QUIC: A UDP-Based Multiplexed and Secure TransportGooglejri@google.comMozillamartin.thomson@gmail.com
Transport
QUICThis 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.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.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 transport for multiple applications.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, and data reliability.Accompanying documents describe QUIC’s loss detection and congestion control
, and the use of TLS 1.3 for key negotiation .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
when, and only when, they appear in all capitals, as shown here.Definitions of terms that are used in this document:
The endpoint initiating a QUIC connection.
The endpoint accepting incoming QUIC connections.
The client or server end of a connection.
A logical, bi-directional channel of ordered bytes within a QUIC connection.
A conversation between two QUIC endpoints with a single encryption context
that multiplexes streams within it.
The 64-bit unsigned number used as an identifier for a QUIC connection.
A well-formed UDP payload that can be parsed by a QUIC receiver.Packet and frame diagrams use the format described in Section 3.1 of
, with the following additional conventions:
Indicates that x is optional
Indicates that x is A bits long
Indicates that x is one of A, B, or C bits long
Indicates that x uses the variable-length encoding in
Indicates that x is variable-lengthThis section briefly describes QUIC’s key mechanisms and benefits. Key
strengths of QUIC include:Low-latency connection establishmentMultiplexing without head-of-line blockingAuthenticated and encrypted header and payloadRich signaling for congestion control and loss recoveryStream and connection flow controlConnection migration and resilience to NAT rebindingVersion negotiationQUIC relies on a combined cryptographic and transport handshake for
setting up a secure transport connection. QUIC connections are
expected to commonly use 0-RTT handshakes, meaning that for most QUIC
connections, data can be sent immediately following the client
handshake packet, without waiting for a reply from the server. QUIC
provides a dedicated stream (Stream ID 0) to be used for performing
the cryptographic handshake and QUIC options negotiation. The format
of the QUIC options and parameters used during negotiation are
described in this document, but the handshake protocol that runs on
Stream ID 0 is described in the accompanying cryptographic handshake
draft .When application messages are transported over TCP, independent application
messages can suffer from head-of-line blocking. When an application multiplexes
many streams atop TCP’s single-bytestream abstraction, a loss of a TCP segment
results in blocking of all subsequent segments until a retransmission arrives,
irrespective of the application streams that are encapsulated in subsequent
segments. QUIC ensures that lost packets carrying data for an individual stream
only impact that specific stream. Data received on other streams can continue
to be reassembled and delivered to the application.QUIC’s packet framing and acknowledgments carry rich information that help both
congestion control and loss recovery in fundamental ways. Each QUIC packet
carries a new packet number, including those carrying retransmitted data. This
obviates the need for a separate mechanism to distinguish acknowledgments for
retransmissions from those for original transmissions, avoiding TCP’s
retransmission ambiguity problem. QUIC acknowledgments also explicitly encode
the delay between the receipt of a packet and its acknowledgment being sent, and
together with the monotonically-increasing packet numbers, this allows for
precise network roundtrip-time (RTT) calculation. QUIC’s ACK frames support
multiple ACK blocks, so QUIC is more resilient to reordering than TCP with SACK
support, as well as able to keep more bytes on the wire when there is reordering
or loss.QUIC implements stream- and connection-level flow control. At a high level, a
QUIC receiver advertises the maximum amount of data that it is willing to
receive on each stream. As data is sent, received, and delivered on a
particular stream, the receiver sends MAX_STREAM_DATA frames that increase the
advertised limit for that stream, allowing the peer to send more data on that
stream.In addition to this stream-level flow control, QUIC implements connection-level
flow control to limit the aggregate buffer that a QUIC receiver is willing to
allocate to all streams on a connection. Connection-level flow control works in
the same way as stream-level flow control, but the bytes delivered and the
limits are aggregated across all streams.TCP headers appear in plaintext on the wire and are not authenticated, causing a
plethora of injection and header manipulation issues for TCP, such as
receive-window manipulation and sequence-number overwriting. While some of
these are mechanisms used by middleboxes to improve TCP performance, others are
active attacks. Even “performance-enhancing” middleboxes that routinely
interpose on the transport state machine end up limiting the evolvability of the
transport protocol, as has been observed in the design of MPTCP and
in its subsequent deployability issues.Generally, QUIC packets are always authenticated and the payload is typically
fully encrypted. The parts of the packet header which are not encrypted are
still authenticated by the receiver, so as to thwart any packet injection or
manipulation by third parties. Some early handshake packets, such as the
Version Negotiation packet, are not encrypted, but information sent in these
unencrypted handshake packets is later verified as part of cryptographic
processing.QUIC connections are identified by a Connection ID, a 64-bit unsigned number
randomly generated by the server. QUIC’s consistent connection ID allows
connections to survive changes to the client’s IP and port, such as those caused
by NAT rebindings or by the client changing network connectivity to a new
