in {{QUIC-INVARIANTS}}.
# Conventions and Definitions
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD",
"SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in BCP 14 {{!RFC2119}} {{!RFC8174}}
when, and only when, they appear in all capitals, as shown here.
Definitions of terms that are used in this document:
Client:
: The endpoint initiating a QUIC connection.
Server:
: The endpoint accepting incoming QUIC connections.
Endpoint:
: The client or server end of a connection.
Stream:
: A logical, bi-directional channel of ordered bytes within a QUIC connection.
Connection:
: A conversation between two QUIC endpoints with a single encryption context
that multiplexes streams within it.
Connection ID:
: An opaque identifier that is used to identify a QUIC connection at an
endpoint. Each endpoint sets a value that its peer includes in packets.
QUIC packet:
: A well-formed UDP payload that can be parsed by a QUIC receiver.
QUIC is a name, not an acronym.
## Notational Conventions
Packet and frame diagrams 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
# 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 octets 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\>.
# Packet Types and Formats
We first describe QUIC's packet types and their formats, since some are
referenced in subsequent mechanisms.
All numeric values are encoded in network byte order (that is, big-endian) and
all field sizes are in bits. When discussing individual bits of fields, the
least significant bit is referred to as bit 0. Hexadecimal notation is used for
describing the value of fields.
Any QUIC packet has either a long or a short header, as indicated by the Header
Form bit. Long headers are expected to be used early in the connection before
version negotiation and establishment of 1-RTT keys. Short headers are minimal
version-specific headers, which are used after version negotiation and 1-RTT
keys are established.
## Long Header {#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| Type (7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Version (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|DCIL(4)|SCIL(4)|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Connection ID (0/32..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Source Connection ID (0/32..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Length (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Packet Number (8/16/32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 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 octet 0 (the first octet) is set to 1 for
long headers.
Long Packet Type:
: The remaining seven bits of octet 0 contain the packet type. This field can
indicate one of 128 packet types. The types specified for this version are
listed in {{long-packet-types}}.
Version:
: The QUIC Version is a 32-bit field that follows the Type. 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 octet 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 octet and the Source Connection ID Length (SCIL) field
occupies the 4 low bits of the octet. An encoded length of 0 indicates that
the connection ID is also 0 octets 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 octets inclusive. For example, an octet with the value 0x50
describes an 8-octet 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 octets in length or between 4 and 18 octets.
{{connection-id-encoding}} 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 octets in length or between 4 and 18 octets.
{{connection-id-encoding}} 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 octets, encoded as a variable-length integer
({{integer-encoding}}).
Packet Number:
: The packet number field is 1, 2, or 4 octets long. The packet number has
confidentiality protection separate from packet protection, as described
in Section 5.3 of {{QUIC-TLS}}. The length of the packet number field is
encoded in the plaintext packet number. See {{packet-numbers}} for details.
Payload:
: The payload of the packet.
The following packet types are defined:
| Type | Name | Section |
|:-----|:------------------------------|:----------------------------|
| 0x7F | Initial | {{packet-initial}} |
| 0x7E | Retry | {{packet-retry}} |
| 0x7D | Handshake | {{packet-handshake}} |
| 0x7C | 0-RTT Protected | {{packet-protected}} |
{: #long-packet-types title="Long Header Packet Types"}
The header form, type, connection ID lengths octet, destination and source
connection IDs, and version fields of a long header packet are
version-independent. The packet number and values for packet types defined in
{{long-packet-types}} 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 the both the Packet Number and Payload fields, both of which are
confidentiality protected and initially of unknown length. The size of the
Payload field is learned once the packet number protection is removed.
Senders can sometimes coalesce multiple packets into one UDP datagram. See
{{packet-coalesce}} for more details.
## 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|K|1|1|0|R R R|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Connection ID (0..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Packet Number (8/16/32) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 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 octet 0 is set to 0 for the short header.
Key Phase Bit:
: The second bit (0x40) of octet 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.
\[\[Editor's Note: this section should be removed and the bit definitions
changed before this draft goes to the IESG.]]
Third Bit:
: The third bit (0x20) of octet 0 is set to 1.
\[\[Editor's Note: this section should be removed and the bit definitions
changed before this draft goes to the IESG.]]
Fourth Bit:
: The fourth bit (0x10) of octet 0 is set to 1.
\[\[Editor's Note: this section should be removed and the bit definitions
changed before this draft goes to the IESG.]]
Google QUIC Demultipexing Bit:
: The fifth bit (0x8) of octet 0 is set to 0. This allows implementations of
Google QUIC to distinguish Google QUIC packets from short header packets sent
by a client because Google QUIC servers expect the connection ID to always be
present.
The special interpretation of this bit SHOULD be removed from this
specification when Google QUIC has finished transitioning to the new header
format.
Reserved:
: The sixth, seventh, and eighth bits (0x7) of octet 0 are reserved for
experimentation. Endpoints MUST ignore these bits on packets they receive
unless they are participating in an experiment that uses these bits. An
endpoint not actively using these bits SHOULD set the value randomly on
packets they send to protect against unwanted inference about particular
values.
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, 2, or 4 octets long. The packet number has
confidentiality protection separate from packet protection, as described in
Section 5.3 of {{QUIC-TLS}}. The length of the packet number field is encoded
in the plaintext packet number. See {{packet-numbers}} for details.
Protected Payload:
: Packets with a short header always include a 1-RTT protected payload.
The header form and 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.
## Retry Packet {#packet-retry}
A Retry packet uses a long packet header with a type value of 0x7E. It carries
an address validation token created by the server. It is used by a server that
wishes to perform a stateless retry (see {{stateless-retry}}).
~~~
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| 0x7e |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Version (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|DCIL(4)|SCIL(4)|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Connection ID (0/32..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Source Connection ID (0/32..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ODCIL(8) | Original Destination Connection ID (*) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 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. The contents of the
Retry packet are not protected. 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 length of the Original Destination Connection ID field. The length is
encoded in the least significant 4 bits of the octet, using the same encoding
as the DCIL and SCIL fields. The most significant 4 bits of this octet are
reserved. Unless a use for these bits has been negotiated, endpoints SHOULD
send randomized values and MUST ignore any value that it receives.
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.
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. The client MUST use this connection ID in the Destination Connection ID
of subsequent packets that it sends.
A Retry packet does not include a packet number and cannot be explicitly
acknowledged by a client.
A server MUST NOT send a Retry in response to packets other than Initial
or 0-RTT packets. A server MAY choose to only send Retry in response to Initial
packets and discard or buffer 0-RTT packets corresponding to unvalidated client
addresses.
If the Original Destination Connection ID field does not match the Destination
Connection ID from the most recent Initial packet it sent, clients MUST discard
the 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 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 MUST NOT reset the packet number to 0 after a Retry
packet, see {{retry-0rtt-pn}}.
A server that might send another Retry packet in response to a subsequent
Initial packet MUST set the Source Connection ID to a new value of at least 8
octets in length. This allows clients to distinguish between Retry packets when
the server sends multiple rounds of Retry packets. Consequently, a valid Retry
packet will always have an Original Destination Connection ID that is at least 8
octets long; clients MUST discard Retry packets that include a shorter value. A
server that will not send additional Retry packets can set the Source Connection
ID to any value.
## Cryptographic Handshake Packets {#handshake-packets}
Once version negotiation is 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.
All these packets use the long header and contain the current QUIC version in
the version field.
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.
## Initial Packet {#packet-initial}
The Initial packet uses long headers with a type value of 0x7F. It carries the
first CRYPTO frames sent by the client and server to perform key exchange, and
carries ACKs in either direction. The Initial packet is protected by Initial
keys as described in {{QUIC-TLS}}.
The Initial packet (shown in {{initial-format}}) has two additional header
fields that are added to the Long Header before the Length field.
~~~
+-+-+-+-+-+-+-+-+
|1| 0x7f |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 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 (8/16/32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 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 frames are also
permitted. The first CRYPTO frame sent always begins at an offset of 0 (see
{{handshake}}).
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}}).
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.
### Connection IDs
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 octets in length. Until a
packet is received from the server, the client MUST use the same value unless it
abandons the connection attempt and starts a new one. The initial Destination
Connection ID is used to determine packet protection keys for Initial packets.
The client populates the Source Connection ID field with a value of its choosing
and sets the 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. Once a client has received an Initial packet from the
server, it MUST discard any packet it receives with a different Source
Connection ID.
### Tokens
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 connections.
Specifically, 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. 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.
If the client received a Retry packet from the server and sends an Initial
packet in response, then it sets the Destination Connection ID 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.
When a server receives an Initial packet with an address validation token, it
SHOULD attempt to validate it. 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 (see {{stateless-retry}}).
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.
### Starting Packet Numbers
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}
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 MAY 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.
### Minimum Packet Size
The payload of a UDP datagram carrying the Initial packet MUST be expanded to at
least 1200 octets (see {{packetization}}), by adding PADDING frames to the
Initial packet and/or by combining the Initial packet with a 0-RTT packet (see
{{packet-coalesce}}).
## Handshake Packet {#packet-handshake}
A Handshake packet uses long headers with a type value of 0x7D. It is
used to carry acknowledgments and cryptographic handshake messages from the
server and client.
A server sends its cryptographic handshake in one or more Handshake packets in
response to an Initial packet if it does not send a Retry packet. Once a client
has received a Handshake packet from a server, it uses Handshake packets to send
subsequent cryptographic handshake messages and acknowledgments to the server.
The Destination Connection ID field in a Handshake packet contains a connection
ID that is chosen by the recipient of the packet; the Source Connection ID
includes the connection ID that the sender of the packet wishes to use (see
{{connection-id-encoding}}).
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.
Servers MUST NOT send more than three datagrams including Initial and Handshake
packets without receiving a packet from a verified source address. Source
addresses can be verified through an address validation token
(delivered via a Retry packet or a NEW_TOKEN frame) or by receiving
any message from the client encrypted using the Handshake keys.
The payload of this packet contains CRYPTO frames and could contain PADDING, or
ACK frames. Handshake packets MAY contain CONNECTION_CLOSE frames if the
handshake is unsuccessful.
## Protected Packets {#packet-protected}
All QUIC packets use packet protection. Packets that are protected with the
static handshake keys or the 0-RTT keys are sent with long headers; all packets
protected with 1-RTT keys are sent with short headers. The different packet
types explicitly indicate the encryption level and therefore the keys that are
used to remove packet protection. 0-RTT and 1-RTT protected packets share a
single packet number space.
Packets protected with handshake keys only use packet protection to ensure that
the sender of the packet is on the network path. This packet protection is not
effective confidentiality protection; 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.
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.
Packets protected with 0-RTT keys use a type value of 0x7C. The connection ID
fields for a 0-RTT packet MUST match the values used in the Initial packet
({{packet-initial}}).
The client can send 0-RTT packets after receiving an Initial
{{packet-initial}} or Handshake ({{packet-handshake}}) packet, if that
packet does not complete the handshake. Even if the client receives a
different connection ID in the Handshake packet, it MUST continue to
use the same Destination Connection ID for 0-RTT packets, see
{{connection-id-encoding}}.
The version field for protected packets is the current QUIC version.
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.
The payload is protected using authenticated encryption. {{QUIC-TLS}} describes
packet protection in detail. After decryption, the plaintext consists of a
sequence of frames, as described in {{frames}}.
## Coalescing Packets {#packet-coalesce}
A sender can coalesce multiple QUIC packets (typically a Cryptographic Handshake
packet and a Protected packet) into one UDP datagram. This can reduce the
number of UDP datagrams needed to send application data during the handshake and
immediately afterwards. It is not necessary for senders to coalesce
packets, though failing to do so will require sending a significantly
larger number of datagrams during the handshake. Receivers MUST
be able to process coalesced packets.
Senders SHOULD coalesce packets in order of increasing encryption levels
(Initial, Handshake, 0-RTT, 1-RTT), as this 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 with different Destination Connection
IDs into a single UDP datagram. Receivers SHOULD ignore any subsequent packets
with a different Destination Connection ID than the first packet in the
datagram.
Every QUIC packet that is coalesced into a single UDP datagram is separate and
complete. 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. If one or more
packets in a datagram cannot be processed yet (because the keys are not yet
available) or processing fails (decryption failure, unknown type, etc.), the
receiver MUST still attempt to process the remaining packets. The skipped
packets MAY either be discarded or buffered for later processing, just as if the
packets were received out-of-order in separate datagrams.
Retry ({{packet-retry}}) and Version Negotiation ({{packet-version}}) packets
cannot be coalesced.
## Connection ID Encoding
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 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.
During the handshake, an endpoint might receive multiple packets with the long
header, and thus be given multiple opportunities to update the Destination
Connection ID it sends. 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.
Short headers 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 connection-ID based load balancer could agree with the load
balancer on a fixed or minimum length and on an encoding for connection IDs.
This 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.
The very first packet sent by a client includes a random value for Destination
Connection ID. The same value MUST be used for all 0-RTT packets sent on that
connection ({{packet-protected}}). This randomized value is used to determine
the packet protection keys for Initial packets (see Section 5.1.1 of
{{QUIC-TLS}}).
A Version Negotiation ({{packet-version}}) packet MUST use both connection IDs
selected by the client, swapped to ensure correct routing toward the client.
The connection ID can change over the lifetime of a connection, especially in
response to connection migration ({{migration}}). NEW_CONNECTION_ID frames
({{frame-new-connection-id}}) are used to provide new connection ID values.
## Packet Numbers {#packet-numbers}
The packet number is an integer in the range 0 to 2^62-1. The value is used in
determining the cryptographic nonce for packet encryption. Each endpoint
maintains a separate packet number for sending and receiving.
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 encryption keys.
Conceptually, a packet number space is the encryption context in which
a packet can be processed and ACKed. Initial packets can only be sent
with Initial encryption keys and ACKed in packets which are also
Initial packets. Similarly, Handshake packets can only be sent and
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.
In the QUIC long and short packet headers, the number of bits required to
represent the packet number is reduced by including only a variable number of
the least significant bits of the packet number. One or two of the most
significant bits of the first octet determine how many bits of the packet
number are provided, as shown in {{pn-encodings}}.
| First octet pattern | Encoded Length | Bits Present |
|:--------------------|:---------------|:-------------|
| 0b0xxxxxxx | 1 octet | 7 |
| 0b10xxxxxx | 2 | 14 |
| 0b11xxxxxx | 4 | 30 |
{: #pn-encodings title="Packet Number Encodings for Packet Headers"}
Note that these encodings are similar to those in {{integer-encoding}}, but
use different values.
The encoded packet number is protected as described in Section 5.3
{{QUIC-TLS}}. Protection of the packet number is removed prior to recovering
the full packet number. The full packet number is reconstructed at the
receiver based on the number of significant bits present, the content 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 packet number 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
0xaa82f30e, then a packet containing a 14-bit value of 0x9b3 will be decoded as
0xaa8309b3.
Example pseudo-code for packet number decoding can be found in
{{sample-packet-number-decoding}}.
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 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 0x6afa2f,
sending a packet with a number of 0x6b2d79 requires a packet number encoding
with 14 bits or more; whereas the 30-bit packet number encoding is needed to
send a packet with a number of 0x6bc107.
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
{{?RFC2406}}.
A Version Negotiation packet ({{packet-version}}) does not include a packet
number. The Retry packet ({{packet-retry}}) has special rules for populating
the packet number field.
# Frames and Frame Types {#frames}
The payload of all packets, after removing packet protection, consists of a
sequence of frames, as shown in {{packet-frames}}. Version Negotiation and
Stateless Reset do not contain frames.
~~~
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Frame 1 (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Frame 2 (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Frame N (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #packet-frames title="Contents of Protected Payload"}
Protected 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 as they are referenced later in the
document.
| Type Value | Frame Type Name | Definition |
|:------------|:------------------|:----------------------------|
| 0x00 | PADDING | {{frame-padding}} |
| 0x01 | RST_STREAM | {{frame-rst-stream}} |
| 0x02 | CONNECTION_CLOSE | {{frame-connection-close}} |
| 0x03 | APPLICATION_CLOSE | {{frame-application-close}} |
| 0x04 | MAX_DATA | {{frame-max-data}} |
| 0x05 | MAX_STREAM_DATA | {{frame-max-stream-data}} |
| 0x06 | MAX_STREAM_ID | {{frame-max-stream-id}} |
| 0x07 | PING | {{frame-ping}} |
| 0x08 | BLOCKED | {{frame-blocked}} |
| 0x09 | STREAM_BLOCKED | {{frame-stream-blocked}} |
| 0x0a | STREAM_ID_BLOCKED | {{frame-stream-id-blocked}} |
| 0x0b | NEW_CONNECTION_ID | {{frame-new-connection-id}} |
| 0x0c | STOP_SENDING | {{frame-stop-sending}} |
| 0x0d | ACK | {{frame-ack}} |
| 0x0e | PATH_CHALLENGE | {{frame-path-challenge}} |
| 0x0f | PATH_RESPONSE | {{frame-path-response}} |
| 0x10 - 0x17 | STREAM | {{frame-stream}} |
| 0x18 | CRYPTO | {{frame-crypto}} |
| 0x19 | NEW_TOKEN | {{frame-new-token}} |
| 0x1a | ACK_ECN | {{frame-ack-ecn}} |
{: #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-octet encoding of the frame types
defined in this document is possible, the Frame Type field for these frames are
encoded on a single octet. For instance, though 0x4007 is a legitimate
two-octet encoding for a variable-length integer with a value of 7, PING frames
are always encoded as a single octet with the value 0x07. An endpoint MUST
treat the receipt of a frame type that uses a longer encoding than necessary as
a connection error of type PROTOCOL_VIOLATION.
## 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.
An IANA registry is used to manage the assignment of frame types, see
{{iana-frames}}.
# Life of a Connection
A QUIC connection is a single conversation between two QUIC endpoints. QUIC's
connection establishment intertwines 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, due to NAT rebinding or mobility, as
described in {{migration}}. Finally a connection may be terminated by either
endpoint, as described in {{termination}}.
## Connection ID
Each connection possesses a set of identifiers, any of which could be used to
distinguish it from other connections. A connection ID can be either 0 octets
in length, or between 4 and 18 octets (inclusive). Connection IDs are selected
independently in each direction.
The primary function of a connection ID is to ensure that changes in addressing
at lower protocol layers (UDP, IP, and below) 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.
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 associate them to
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 will issue a
series of connection IDs over the lifetime of a connection. The series of
connection IDs issued by an endpoint is ordered, with the final connection ID
selected during the handshake coming first. Additional connection IDs are
provided using the NEW_CONNECTION_ID frame ({{frame-new-connection-id}}), each
with a specified sequence number. The series of connection IDs issued SHOULD be
contiguous, but might not appear to be upon receipt due to reordering or loss.