address. QUIC provides automatic cryptographic verification of a rebound client,
since the client continues to use the same session key for encrypting and
decrypting packets. The consistent connection ID can be used to allow migration
of the connection to a new server IP address as well, since the Connection ID
remains consistent across changes in the client’s and the server’s network
addresses.QUIC version negotiation allows for multiple versions of the protocol to be
deployed and used concurrently. Version negotiation is described in
.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.Version 0x00000001 of QUIC uses TLS as a cryptographic handshake protocol, as
described in .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.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 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
(). The long form allows for special packets - such as the
Version Negotiation packet - to be represented in this uniform fixed-length
packet format. A long header contains the following fields:
The most significant bit (0x80) of octet 0 (the first octet) is set to 1 for
long headers.
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 .
Octets 1 through 8 contain the connection ID. describes the
use of this field in more detail.
Octets 9 to 12 contain the selected protocol version. This field indicates
which version of QUIC is in use and determines how the rest of the protocol
fields are interpreted.
Octets 13 to 16 contain the packet number. describes the
use of packet numbers.
Octets from 17 onwards (the rest of QUIC packet) are the payload of the
packet.The following packet types are defined:TypeNameSection0x7FInitial0x7ERetry0x7DHandshake0x7C0-RTT ProtectedThe header form, packet type, connection ID, packet number and version fields of
a long header packet are version-independent. The types of packets defined in
are version-specific. See 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 short header can be used after the version and 1-RTT keys are negotiated.
This header form has the following fields:
The most significant bit (0x80) of octet 0 is set to 0 for the short header.
The second bit (0x40) of octet 0 indicates whether the Connection ID field is
omitted. If set to 0, then the Connection ID field is present; if set to 1,
the Connection ID field is omitted. The Connection ID field can
only be omitted if the omit_connection_id transport parameter
() is specified by the intended recipient
of the packet.
The third bit (0x20) 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 for details.
The remaining 5 bits of octet 0 include one of 32 packet types.
lists the types that are defined for short packets.
If the Omit Connection ID Flag is not set, a connection ID occupies octets
1 through 8 of the packet. See for more details.
The length of the packet number field depends on the packet type. This field
can be 1, 2 or 4 octets long depending on the short packet type.
Packets with a short header always include a 1-RTT protected payload.The packet type in a short header currently determines only the size of the
packet number field. Additional types can be used to signal the presence of
other fields.TypePacket Number Size0x1F1 octet0x1E2 octets0x1D4 octetsThe header form, omit connection ID flag, and connection ID of a short header
packet are version-independent. The remaining fields are specific to the
selected QUIC version. See for details on how packets
from different versions of QUIC are interpreted.A Version Negotiation packet is inherently not version-specific, and does not
use the packet headers defined above. 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.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:The value in the Unused field is selected randomly by the server. The Connection
ID field echoes the corresponding value from the triggering client packet. This
allows clients some assurance that the server received the packet and that the
Version Negotiation packet is in fact from the server. The Version field MUST
be set to 0x00000000. 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.See for a description of the version negotiation
process.Once version negotiation is complete, the cryptographic handshake is used to
agree on cryptographic keys. The cryptographic handshake is carried in Initial
(), Retry () and 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, handshake packets
are protected with a connection- and version-specific key, as described in
. This protection does not provide confidentiality or integrity
against on-path attackers, but provides some level of protection against
off-path attackers.The Initial packet uses long headers with a type value of 0x7F. It carries the
first cryptographic handshake message sent by the client.The client populates the connection ID field with randomly selected values,
unless it has received a packet from the server. If the client has received a
packet from the server, the connection ID field uses the value provided by the
server.The first Initial packet that is sent by a client contains a randomized packet
number. All subsequent packets contain a packet number that is incremented by
one, see ().The payload of a Initial packet consists of a STREAM frame (or frames)
for stream 0 containing a cryptographic handshake message, with enough PADDING
frames that the packet is at least 1200 octets (see ). The
stream in this packet always starts at an offset of 0 (see )
and the complete cryptographic handshake message MUST fit in a single packet
(see ).The client uses 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, this includes
the packets sent after receiving a Version Negotiation () or
Retry packet ().A Retry packet uses long headers with a type value of 0x7E. It carries
cryptographic handshake messages and acknowledgments. It is used by a server
that wishes to perform a stateless retry (see ).The server includes a connection ID of its choice in the connection ID field.