Each connection ID MUST be used on only one local address. When packets are sent
for the first time on a new local address, a new connection ID MUST be used with
a higher sequence number than any connection ID previously used on any local
address. At any time, an endpoint MAY change to a new connection ID on a local
address already in use.
An endpoint MUST NOT send packets with a connection ID which has a lower
sequence number than the highest sequence number of any connection ID ever sent
or received on that local address. This ensures that when an endpoint migrates
to a new path or changes connection ID on an existing path, the packets will use
a new connection ID in both directions.
Implementations SHOULD ensure that peers have a connection ID with a matching
sequence number available when changing to a new connection ID. An
implementation could do this by always supplying a corresponding connection ID
to a peer for each connection ID received from that peer.
While endpoints select connection IDs as appropriate for their implementation,
the connection ID MUST NOT include the unprotected sequence number. Endpoints
need to be able to recover the sequence number associated with each connection
ID they generate without relying on information available to unaffiliated
parties. A connection ID that encodes an unencrypted sequence number could be
used to correlate connection IDs across network paths.
## 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.
### Client Packet Handling {#client-pkt-handling}
Valid packets sent to clients always include a Destination Connection ID that
matches the 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 MAY be 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 and
sufficient length to be an Initial packet for some 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 content 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 as described in {{packet-handling}}. If not matched, 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 a
Handshake 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. They
SHOULD send a Stateless Reset ({{stateless-reset}}) if a connection ID
is present in the header.
## Version Negotiation
Version negotiation ensures that client and server agree to a QUIC version
that is mutually supported. A server sends a Version Negotiation packet in
response to each packet that might initiate a new connection, see
{{packet-handling}} for details.
The size of the first packet sent by a client will determine whether a server
sends a Version Negotiation packet. Clients that support multiple QUIC
versions SHOULD pad their Initial packets to reflect the largest minimum
Initial packet size of all their versions. This ensures that the server
responds if there are any mutually supported versions.
### 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.
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.
### 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
contents 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 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
1.3 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 ensures confidentiality and integrity protection
that meets the requirements of the cryptographic handshake protocol:
* 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)
* for the server, the ability to carry data that provides assurance that the
client can receive packets that are addressed with the transport address that
is claimed by the client (see {{address-validation}})
The initial CRYPTO frame MUST be sent in a single packet. Any second attempt
that is triggered by address validation 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 octet QUIC packet payload. This includes overheads that reduce the space
available to the cryptographic handshake protocol.
The CRYPTO frame can be sent in different packet number spaces. CRYPTO frames
in each packet number space carry a separate sequence of handshake data starting
from an offset of 0.
## Example Handshake Flows
Details of how TLS is integrated with QUIC are provided in {{QUIC-TLS}}, but
some examples are provided here.
{{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 contents. 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 ACKs the 0-RTT data at the 1-RTT encryption level, and the client's
sequence numbers at the 1-RTT encryption level continue to increment from its
0-RTT packets.
~~~~
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]
0-RTT[1]: CRYPTO[EOED]
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 1-RTT Handshake"}
## 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 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 {
initial_max_stream_data_bidi_local(0),
initial_max_data(1),
initial_max_bidi_streams(2),
idle_timeout(3),
preferred_address(4),
max_packet_size(5),
stateless_reset_token(6),
ack_delay_exponent(7),
initial_max_uni_streams(8),
disable_migration(9),
initial_max_stream_data_bidi_remote(10),
initial_max_stream_data_uni(11),
(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<22..2^16-1>;
} TransportParameters;
struct {
enum { IPv4(4), IPv6(6), (15) } ipVersion;
opaque ipAddress<4..2^8-1>;
uint16 port;
opaque connectionId<0..18>;
opaque statelessResetToken[16];
} PreferredAddress;
~~~
{: #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 encode the transport parameters.
QUIC encodes transport parameters into a sequence of octets, which are then
included 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.
### Transport Parameter Definitions
An endpoint MUST include the following parameters in its encoded
TransportParameters:
idle_timeout (0x0003):
: The idle timeout is a value in seconds that is encoded as an unsigned 16-bit
integer. The maximum value is 600 seconds (10 minutes).
An endpoint MAY use the following transport parameters:
initial_max_data (0x0001):
: The initial maximum data parameter contains the initial value for the maximum
amount of data that can be sent on the connection. This parameter is encoded
as an unsigned 32-bit integer in units of octets. This is equivalent to
sending a MAX_DATA ({{frame-max-data}}) for the connection immediately after
completing the handshake. If the transport parameter is absent, the connection
starts with a flow control limit of 0.
initial_max_bidi_streams (0x0002):
: The initial maximum bidirectional streams parameter contains the initial
maximum number of bidirectional streams the peer may initiate, encoded as an
unsigned 16-bit integer. If this parameter is absent or zero, bidirectional
streams cannot be created until a MAX_STREAM_ID frame is sent. Setting this
parameter is equivalent to sending a MAX_STREAM_ID ({{frame-max-stream-id}})
immediately after completing the handshake containing the corresponding Stream
ID. For example, a value of 0x05 would be equivalent to receiving a
MAX_STREAM_ID containing 16 when received by a client or 17 when received by a
server.
initial_max_uni_streams (0x0008):
: The initial maximum unidirectional streams parameter contains the initial
maximum number of unidirectional streams the peer may initiate, encoded as an
unsigned 16-bit integer. If this parameter is absent or zero, unidirectional
streams cannot be created until a MAX_STREAM_ID frame is sent. Setting this
parameter is equivalent to sending a MAX_STREAM_ID ({{frame-max-stream-id}})
immediately after completing the handshake containing the corresponding Stream
ID. For example, a value of 0x05 would be equivalent to receiving a
MAX_STREAM_ID containing 18 when received by a client or 19 when received by a
server.
max_packet_size (0x0005):
: The maximum packet size parameter places a limit on the size of packets that
the endpoint is willing to receive, encoded as an unsigned 16-bit integer.
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}}).
ack_delay_exponent (0x0007):
: An 8-bit unsigned integer value indicating an exponent used to decode the ACK
Delay field in the ACK frame, see {{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.
disable_migration (0x0009):
: The endpoint does not support connection migration ({{migration}}). Peers 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.
Either peer MAY advertise an initial value for the flow control on each type of
stream on which they might receive data. Each of the following transport
parameters is encoded as an unsigned 32-bit integer in units of octets:
initial_max_stream_data_bidi_local (0x0000):
: The initial stream maximum data for bidirectional, locally-initiated streams
parameter contains the initial flow control limit for newly created
bidirectional streams opened by the endpoint that sets the transport
parameter. In client transport parameters, this applies to streams with an
identifier ending in 0x0; in server transport parameters, this applies to
streams ending in 0x1.
initial_max_stream_data_bidi_remote (0x000a):
: The initial stream maximum data for bidirectional, peer-initiated streams
parameter contains the initial flow control limit for newly created
bidirectional streams opened by the endpoint that receives the transport
parameter. In client transport parameters, this applies to streams with an
identifier ending in 0x1; in server transport parameters, this applies to
streams ending in 0x0.
initial_max_stream_data_uni (0x000b):
: The initial stream maximum data for unidirectional streams parameter contains
the initial flow control limit for newly created unidirectional streams opened
by the endpoint that receives the transport parameter. In client transport
parameters, this applies to streams with an identifier ending in 0x3; in
server transport parameters, this applies to streams ending in 0x2.
If present, transport parameters that set initial stream flow control limits 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 server MAY include the following transport parameters:
stateless_reset_token (0x0006):
: The Stateless Reset Token is used in verifying a stateless reset, see
{{stateless-reset}}. This parameter is a sequence of 16 octets.
preferred_address (0x0004):
: 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}}.
A client MUST NOT include a stateless reset token or a preferred address. A
server MUST treat receipt of either transport parameter as a connection error of
type TRANSPORT_PARAMETER_ERROR.
### 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 initial_max_data,
initial_max_stream_data_bidi_local, initial_max_stream_data_bidi_remote, and
initial_max_stream_data_uni that are smaller than the remembered value of those
parameters. Similarly, a server MUST NOT reduce the value of
initial_max_bidi_streams or initial_max_uni_streams.
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_bidi_streams and initial_max_stream_data_bidi_remote, or
initial_max_uni_streams 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 reject 0-RTT data or even 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-parameters}}). 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.
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.
## Stateless Retries {#stateless-retry}
A server can process an initial cryptographic handshake messages from a client
without committing any state. This allows a server to perform address validation
({{address-validation}}), or to defer connection establishment costs.
A server that generates a response to an Initial packet without retaining
connection state MUST use the Retry packet ({{packet-retry}}). This packet
causes a client to restart the connection attempt and includes the token in the
new Initial packet ({{packet-initial}}) to prove source address ownership.
## Using Explicit Congestion Notification {#using-ecn}
QUIC endpoints 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 and peer support ECN
marking. Verifying the path occurs at the beginning of a connection and when the
connection migrates to a new path (see {{migration}}).
Each endpoint independently verifies and enables ECN for the path from it to the
peer.
To verify that both a path and the peer support ECN, an endpoint MUST set one of
the ECN Capable Transport (ECT) codepoints -- ECT(0) or ECT(1) -- in the IP
header {{!RFC8311}} of all outgoing packets.
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.
On receiving a packet with an ECT or CE codepoint, an endpoint that supports ECN
increases the corresponding ECT(0), ECT(1), or CE count, and includes these
counters in subsequent (see {{processing-and-ack}}) ACK_ECN frames (see
{{frame-ack-ecn}}).
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 packet without an ECT or CE codepoint, it responds per
{{processing-and-ack}} with an ACK frame.
If an endpoint does not support ECN or does not have access to received ECN
codepoints, it acknowledges received packets per {{processing-and-ack}} with an
ACK frame.
If a packet sent with an ECT codepoint is newly acknowledged by the peer in an
ACK frame, the endpoint stops setting ECT codepoints in subsequent packets, with
the expectation that either the network or the peer no longer supports ECN.
To protect the connection from arbitrary corruption of ECN codepoints by the
network, an endpoint verifies the following when an ACK_ECN frame is received:
* The increase in ECT(0) and ECT(1) counters MUST be at least the number of
packets newly acknowledged that were sent with the corresponding codepoint.
* The total increase in ECT(0), ECT(1), and CE counters reported in the ACK_ECN
frame MUST be at least the total number of packets newly acknowledged in this
ACK_ECN frame.
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
counters to be greater than the number of packets acknowledged in an ACK frame.
When this happens, the local reference counts MUST be increased to match the
counters 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 packets with the expectation that either the network or the peer does
not support ECN.
If an endpoint sets ECT codepoints on outgoing 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 a network
element might be corrupting ECN codepoints, the endpoint MAY cease setting ECT
codepoints in subsequent packets. Doing so allows the connection to traverse
network elements that drop or corrupt ECN codepoints in the IP header.
## Proof of Source Address Ownership {#address-validation}
Transport protocols commonly spend a round trip checking that a client owns the
transport address (IP and port) that it claims. Verifying that a client can
receive packets sent to its claimed transport address protects against spoofing
of this information by malicious clients.
This technique is used primarily to avoid QUIC from being used for 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.
Several methods are used in QUIC to mitigate this attack. Firstly, the initial
handshake packet is sent in a UDP datagram that contains at least 1200 octets of
UDP payload. This allows a server to send a similar amount of data without
risking causing an amplification attack toward an unproven remote address.
A server eventually confirms that a client has received its messages when the
first Handshake-level message is received. This might be insufficient,
either because the server wishes to avoid the computational cost of completing
the handshake, or it might be that the size of the packets that are sent during
the handshake is too large. This is especially important for 0-RTT, where the
server might wish to provide application data traffic - such as a response to a
request - in response to the data carried in the early data from the client.
To send additional data prior to completing the cryptographic handshake, the
server then needs to validate that the client owns the address that it claims.
Source address validation is therefore performed by the core transport
protocol during the establishment of a connection.
A different type of source address validation is performed after a connection
migration, see {{migrate-validate}}.
### Client Address Validation Procedure
QUIC uses token-based address validation. Any time the server wishes
to validate a client address, it provides the client with a token. As
long as the token's authenticity can be checked (see
{{token-integrity}}) and the client is able to return that token, it
proves to the server that it received the token.
Upon receiving the client's Initial packet, the server can request
address validation by sending a Retry packet containing a token. This
token is repeated in the client's next Initial packet. Because the
token is consumed by the server that generates it, there is no need
for a single well-defined format. 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.
The Retry packet is sent to the client and a legitimate client will
respond with an Initial packet containing the token from the Retry packet
when it continues the handshake. In response to receiving the token, a
server can either abort the connection or permit it to proceed.
A connection MAY be accepted without address validation - or with only limited
validation - but a server SHOULD limit the data it sends toward an unvalidated
address. Successful completion of the cryptographic handshake implicitly
provides proof that the client has received packets from the server.
The client should allow for additional Retry packets being sent in
response to Initial packets sent containing a token. There are several
situations in which the server might not be able to use the previously
generated token to validate the client's address and must send a new
Retry. A reasonable limit to the number of tries the client allows
for, before giving up, is 3. That is, the client MUST echo the
address validation token from a new Retry packet up to 3 times. After
that, it MAY give up on the connection attempt.
### Address Validation for Future Connections
A server MAY provide clients with an address validation token during one
connection that can be used on a subsequent connection. Address validation is
especially important with 0-RTT because a server potentially sends a significant
amount of data to a client in response to 0-RTT data.
The server uses the NEW_TOKEN frame {{frame-new-token}} to provide the
client with an address validation token that can be used to validate
future connections. The client may then use this token to validate
future connections by including it in the Initial packet's header.
The client MUST NOT use the token provided in a Retry for future
connections.
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.
### 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.
## Path Validation {#migrate-validate}
Path validation is used by an endpoint to verify reachability of a peer over a
specific path. That is, it tests 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.
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. Path validation is also used by the peer to
verify that the migrating endpoint is able to receive packets sent to the its
new address. That is, 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. For instance, 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.
### Initiation
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 additional PATH_CHALLENGE frames to handle 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.
### Response
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,
with the following stipulation. Since a PATH_CHALLENGE might be sent from a
spoofed address, an endpoint MAY 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:
from the same local address on which the PATH_CHALLENGE was received, to the
same remote address from which the PATH_CHALLENGE was received.
### Completion
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. 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 considered to have failed, even if
the data matches that sent in the PATH_CHALLENGE. Thus, the 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.
### Abandonment
An endpoint SHOULD abandon path validation after sending some number of
PATH_CHALLENGE frames or after some time has passed. When setting this timer,
implementations are cautioned that the new path could have a longer round-trip
time than the original.
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.
If path validation fails, the path is deemed 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}
QUIC allows connections to survive changes to endpoint addresses (that is, IP
address and/or port), such as those caused by a 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.
Endpoints SHOULD perform path validation ({{migrate-validate}}) if a NAT
rebinding does not cause the connection to fail.
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 it sees a non-probing packet from that address. If a
client receives packets from an unknown server address, the client MAY 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.
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, 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}
A endpoint can migrate a connection to a new local address by sending packets
containing frames other than 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 {{using-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.
#### Handling Address Spoofing by a Peer {#address-spoofing}
It is possible that a peer is spoofing its source address to cause an endpoint
to send excessive amounts of data to an unwilling host. If the endpoint sends
significantly more data than the spoofing peer, connection migration might be
used to amplify the volume of data that an attacker can generate toward a
victim.
As described in {{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.
#### Handling Address Spoofing by an On-path Attacker {#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.
### 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. 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}}.
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.
## 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 the handshake is completed. If this packet contains a PATH_CHALLENGE
frame, the server sends a PATH_RESPONSE frame as per {{migrate-validate}}, but
the server MUST continue sending all other 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 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 using the original address from the client's new 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 or APPLICATION_CLOSE frame (see
{{immediate-close}} for details).
In the closing state, only a packet containing a closing frame can be sent. An
endpoint retains only enough information to generate a packet containing a
closing 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 closing 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 closing
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
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 their own idle timeout to their 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 4.3.3 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 closing frame (CONNECTION_CLOSE or APPLICATION_CLOSE) to
terminate the connection immediately. Any closing frame causes all streams to
immediately become closed; open streams can be assumed to be implicitly reset.
After sending a closing frame, endpoints immediately enter the closing state.
During the closing period, an endpoint that sends a closing frame SHOULD respond
to any packet that it receives with another packet containing a closing 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 closing 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.
After receiving a closing frame, endpoints enter the draining state. An
endpoint that receives a closing frame MAY send a single packet containing a
closing 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 closing 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 APPLICATION_CLOSE message with an appropriate error code to
signal closure.
### Stateless Reset {#stateless-reset}
A stateless reset is provided as an option of last resort for an endpoint that
does not have access to the state of a connection. A crash or outage might
result in peers continuing to send data to an endpoint that is unable to
properly continue the connection. An endpoint that wishes to communicate a
fatal connection error MUST use a closing 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.
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|K|1|1|0|0|0|0|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Random Octets (160..) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ +
| |
+ 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.
The message consists of a header octet, followed by an arbitrary number of
random octets, followed by 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 Octets field needs to
include at least 20 octets of random or unpredictable values. This is intended
to allow for a destination connection ID of the maximum length permitted, a
packet number, and minimal payload. The Stateless Reset Token corresponds to
the minimum expansion of the packet protection AEAD. More random octets 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 less than 19 octets long 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
ineffective in causing errors to be quickly detected and recovered. 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 octets of the packet are set to the value of the Stateless
Reset Token.
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 or APPLICATION_CLOSE frame if it has sufficient state to do so.
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 octets 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 octets 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
a 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 octets 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 MUST NOT
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 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
octets 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.
# Frame Types and Formats
As described in {{frames}}, packets contain one or more frames. This section
describes the format and semantics of the core QUIC frame types.
## Variable-Length Integer Encoding {#integer-encoding}
QUIC frames commonly use a variable-length encoding for non-negative integer
values. This encoding ensures that smaller integer values need fewer octets to
encode.
The QUIC variable-length integer encoding reserves the two most significant bits
of the first octet to encode the base 2 logarithm of the integer encoding length
in octets. 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 octets 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 octet sequence c2 19 7c 5e ff 14 e8 8c (in hexadecimal)
decodes to the decimal value 151288809941952652; the four octet sequence 9d 7f
3e 7d decodes to 494878333; the two octet sequence 7b bd decodes to 15293; and
the single octet 25 decodes to 37 (as does the two octet sequence 40 25).