The client MUST use this connection ID for any subsequent packets that it sends.The packet number field echoes the packet number field from the triggering
client packet.A Retry packet is never explicitly acknowledged in an ACK frame
by a client. Receiving another Initial packet implicitly acknowledges a Retry
packet.After receiving a Retry packet, the client uses a new
Initial packet containing the next cryptographic handshake message. The client
retains the state of its cryptographic handshake, but discards all transport
state. The Initial packet that is generated in response to a Retry packet
includes STREAM frames on stream 0 that start again at an offset of 0.Continuing the cryptographic handshake is necessary to ensure that an attacker
cannot force a downgrade of any cryptographic parameters. In addition to
continuing the cryptographic handshake, the client MUST remember the results of
any version negotiation that occurred (see ). The client
MAY also retain any observed RTT or congestion state that it has accumulated for
the flow, but other transport state MUST be discarded.The payload of the Retry packet contains a single STREAM frame
on stream 0 with offset 0 containing the server’s cryptographic stateless retry
material. It MUST NOT contain any other frames. The next STREAM frame sent by
the server will also start at stream offset 0.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.The connection ID field in a Handshake packet contains a connection ID
that is chosen by the server (see ).The first Handshake packet sent by a server contains a randomized packet number.
This value is increased for each subsequent packet sent by the server as
described in . The client increments the packet number from
its previous packet by one for each Handshake packet that it sends (which might
be an Initial, 0-RTT Protected, or Handshake packet).The payload of this packet contains STREAM frames and could contain PADDING and
ACK frames.Packets that are protected with 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.Packets protected with 0-RTT keys use a type value of 0x7C. The connection ID
field for a 0-RTT packet is selected by the client.The client can send 0-RTT packets after receiving a 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 connection ID selected by the client for 0-RTT packets, see
.The version field for protected packets is the current QUIC version.The packet number field contains a packet number, which increases with each
packet sent, see for details.The payload is protected using authenticated encryption. describes
packet protection in detail. After decryption, the plaintext consists of a
sequence of frames, as described in .QUIC connections are identified by their 64-bit Connection ID. All long headers
contain a Connection ID. Short headers indicate the presence of a Connection ID
using the Omit Connection ID flag. When present, the Connection ID is in the
same location in all packet headers, making it straightforward for middleboxes,
such as load balancers, to locate and use it.The client MUST choose a random connection ID and use it in Initial packets
() and 0-RTT packets ().When the server receives a Initial packet and decides to proceed with the
handshake, it chooses a new value for the connection ID and sends that in a
Retry () or Handshake () packet. The server
MAY choose to use the value that the client initially selects.Once the client receives the connection ID that the server has chosen, it MUST
use it for all subsequent Handshake () and 1-RTT
() packets but not for 0-RTT packets ().Server’s Version Negotiation () and Retry ()
packets MUST use connection ID selected by the client.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. The packet number
for sending MUST increase by at least one after sending any packet, unless
otherwise specified (see ).A QUIC endpoint MUST NOT reuse a packet number within the same 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; a server MAY send a Stateless
Reset () in response to further packets that it receives.For the packet header, the number of bits required to represent the packet
number are reduced by including only the least significant bits of the packet
number. The actual packet number for each packet is reconstructed at the
receiver based on the largest packet number received on a successfully