Error codes ({{error-codes}}) are described using integers, but do not use this
encoding.
## 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
octet that identifies the frame as a PADDING frame.
## RST_STREAM Frame {#frame-rst-stream}
An endpoint may use a RST_STREAM frame (type=0x01) to abruptly terminate a
stream.
After sending a RST_STREAM, an endpoint ceases transmission and retransmission
of STREAM frames on the identified stream. A receiver of RST_STREAM can discard
any data that it already received on that stream.
An endpoint that receives a RST_STREAM frame for a send-only stream MUST
terminate the connection with error PROTOCOL_VIOLATION.
The RST_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) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
The fields are:
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 RST_STREAM sender.
## CONNECTION_CLOSE frame {#frame-connection-close}
An endpoint sends a CONNECTION_CLOSE frame (type=0x02) to notify its peer that
the connection is being closed. CONNECTION_CLOSE is used to signal errors at
the QUIC layer, or the absence of errors (with the NO_ERROR code).
If there are open streams that haven't been explicitly closed, they are
implicitly closed when the connection is closed.
The CONNECTION_CLOSE 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Error Code (16) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Frame Type (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Reason Phrase Length (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Reason Phrase (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
The fields of a CONNECTION_CLOSE frame are as follows:
Error Code:
: A 16-bit error code which indicates the reason for closing this connection.
CONNECTION_CLOSE uses codes from the space defined in {{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.
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}}.
## APPLICATION_CLOSE frame {#frame-application-close}
An APPLICATION_CLOSE frame (type=0x03) is used to signal an error with the
protocol that uses QUIC.
The APPLICATION_CLOSE 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Error Code (16) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Reason Phrase Length (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Reason Phrase (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
The fields of a APPLICATION_CLOSE frame are as follows:
Error Code:
: A 16-bit error code which indicates the reason for closing this connection.
APPLICATION_CLOSE uses codes from the application protocol error code space,
see {{app-error-codes}}.
Reason Phrase Length:
: This field is identical in format and semantics to the Reason Phrase Length
field from CONNECTION_CLOSE.
Reason Phrase:
: This field is identical in format and semantics to the Reason Phrase field
from CONNECTION_CLOSE.
APPLICATION_CLOSE has similar format and semantics to the CONNECTION_CLOSE frame
({{frame-connection-close}}). Aside from the semantics of the Error Code field
and the omission of the Frame Type field, both frames are used to close the
connection.
## MAX_DATA Frame {#frame-max-data}
The MAX_DATA frame (type=0x04) 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 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) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
The fields in the MAX_DATA frame are as follows:
Maximum Data:
: A variable-length integer indicating the maximum amount of data that can be
sent on the entire connection, in units of octets.
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 QUIC_FLOW_CONTROL_RECEIVED_TOO_MUCH_DATA 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=0x05) 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 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) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
The fields in the MAX_STREAM_DATA frame are as follows:
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 octets.
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_STREAM_ID Frame {#frame-max-stream-id}
The MAX_STREAM_ID frame (type=0x06) informs the peer of the maximum stream ID
that they are permitted to open.
The 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 Stream ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
The fields in the MAX_STREAM_ID frame are as follows:
Maximum Stream ID:
: ID of the maximum unidirectional or bidirectional peer-initiated stream ID for
the connection encoded as a variable-length integer. The limit applies to
unidirectional steams if the second least signification bit of the stream ID
is 1, and applies to bidirectional streams if it is 0.
Loss or reordering can mean that a MAX_STREAM_ID frame can be received which
states a lower stream limit than the client has previously received.
MAX_STREAM_ID frames which do not increase the maximum stream ID MUST be
ignored.
A peer MUST NOT initiate a stream with a higher stream ID than the greatest
maximum stream ID it has received. An endpoint MUST terminate a connection with
a STREAM_ID_ERROR error if a peer initiates a stream with a higher stream ID
than it has sent, unless this is a result of a change in the initial limits (see
{{zerortt-parameters}}).
## PING Frame {#frame-ping}
Endpoints can use PING frames (type=0x07) 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.
## BLOCKED Frame {#frame-blocked}
A sender SHOULD send a BLOCKED frame (type=0x08) when it wishes to send data,
but is unable to due to connection-level flow control (see {{blocking}}).
BLOCKED frames can be used as input to tuning of flow control algorithms (see
{{fc-credit}}).
The 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Offset (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
The BLOCKED frame contains a single field.
Offset:
: A variable-length integer indicating the connection-level offset at which
the blocking occurred.
## STREAM_BLOCKED Frame {#frame-stream-blocked}
A sender SHOULD send a STREAM_BLOCKED frame (type=0x09) when it wishes to send
data, but is unable to due to stream-level flow control. This frame is
analogous to BLOCKED ({{frame-blocked}}).
An endpoint that receives a STREAM_BLOCKED frame for a send-only stream MUST
terminate the connection with error PROTOCOL_VIOLATION.
The STREAM_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) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Offset (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
The STREAM_BLOCKED frame contains two fields:
Stream ID:
: A variable-length integer indicating the stream which is flow control blocked.
Offset:
: A variable-length integer indicating the offset of the stream at which the
blocking occurred.
## STREAM_ID_BLOCKED Frame {#frame-stream-id-blocked}
A sender MAY send a STREAM_ID_BLOCKED frame (type=0x0a) when it wishes to open a
stream, but is unable to due to the maximum stream ID limit set by its peer (see
{{frame-max-stream-id}}). This does not open the stream, but informs the peer
that a new stream was needed, but the stream limit prevented the creation of the
stream.
The STREAM_ID_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) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
The STREAM_ID_BLOCKED frame contains a single field.
Stream ID:
: A variable-length integer indicating the highest stream ID that the sender
was permitted to open.
## NEW_CONNECTION_ID Frame {#frame-new-connection-id}
An endpoint sends a NEW_CONNECTION_ID frame (type=0x0b) 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 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 (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Length (8) | Connection ID (32..144) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ +
| |
+ Stateless Reset Token (128) +
| |
+ +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
The fields are:
Sequence:
: A variable-length integer. This value starts at 0 and increases by 1 for each
connection ID that is provided by the server. The connection ID that is
assigned during the handshake is assumed to have a sequence of -1. That is,
the value selected during the handshake comes immediately before the first
value that a server can send.
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 to 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.
## STOP_SENDING Frame {#frame-stop-sending}
An endpoint may use a STOP_SENDING frame (type=0x0c) 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 only valid for a send stream that exists and
is not in the "Ready" state (see {{stream-send-states}}). Receiving a
STOP_SENDING frame for a send stream that is "Ready" or non-existent 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) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
The fields are:
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}}).
## ACK Frame {#frame-ack}
Receivers send ACK frames (type=0x0d) to inform senders which packets they have
received and processed. The ACK frame contains any number of ACK blocks.
ACK blocks are ranges of acknowledged packets.
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.
A client MUST NOT acknowledge Retry packets. Retry packets include the packet
number from the Initial packet it responds to. Version Negotiation packets
cannot be acknowledged because they do not contain a packet number. Rather than
relying on ACK frames, these packets are implicitly acknowledged by the next
Initial packet sent by the client.
An ACK frame is shown 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Largest Acknowledged (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ACK Delay (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ACK Block Count (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ACK Blocks (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #ack-format title="ACK Frame Format"}
The fields in the ACK frame are as follows:
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 the 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 octets.
~~~
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.
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.
### Sending ACK Frames
Implementations MUST NOT generate packets that only contain ACK frames in
response to packets which only contain ACK frames. However, they MUST
acknowledge packets containing only ACK frames when sending ACK frames in
response to other packets. Implementations MUST NOT send more than one packet
containing only ACK frames per received packet that contains frames other than
ACK frames. Packets containing non-ACK frames MUST be acknowledged immediately
or when a delayed ack timer expires.
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.
Endpoints can only acknowledge packets sent in a particular packet
number space by sending ACK frames in packets from the same packet
number space.
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 4.3.1 of {{QUIC-RECOVERY}}.
## ACK_ECN Frame {#frame-ack-ecn}
The ACK_ECN frame (type=0x1a) is used by an endpoint that supports ECN to
acknowledge packets received with 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Largest Acknowledged (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ACK Delay (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ECT(0) Count (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ECT(1) Count (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ECN-CE Count (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ACK Block Count (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ACK Blocks (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #ACN_ECN_FRAME_FORMAT title="ACK_ECN Frame Format"}
An ACK_ECN frame contains all the elements of the ACK frame ({{frame-ack}}) with
the addition of three counts following the ACK Delay field.
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.
## PATH_CHALLENGE Frame {#frame-path-challenge}
Endpoints can use PATH_CHALLENGE frames (type=0x0e) to check reachability to the
peer and for path validation during connection establishment and connection
migration.
PATH_CHALLENGE frames contain an 8-byte payload.
~~~
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) +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
Data:
: This 8-byte field contains arbitrary data.
A PATH_CHALLENGE frame containing 8 octets 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=0x0f) 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.
## NEW_TOKEN frame {#frame-new-token}
A server sends a NEW_TOKEN frame (type=0x19) 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 (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
The fields of a NEW_TOKEN frame are as follows:
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 0b00010XXX (or the set of values from 0x10 to 0x17). 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 octets 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.
A STREAM frame is 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Offset (i)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Length (i)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream Data (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #stream-format title="STREAM Frame Format"}
The STREAM frame contains 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
octets 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.
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.
Implementation note: 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. An implementation is therefore
advised to bundle as few streams as necessary in outgoing packets without losing
transmission efficiency to underfilled packets.
## CRYPTO Frame {#frame-crypto}
The CRYPTO frame (type=0x18) 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.
A CRYPTO frame is 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Offset (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Length (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Crypto Data (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
~~~
{: #crypto-format title="CRYPTO Frame Format"}
The CRYPTO frame contains 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.
# Packetization and Reliability {#packetization}
A sender bundles one or more frames in a QUIC packet (see {{frames}}).
A sender SHOULD 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.
## 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. Any stream
state transitions triggered by the frame MUST have occurred. 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. 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 acknowledge packets containing only ACK or PADDING frames in the next ACK
frame that it sends.
While PADDING frames do not elicit an ACK frame from a receiver, they 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.
Strategies and implications of the frequency of generating acknowledgments are
discussed in more detail in {{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 are retransmitted according to the rules in
{{QUIC-RECOVERY}}, until either all data has been acknowledged or the crypto
state machine implicitly knows that the peer received the data.
* Application data sent in STREAM frames is retransmitted in new STREAM frames
unless the endpoint has sent a RST_STREAM for that stream. Once an endpoint
sends a RST_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 RST_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 RST_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 those that use CONNECTION_CLOSE and
APPLICATION_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 maximum stream ID for a stream of a given type is sent in MAX_STREAM_ID
frames. Like MAX_DATA, an updated value is sent when a packet containing the
most recent MAX_STREAM_ID 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 BLOCKED, STREAM_BLOCKED, and STREAM_ID_BLOCKED
frames. BLOCKED streams have connection scope, STREAM_BLOCKED frames have
stream scope, and STREAM_ID_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.
* PADDING frames contain no information, so lost PADDING frames do not require
repair.
Upon detecting losses, a sender MUST take appropriate congestion control action.
The details of loss detection and congestion control are described in
{{QUIC-RECOVERY}}.
## Packet Size {#packet-size}
The QUIC packet size includes the QUIC header and integrity check, but not the
UDP or IP header.
Clients MUST ensure that the first Initial packet it sends is sent in a UDP
datagram that is at least 1200 octets. Padding the Initial packet or including a
0-RTT packet in the same datagram are ways to meet this requirement. 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.
The datagram containing the first Initial packet from a client MAY exceed 1200
octets 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 octets. 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.
## Path Maximum Transmission Unit
The Path Maximum Transmission Unit (PMTU) is the maximum size of the entire IP
header, UDP header, and UDP payload. The UDP payload includes the QUIC packet
header, protected payload, and any authentication fields.
All QUIC packets SHOULD be sized to fit within the estimated PMTU to avoid IP
fragmentation or packet drops. To optimize bandwidth efficiency, endpoints
SHOULD use Packetization Layer PMTU Discovery ({{!PLPMTUD=RFC4821}}). Endpoints
MAY use PMTU Discovery ({{!PMTUDv4=RFC1191}}, {{!PMTUDv6=RFC8201}}) for
detecting the PMTU, setting the PMTU appropriately, and storing the result of
previous PMTU determinations.
In the absence of these mechanisms, QUIC endpoints SHOULD NOT send IP packets
larger than 1280 octets. Assuming the minimum IP header size, this results in
a QUIC packet size of 1232 octets for IPv6 and 1252 octets for IPv4. Some
QUIC implementations MAY wish to be more conservative in computing allowed
QUIC packet size given unknown tunneling overheads or IP header options.
QUIC endpoints that implement any kind of PMTU discovery SHOULD maintain an
estimate for each combination of local and remote IP addresses. Each pairing of
local and remote addresses could have a different maximum MTU in the path.
QUIC depends on the network path supporting a MTU of at least 1280 octets. This
is the IPv6 minimum MTU and therefore also supported by most modern IPv4
networks. An endpoint MUST NOT reduce its MTU below this number, even if it
receives signals that indicate a smaller limit might exist.
If a QUIC endpoint determines that the PMTU between any pair of local and remote
IP addresses has fallen below 1280 octets, it MUST immediately cease sending
QUIC packets on the affected path. This could result in termination of the
connection if an alternative path cannot be found.
### IPv4 PMTU Discovery {#v4-pmtud}
Traditional ICMP-based path MTU discovery in IPv4 {{!PMTUDv4}} is potentially
vulnerable to off-path attacks that successfully guess the IP/port 4-tuple and
reduce the MTU to a bandwidth-inefficient value. TCP connections mitigate this
risk by using the (at minimum) 8 bytes of transport header echoed in the ICMP
message to validate the TCP sequence number as valid for the current
connection. However, as QUIC operates over UDP, in IPv4 the echoed information
could consist only of the IP and UDP headers, which usually has insufficient
entropy to mitigate off-path attacks.
As a result, endpoints that implement PMTUD in IPv4 SHOULD take steps to
mitigate this risk. For instance, an application could:
* Set the IPv4 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.
* Store additional information from the IP or UDP headers from DF packets (for
example, the IP ID or UDP checksum) to further authenticate incoming Datagram
Too Big messages.
* Any reduction in PMTU due to a report contained in an ICMP packet is
provisional until QUIC's loss detection algorithm determines that the packet is
actually lost.
### Special Considerations for Packetization Layer PMTU Discovery
The PADDING frame provides a useful option for PMTU probe packets. PADDING
frames generate acknowledgements, but they need not be delivered reliably. As a
result, the loss of PADDING frames in probe packets does not require
delay-inducing retransmission. However, PADDING frames do consume congestion
window, which may delay the transmission of subsequent application data.
When implementing the algorithm in Section 7.2 of {{!PLPMTUD}}, the initial
value of search_low SHOULD be consistent with the IPv6 minimum packet size.
Paths that do not support this size cannot deliver Initial packets, and
therefore are not QUIC-compliant.
Section 7.3 of {{!PLPMTUD}} discusses trade-offs between small and large
increases in the size of probe packets. As QUIC probe packets need not contain
application data, aggressive increases in probe size carry fewer consequences.
# Streams: QUIC's Data Structuring Abstraction {#streams}
Streams in QUIC provide a lightweight, ordered byte-stream abstraction.
There are two basic types of stream in QUIC. Unidirectional streams carry data
in one direction only; bidirectional streams allow for data to be sent in both
directions. Different stream identifiers are used to distinguish between
unidirectional and bidirectional streams, as well as to create a separation
between streams that are initiated by the client and server (see {{stream-id}}).
Either type of stream can be created by either endpoint, can concurrently send
data interleaved with other streams, and can be cancelled.
Stream offsets allow for the octets on a stream to be placed in order. An
endpoint MUST be capable of delivering data received on a stream in order.
Implementations MAY choose to offer the ability to deliver data out of order.
There is no means of ensuring ordering between octets on different streams.
The creation and destruction of streams are expected to have minimal bandwidth
and computational cost. A single STREAM frame may create, carry data for, and
terminate a stream, or a stream may last the entire duration of a connection.
Streams are individually flow controlled, allowing an endpoint to limit memory
commitment and to apply back pressure. The creation of streams is also flow
controlled, with each peer declaring the maximum stream ID it is willing to
accept at a given time.
An alternative view of QUIC streams is as an elastic "message" abstraction,
similar to the way ephemeral streams are used in SST
{{?SST=DOI.10.1145/1282427.1282421}}, which may be a more appealing description
for some applications.
## Stream Identifiers {#stream-id}
Streams are identified by an unsigned 62-bit integer, referred to as the Stream
ID. The least significant two bits of the Stream ID are used to identify the
type of stream (unidirectional or bidirectional) and the initiator of the
stream.
The least significant bit (0x1) of the Stream ID identifies the initiator of the
stream. Clients initiate even-numbered streams (those with the least
significant bit set to 0); servers initiate odd-numbered streams (with the bit
set to 1). Separation of the stream identifiers ensures that client and server
are able to open streams without the latency imposed by negotiating for an
identifier.
If an endpoint receives a frame for a stream that it expects to initiate (i.e.,
odd-numbered for the client or even-numbered for the server), but which it has
not yet opened, it MUST close the connection with error code STREAM_STATE_ERROR.
The second least significant bit (0x2) of the Stream ID differentiates between
unidirectional streams and bidirectional streams. Unidirectional streams always
have this bit set to 1 and bidirectional streams have this bit set to 0.
The two type bits from a Stream ID therefore identify streams as summarized in
{{stream-id-types}}.
| Low 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"}
The first bi-directional stream opened by the client is stream 0.
A QUIC endpoint MUST NOT reuse a Stream ID. Streams of each type are created in
numeric order. Streams that are used out of order result in opening all
lower-numbered streams of the same type in the same direction.
Stream IDs are encoded as a variable-length integer (see {{integer-encoding}}).
## Stream States {#stream-states}
This section describes the two types of QUIC stream in terms of the states of
their send or receive components. Two state machines are described: one for
streams on which an endpoint transmits data ({{stream-send-states}}); another
for streams from 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 can open streams up to its maximum stream limit in any order,
however endpoints SHOULD open the send side of streams for each type in order.
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)
| Create Bidirectional Stream (Receiving)
v
+-------+
| Ready | Send RST_STREAM
| |-----------------------.