authenticated packet.A 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 16-bit
value of 0x1f94 will be decoded as 0xaa831f94.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 0x6b4264 requires a 16-bit or larger packet
number encoding; whereas a 32-bit packet number is needed to send a packet with
a number of 0x6bc107.Version Negotiation () and Retry () packets
have special rules for populating the packet number field.The initial value for packet number MUST be selected randomly from a range
between 0 and 2^32 - 1025 (inclusive). This value is selected so that Initial
and Handshake packets exercise as many possible values for the Packet Number
field as possible.Limiting the range allows both for loss of packets and for any stateless
exchanges. Packet numbers are incremented for subsequent packets, but packet
loss and stateless handling can both mean that the first packet sent by an
endpoint isn’t necessarily the first packet received by its peer. The first
packet received by a peer cannot be 2^32 or greater or the recipient will
incorrectly assume a packet number that is 2^32 values lower and discard the
packet.Use of a secure random number generator is not necessary for
generating the initial packet number, nor is it necessary that the value be
uniformly distributed.Between different versions the following things are guaranteed to remain
constant:the location of the header form flag,the location of the Omit Connection ID flag in short headers,the location and size of the Connection ID field in both header forms,the location and size of the Version field in long headers,the format and semantics of the Version Negotiation packet.Implementations MUST assume that an unsupported version uses an unknown packet
format. All other fields MUST be ignored when processing a packet that contains
an unsupported version.The payload of all packets, after removing packet protection, consists of a
sequence of frames, as shown in . Version Negotiation and
Stateless Reset do not contain frames.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 byte, indicating its type,
followed by additional type-dependent fields:Frame types are listed in . Note that the Frame Type byte in
STREAM and ACK frames is used to carry other frame-specific flags. For all
other frames, the Frame Type byte simply identifies the frame. These frames are
explained in more detail as they are referenced later in the document.Type ValueFrame Type NameDefinition0x00PADDING0x01RST_STREAM0x02CONNECTION_CLOSE0x03APPLICATION_CLOSE0x04MAX_DATA0x05MAX_STREAM_DATA0x06MAX_STREAM_ID0x07PING0x08BLOCKED0x09STREAM_BLOCKED0x0aSTREAM_ID_BLOCKED0x0bNEW_CONNECTION_ID0x0cSTOP_SENDING0x0dPONG0x0eACK0x10 - 0x17STREAMA 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 . Once established, a connection may migrate to a
different IP or port at either endpoint, due to NAT rebinding or mobility, as
described in . Finally a connection may be terminated by either
endpoint, as described in .Incoming packets are classified on receipt. Packets can either be associated
with an existing connection, be discarded, or - for servers - potentially create
a new connection.Packets that can be associated with an existing connection are handled according
to the current state of that connection. Packets are associated with existing
connections using connection ID if it is present; this might include connection
IDs that were advertised using NEW_CONNECTION_ID ().
Packets without connection IDs and long-form packets for connections that have
incomplete cryptographic handshakes are associated with an existing connection
using the tuple of source and destination IP addresses and ports.A packet that uses the short header could be associated with an existing
connection with an incomplete cryptographic handshake. Such a packet could be a
valid packet that has been reordered with respect to the long-form packets that
will complete the cryptographic handshake. This might happen after the final
set of cryptographic handshake messages from either peer. These packets are
expected to be correlated with a connection using the tuple of IP addresses and
ports. Packets that might be reordered in this fashion SHOULD be buffered in
anticipation of the handshake completing.0-RTT packets might be received prior to a Client Initial packet at a server.