+-------+ |
| |
| Send STREAM / |
| STREAM_BLOCKED |
v |
+-------+ |
| Send | Send RST_STREAM |
| |---------------------->|
+-------+ |
| |
| Send STREAM + FIN |
v v
+-------+ +-------+
| Data | Send RST_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 or application
protocol. 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.
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 if the receiving part
enters the "Recv" state.
Sending the first STREAM or STREAM_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.
In the "Send" state, an endpoint transmits - and retransmits as necessary - data
in STREAM frames. The endpoint respects the flow control limits of its peer,
accepting MAX_STREAM_DATA frames. An endpoint in the "Send" state generates
STREAM_BLOCKED frames if it encounters flow control limits.
After the application indicates that stream data is complete 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 no longer needs to track flow control limits or send STREAM_BLOCKED
frames for a send stream in this state. The endpoint can ignore any
MAX_STREAM_DATA frames it receives from its peer in this state; MAX_STREAM_DATA
frames might be received until the peer receives the final stream offset.
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. Similarly, the
endpoint might receive a STOP_SENDING frame from its peer. In either case, the
endpoint sends a RST_STREAM frame, which causes the stream to enter the "Reset
Sent" state.
An endpoint MAY send a RST_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 RST_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 doesn't 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 or
application protocol some of which cannot be observed by the sender.
~~~
o
| Recv STREAM / STREAM_BLOCKED / RST_STREAM
| Create Bidirectional Stream (Sending)
| Recv MAX_STREAM_DATA
| Create Higher-Numbered Stream
v
+-------+
| Recv | Recv RST_STREAM
| |-----------------------.
+-------+ |
| |
| Recv STREAM + FIN |
v |
+-------+ |
| Size | Recv RST_STREAM |
| Known +---------------------->|
+-------+ |
| |
| Recv All Data |
v v
+-------+ +-------+
| Data | Recv RST_STREAM | Reset |
| Recvd +<-- (optional) --->| Recvd |
+-------+ +-------+
| |
| 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) are created when the first STREAM, STREAM_BLOCKED,
RST_STREAM, or MAX_STREAM_DATA (bidirectional only, see below) is received for
that stream. The initial state for a receive stream is "Recv". Receiving a
RST_STREAM frame causes the receive stream to immediately transition to the
"Reset Recvd".
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.
A bidirectional stream also opens when a MAX_STREAM_DATA frame is received.
Receiving a MAX_STREAM_DATA frame implies that the remote peer has opened the
stream and is providing flow control credit. A MAX_STREAM_DATA frame might
arrive before a STREAM or STREAM_BLOCKED frame if packets are lost or reordered.
Before creating a stream, all lower-numbered streams of the same type MUST be
created. That means that receipt of a frame that would open a stream causes all
lower-numbered streams of the same type to be opened in numeric order. This
ensures that the creation order for streams is consistent on both endpoints.
In the "Recv" state, the endpoint receives STREAM and STREAM_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 (see
{{final-offset}}) is known. 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_BLOCKED frames it receives for the
stream can be discarded.
The "Data Recvd" state persists until stream data has been delivered to the
application or application protocol. Once stream data has been delivered, the
stream enters the "Data Read" state, which is a terminal state.
Receiving a RST_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 RST_STREAM is received
(that is, from the "Data Recvd" state). Similarly, it is possible for remaining
stream data to arrive after receiving a RST_STREAM frame (the "Reset Recvd"
state). An implementation is able to manage this situation as they choose.
Sending RST_STREAM means that an endpoint cannot guarantee delivery of stream
data; however there is no requirement that stream data not be delivered if a
RST_STREAM is received. An implementation MAY interrupt delivery of stream
data, discard any data that was not consumed, and signal the existence of the
RST_STREAM immediately. Alternatively, the RST_STREAM signal might be
suppressed or withheld if stream data is completely received. In the latter
case, the receive stream effectively transitions to "Data Recvd" from "Reset
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_BLOCKED
({{frame-stream-blocked}}), and RST_STREAM ({{frame-rst-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_BLOCKED after
sending a RST_STREAM; that is, in the "Reset Sent" state in addition to the
terminal states. A receiver could receive any of these frames in any state, but
only 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 RST_STREAM frame;
that is states other than "Reset Recvd" or "Reset Read". However there is
little value in sending a STOP_SENDING frame after all stream data has been
received in the "Data Recvd" state. A sender could receive these 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
is not a guarantee that incoming data will be ignored.
STREAM frames received after sending STOP_SENDING are still counted toward the
connection and stream flow-control windows, even though these frames will be
discarded upon receipt. This avoids potential ambiguity about which STREAM
frames count toward flow control.
A STOP_SENDING frame requests that the receiving endpoint send a RST_STREAM
frame. An endpoint that receives a STOP_SENDING frame MUST send a RST_STREAM
frame for that stream, and can use an error code of STOPPING. If the
STOP_SENDING frame is received on a send stream that is already in the "Data
Sent" state, a RST_STREAM frame MAY still be sent in order to cancel
retransmission of previously-sent STREAM frames.
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 RST_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.
## Stream Concurrency {#stream-concurrency}
An endpoint limits the number of concurrently active incoming streams by
adjusting the maximum stream ID. An initial value is set in the transport
parameters (see {{transport-parameter-definitions}}) and is subsequently
increased by MAX_STREAM_ID frames (see {{frame-max-stream-id}}).
The maximum stream ID is specific to each endpoint and applies only to the peer
that receives the setting. That is, clients specify the maximum stream ID the
server can initiate, and servers specify the maximum stream ID the client can
initiate. Each endpoint may respond on streams initiated by the other peer,
regardless of whether it is permitted to initiated new streams.
Endpoints MUST NOT exceed the limit set by their peer. An endpoint that
receives a STREAM frame with an ID greater than the limit it has sent MUST treat
this as a stream error of type STREAM_ID_ERROR ({{error-handling}}), unless this
is a result of a change in the initial limits (see {{zerortt-parameters}}).
A receiver cannot renege on an advertisement; that is, once a receiver
advertises a stream ID via a MAX_STREAM_ID frame, advertising a smaller maximum
ID has no effect. A sender MUST ignore any MAX_STREAM_ID frame that does not
increase the maximum stream ID.
## Sending and Receiving Data
Once a stream is created, endpoints may use the stream to send and receive data.
Each endpoint may send a series of STREAM frames encapsulating data on a stream
until the stream is terminated in that direction. Streams are an ordered
byte-stream abstraction, and they have no other structure within them. STREAM
frame boundaries are not expected to be preserved in retransmissions from the
sender or during delivery to the application at the receiver.
When new data is to be sent on a stream, a sender MUST set the encapsulating
STREAM frame's offset field to the stream offset of the first byte of this new
data. The first octet of data on a stream has an offset of 0. An endpoint is
expected to send every stream octet. The largest offset delivered on a stream
MUST be less than 2^62.
QUIC makes no specific allowances for partial reliability or delivery of stream
data out of 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.
An endpoint could receive the same octets multiple times; octets that have
already been received can be discarded. The value for a given octet MUST NOT
change if it is sent multiple times; an endpoint MAY treat receipt of a changed
octet as a connection error of type PROTOCOL_VIOLATION.
An endpoint MUST NOT send data on any stream without ensuring that it is within
the data limits set by its peer.
Flow control is described in detail in {{flow-control}}, and congestion control
is described in the companion document {{QUIC-RECOVERY}}.
## Stream Prioritization
Stream multiplexing has a significant effect on application performance if
resources allocated to streams are correctly prioritized. Experience with other
multiplexed protocols, such as HTTP/2 {{?HTTP2}}, shows that effective
prioritization strategies have a significant positive impact on performance.
QUIC does not provide frames for exchanging prioritization information. Instead
it relies on receiving priority information from the application that uses QUIC.
Protocols that use QUIC are able to define any prioritization scheme that suits
their application semantics. A protocol might define explicit messages for
signaling priority, such as those defined in HTTP/2; it could define rules that
allow an endpoint to determine priority based on context; or it could leave the
determination to the application.
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, QUIC SHOULD use the information provided by the application.
Failure to account for priority of streams can result in suboptimal performance.
Stream priority is most relevant when deciding which stream data will be
transmitted. Often, there will be limits on what can be transmitted as a result
of connection flow control or the current congestion controller state.
Giving preference to the transmission of its own management frames ensures that
the protocol functions efficiently. That is, prioritizing frames other than
STREAM frames ensures that loss recovery, congestion control, and flow control
operate effectively.
CRYPTO frames SHOULD be prioritized over other streams prior to the completion
of the cryptographic handshake. This includes the retransmission of the second
flight of client handshake messages, that is, the TLS Finished and any client
authentication messages.
STREAM data in frames determined to be lost SHOULD be retransmitted before
sending new data, unless application priorities indicate otherwise.
Retransmitting lost stream data can fill in gaps, which allows the peer to
consume already received data and free up flow control window.
# Flow Control {#flow-control}
It is necessary to limit the amount of data that a sender may have outstanding
at any time, so as to prevent a fast sender from overwhelming a slow receiver,
or to prevent a malicious sender from consuming significant resources at a
receiver. This section describes QUIC's flow-control mechanisms.
QUIC employs a credit-based flow-control scheme similar to HTTP/2's flow control
{{?HTTP2}}. A receiver advertises the number of octets it is prepared to
receive on a given stream and for the entire connection. This leads to two
levels of flow control in QUIC: (i) Connection flow control, which prevents
senders from exceeding a receiver's buffer capacity for the connection, and (ii)
Stream flow control, which prevents a single stream from consuming the entire
receive buffer for a connection.
A data receiver sends MAX_STREAM_DATA or MAX_DATA frames to the sender
to advertise additional credit. MAX_STREAM_DATA frames send the
maximum absolute byte offset of a stream, while MAX_DATA sends the
maximum of the sum of the absolute byte offsets of all streams.
A receiver MAY advertise a larger offset at any point by sending MAX_DATA or
MAX_STREAM_DATA frames. A receiver cannot renege on an advertisement; that is,
once a receiver advertises an offset, advertising a smaller offset has no
effect. A sender MUST therefore ignore any MAX_DATA or MAX_STREAM_DATA frames
that do not increase flow control limits.
A receiver MUST close the connection with a FLOW_CONTROL_ERROR error
({{error-handling}}) if the peer violates the advertised connection or stream
data limits.
A sender SHOULD send BLOCKED or STREAM_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 receiver advertises credit for a stream by sending a MAX_STREAM_DATA frame
with the Stream ID set appropriately. 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.
Connection flow control is a limit to the total bytes of stream data sent in
STREAM frames on all streams. A receiver advertises credit for a connection by
sending a MAX_DATA frame. A receiver maintains a cumulative sum of bytes
received on all contributing streams, which are used to check for flow control
violations. A receiver might use a sum of bytes consumed on all contributing
streams to determine the maximum data limit to be advertised.
## Edge Cases and Other Considerations
There are some edge cases which must be considered when dealing with stream and
connection level flow control. Given enough time, both endpoints must agree on
flow control state. If one end believes it can send more than the other end is
willing to receive, the connection will be torn down when too much data arrives.
Conversely if a sender believes it is blocked, while endpoint B expects more
data can be received, then the connection can be in a deadlock, with the sender
waiting for a MAX_DATA or MAX_STREAM_DATA frame which will never come.
On receipt of a RST_STREAM frame, an endpoint will tear down state for the
matching stream and ignore further data arriving on that stream. This could
result in the endpoints getting out of sync, since the RST_STREAM frame may have
arrived out of order and there may be further bytes in flight. The data sender
would have counted the data against its connection level flow control budget,
but a receiver that has not received these bytes would not know to include them
as well. The receiver must learn the number of bytes that were sent on the
stream to make the same adjustment in its connection flow controller.
To avoid this de-synchronization, a RST_STREAM sender MUST include the final
byte offset sent on the stream in the RST_STREAM frame. On receiving a
RST_STREAM frame, a receiver definitively knows how many bytes were sent on that
stream before the RST_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.
### Response to a RST_STREAM
RST_STREAM terminates one direction of a stream abruptly. Whether any action or
response can or should be taken on the data already received is an
application-specific issue, but it will often be the case that upon receipt of a
RST_STREAM an endpoint will choose to stop sending data in its own direction. If
the sender of a RST_STREAM wishes to explicitly state that no future data will
be processed, that endpoint MAY send a STOP_SENDING frame at the same time.
### Data Limit Increments {#fc-credit}
This document leaves when and how many bytes to advertise in a MAX_DATA or
MAX_STREAM_DATA 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, infrequent updates require
larger increments to limits if blocking is to be avoided. Thus, larger updates
require a receiver to commit to larger resource commitments. 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 that
it increases data limits based on a round-trip time estimate and the rate at
which the receiving application consumes data, similar to common TCP
implementations.
## Stream Limit Increment
As with flow control, this document leaves when and how many streams to make
available to a peer via MAX_STREAM_ID to implementations, but offers a few
considerations. MAX_STREAM_ID frames constitute minimal overhead, while
withholding MAX_STREAM_ID frames can prevent the peer from using the available
parallelism.
Implementations will likely want to increase the maximum stream ID as
peer-initiated streams close. A receiver MAY also advance the maximum stream ID
based on current activity, system conditions, and other environmental factors.
### Blocking on Flow Control {#blocking}
If a sender does not receive a MAX_DATA or MAX_STREAM_DATA frame when it has run
out of flow control credit, the sender will be blocked and SHOULD send a BLOCKED
or STREAM_BLOCKED frame. These frames are expected to be useful for debugging
at the receiver; they do not require any other action. A receiver SHOULD NOT
wait for a BLOCKED or STREAM_BLOCKED frame before sending MAX_DATA or
MAX_STREAM_DATA, since doing so will mean that a sender is unable to send for an
entire round trip.
For smooth operation of the congestion controller, it is generally considered
best to not let the sender go into quiescence if avoidable. 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 sender sends a single BLOCKED or STREAM_BLOCKED frame only once when it
reaches a data limit. A sender SHOULD NOT send multiple BLOCKED or
STREAM_BLOCKED frames for the same data limit, unless the original frame is
determined to be lost. Another BLOCKED or STREAM_BLOCKED frame can be sent
after the data limit is increased.
## Stream Final Offset {#final-offset}
The final offset is the count of the number of octets that are transmitted on a
stream. For a stream that is reset, the final offset is carried explicitly in
a RST_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.
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 RST_STREAM or
STREAM frame causes the final offset to change for a 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.
## Flow Control for Cryptographic Handshake {#flow-control-crypto}
Data sent in CRYPTO frames is not flow controlled in the same way as STREAM
frames. 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.
# 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, APPLICATION_CLOSE, or RST_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 or APPLICATION_CLOSE frame
({{frame-connection-close}}, {{frame-application-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_CLOSE frame. Errors that are specific to the transport, including
all those described in this document, are carried in a CONNECTION_CLOSE frame.
Other than the type of error code they carry, these frames are identical in
format and semantics.
A CONNECTION_CLOSE or APPLICATION_CLOSE frame could be sent in a packet that is
lost. An endpoint SHOULD be prepared to retransmit a packet containing either
frame type 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 CONNECTION_CLOSE
or APPLICATION_CLOSE 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 or APPLICATION_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 RST_STREAM frame
({{frame-rst-stream}}) with an appropriate error code to terminate just the
affected stream.
Other than STOPPING ({{solicited-state-transitions}}), RST_STREAM MUST be
instigated by the application and MUST carry 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.
## 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_ID_ERROR (0x4):
: An endpoint received a frame for a stream identifier that exceeded its
advertised maximum stream ID.
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 RST_STREAM
frame containing a final offset that was lower than the maximum offset of data
that was already received. Or an endpoint received a RST_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, an empty
STREAM frame that omitted the FIN flag, 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 RST_STREAM
({{frame-rst-stream}}) and APPLICATION_CLOSE ({{frame-application-close}})
frames.
There is no restriction on the use of the 16-bit error code space for
application protocols. However, QUIC reserves the error code with a value of 0
to mean STOPPING. The application error code of STOPPING (0) is used by the
transport to cancel a stream in response to receipt of a STOP_SENDING frame.
# 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 {{v4-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.
## Spoofed ACK Attack
An attacker might be able to receive an address validation token
({{address-validation}}) from the server and then release the IP address it
used to acquire that token. The attacker may, in the future, spoof this same
address (which now presumably addresses a different endpoint), and initiate a
0-RTT connection with a server on the victim's behalf. The attacker can then
spoof ACK frames to the server which cause the server to send excessive amounts
of data toward the new owner of the IP address.
There are two possible mitigations to this attack. The simplest one is that a
server can unilaterally create a gap in packet-number space. In the non-attack
scenario, the client will send an ACK frame with the larger value for largest
acknowledged. In the attack scenario, the attacker could acknowledge a packet
in the gap. If the server sees an acknowledgment for a packet that was never
sent, the connection can be aborted.
The second mitigation is that the server can require that acknowledgments for
sent packets match the encryption level of the sent packet. This mitigation is
useful if the connection has an ephemeral forward-secure key that is generated
and used for every new connection. If a packet sent is protected with a
forward-secure key, then any acknowledgments that are received for them MUST
also be forward-secure protected. Since the attacker will not have the forward
secure key, the attacker will not be able to generate forward-secure protected
packets with ACK frames.
## 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 endpoint might intentionally fragment the data on stream buffers
in order to cause disproportionate memory commitment. An adversarial endpoint
could open a stream and send some STREAM frames containing arbitrary fragments
of the stream content.
The attack 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 the stream fragmentation
attack. Mitigations could consist of avoiding over-committing memory, 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 {{stream-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 {{using-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 | initial_max_stream_data_bidi_local | {{transport-parameter-definitions}} |
| 0x0001 | initial_max_data | {{transport-parameter-definitions}} |
| 0x0002 | initial_max_bidi_streams | {{transport-parameter-definitions}} |
| 0x0003 | idle_timeout | {{transport-parameter-definitions}} |
| 0x0004 | preferred_address | {{transport-parameter-definitions}} |
| 0x0005 | max_packet_size | {{transport-parameter-definitions}} |
| 0x0006 | stateless_reset_token | {{transport-parameter-definitions}} |
| 0x0007 | ack_delay_exponent | {{transport-parameter-definitions}} |
| 0x0008 | initial_max_uni_streams | {{transport-parameter-definitions}} |
| 0x0009 | disable_migration | {{transport-parameter-definitions}} |
| 0x000a | initial_max_stream_data_bidi_remote | {{transport-parameter-definitions}} |
| 0x000b | initial_max_stream_data_uni | {{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_ID_ERROR | Invalid stream ID | {{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 packet number 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-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, RST_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 RST_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 RST_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)
- RST_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 RST_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.