If the version of these packets is acceptable to the server, it MAY buffer these
packets in anticipation of receiving a reordered Client Initial packet.Buffering ensures that data is not lost, which improves performance; conversely,
discarding these packets could create false loss signals for the congestion
controllers. However, limiting the number and size of buffered packets might be
needed to prevent exposure to denial of service.For clients, any packet that cannot be associated with an existing connection
SHOULD be discarded if it is not buffered. Discarded packets MAY be logged for
diagnostic or security purposes.For servers, packets that aren’t associated with a connection potentially create
a new connection. However, only packets that use the long packet header and
that are at least the minimum size defined for the protocol version can be
initial packets. A server MAY discard packets with a short header or packets
that are smaller than the smallest minimum size for any version that the server
supports. A server that discards a packet that cannot be associated with a
connection MAY also generate a stateless reset ().This version of QUIC defines a minimum size for initial packets of 1200 octets
(see ). Versions of QUIC that define smaller minimum initial
packet sizes need to be aware that initial packets will be discarded without
action by servers that only support versions with larger minimums. Clients that
support multiple QUIC versions can avoid this problem by ensuring that they
increase the size of their initial packets to the largest minimum size across
all of the QUIC versions they support. Servers need to recognize initial
packets that are the minimum size of all QUIC versions they support.QUIC’s connection establishment begins with version negotiation, since all
communication between the endpoints, including packet and frame formats, relies
on the two endpoints agreeing on a version.A QUIC connection begins with a client sending an Initial packet
(). The details of the handshake mechanisms are described in
, but any Initial packet sent from the client to the server MUST
use the long header format - which includes the version of the protocol being
used - and they MUST be padded to at least 1200 octets.The server receives this packet and determines whether it potentially creates a
new connection (see ). If the packet might generate a new
connection, the server then checks whether it understands the version that the
client has selected.If the packet contains a version that is acceptable to the server, the server
proceeds with the handshake (). This commits the server to the
version that the client selected.If the version selected by the client is not acceptable to the server, the
server responds with a Version Negotiation packet (). This
includes a list of versions that the server will accept.A server sends a Version Negotiation packet for any packet with an unacceptable
version if that packet could create a new connection. This allows a server to
process packets with unsupported versions without retaining state. Though
either the Client 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.When the client receives a Version Negotiation packet, it first checks that the
connection ID matches the connection ID 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 Client 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.Version negotiation packets have no cryptographic protection. The
result of the negotiation MUST be revalidated as part of the
cryptographic handshake (see ).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 ) 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 ) 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.QUIC relies on a combined cryptographic and transport handshake to minimize
connection establishment latency. QUIC allocates stream 0 for the cryptographic
handshake. Version 0x00000001 of QUIC uses TLS 1.3 as described in
; a different QUIC version number could indicate that a different
cryptographic handshake protocol is in use.QUIC provides this stream with reliable, ordered delivery of data. In return,
the cryptographic handshake provides QUIC with: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, and1-RTT keys have forward secrecyauthenticated values for the transport parameters of the peer (see
)authenticated confirmation of version negotiation (see )authenticated negotiation of an application protocol (TLS uses ALPN
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 )The initial cryptographic handshake message 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. Reassembling messages requires that a server maintain state
prior to establishing a connection, exposing the server to a denial of service
risk.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.Details of how TLS is integrated with QUIC is provided in more detail in
.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
. This is described using the presentation
language from Section 3 of .;
} 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;
]]>The extension_data field of the quic_transport_parameters extension defined in
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 ) before the connection
establishment is considered properly complete.Definitions for each of the defined transport parameters are included in
. 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.An endpoint MUST include the following parameters in its encoded
TransportParameters:
The initial stream maximum data parameter contains the initial value for the
maximum data that can be sent on any newly created stream. This parameter is
encoded as an unsigned 32-bit integer in units of octets. This is equivalent
to an implicit MAX_STREAM_DATA frame () being sent on
all streams immediately after opening.
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 () for the connection immediately after
completing the handshake.
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).A server MUST include the following transport parameters:
The Stateless Reset Token is used in verifying a stateless reset, see
. This parameter is a sequence of 16 octets.A client MUST NOT include a stateless reset token. A server MUST treat receipt
of a stateless_reset_token transport parameter as a connection error of type
TRANSPORT_PARAMETER_ERROR.An endpoint MAY use the following transport parameters:
The initial maximum stream ID parameter contains the initial maximum stream
number the peer may initiate for bidirectional streams, encoded as an unsigned
32-bit integer. This value MUST be a valid bidirectional stream ID for a
peer-initiated stream (that is, the two least significant bits are set to 0 by
a server and to 1 by a client). If an invalid value is provided, the
recipient MUST generate a connection error of type TRANSPORT_PARAMETER_ERROR.
Setting this parameter is equivalent to sending a MAX_STREAM_ID
() immediately after completing the handshake. The
maximum bidirectional stream ID is set to 0 if this parameter is absent,
preventing the creation of new bidirectional streams until a MAX_STREAM_ID
frame is sent. Note that a default value of 0 does not prevent the
cryptographic handshake stream (that is, stream 0) from being used.
The initial maximum stream ID parameter contains the initial maximum stream
number the peer may initiate for unidirectional streams, encoded as an
unsigned 32-bit integer. The value MUST be a valid unidirectional ID for the
recipient (that is, the two least significant bits are set to 2 by a server
and to 3 by a client). If an invalid value is provided, the recipient MUST
generate a connection error of type TRANSPORT_PARAMETER_ERROR. Setting this
parameter is equivalent to sending a MAX_STREAM_ID ()
immediately after completing the handshake. The maximum unidirectional stream
ID is set to 0 if this parameter is absent, preventing the creation of new
unidirectional streams until a MAX_STREAM_ID frame is sent.
The omit connection identifier parameter indicates that packets sent to the
endpoint that advertises this parameter MAY omit the connection ID in packets
using short header format. This can be used by an endpoint where it knows
that source and destination IP address and port are sufficient for it to
identify a connection. This parameter is zero length. Absence of this
parameter means that the connection ID MUST be present in every packet sent to
this endpoint.
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
().
An 8-bit unsigned integer value indicating an exponent used to decode the ACK
Delay field in the ACK frame, see . 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, Handshake, and
Retry packets. Values above 20 are invalid.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 or
initial_max_stream_data that are smaller than the remembered value of those
parameters. Similarly, a server MUST NOT reduce the value of
initial_max_stream_id_bidi or initial_max_stream_id_uni.Omitting or setting a zero value for certain transport parameters can result in
0-RTT data being enabled, but not usable. The following transport parameters
SHOULD be set to non-zero values for 0-RTT: initial_max_stream_id_bidi,
initial_max_stream_id_uni, initial_max_data, initial_max_stream_data.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 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
.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
).The cryptographic handshake provides integrity protection for the negotiated
version as part of the transport parameters (see ). 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 , 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 () 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.A server can process an initial cryptographic handshake messages from a client
without committing any state. This allows a server to perform 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 (). This packet
causes a client to reset its transport state and to continue the connection
attempt with new connection state while maintaining the state of the
cryptographic handshake.A server MUST NOT send multiple Retry packets in response to a client handshake
packet. Thus, any cryptographic handshake message that is sent MUST fit within
a single packet.In TLS, the Retry packet type is used to carry the HelloRetryRequest message.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 padded to at least 1200 octets. 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
cryptographic handshake successfully completes. 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 during the establishment of a
connection. TLS provides the tools that support the feature, but basic
validation is performed by the core transport protocol.A different type of source address validation is performed after a connection
migration, see .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 cannot be easily guessed (see ), if the client is able
to return that token, it proves to the server that it received the token.During the processing of the cryptographic handshake messages from a client, TLS
will request that QUIC make a decision about whether to proceed based on the
information it has. TLS will provide QUIC with any token that was provided by
the client. For an initial packet, QUIC can decide to abort the connection,
allow it to proceed, or request address validation.If QUIC decides to request address validation, it provides the cryptographic
handshake with a token. The contents of this token are consumed by the server
that generates the token, so 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 cryptographic handshake is responsible for enacting validation by sending
the address validation token to the client. A legitimate client will include a
copy of the token when it attempts to continue the handshake. The cryptographic
handshake extracts the token then asks QUIC a second time whether the token is
acceptable. In response, QUIC 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.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.A different type of token is needed when resuming. Unlike the token that is
created during a handshake, 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. 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.This token can be provided to the cryptographic handshake immediately after
establishing a connection. QUIC might also generate an updated token if
significant time passes or the client address changes for any reason (see
). The cryptographic handshake is responsible for providing the
client with the token. In TLS the token is included in the ticket that is used
for resumption and 0-RTT, which is carried in a NewSessionTicket message.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.In TLS the address validation token is often bundled with the information that
TLS requires, such as the resumption secret. In this case, adding integrity
protection can be delegated to the cryptographic handshake protocol, avoiding
redundant protection. If integrity protection is delegated to the cryptographic
handshake, an integrity failure will result in immediate cryptographic handshake
failure. If integrity protection is performed by QUIC, QUIC MUST abort the
connection if the integrity check fails with a PROTOCOL_VIOLATION error code.QUIC connections are identified by their 64-bit Connection ID. QUIC’s
consistent connection ID allows connections to survive changes to the client’s
IP and/or port, such as those caused by client or server migrating to a new
network. Connection migration allows a client to retain any shared state with a
connection when they move networks. This includes state that can be hard to
recover such as outstanding requests, which might otherwise be lost with no easy
way to retry them.An endpoint that receives packets that contain a source IP address and port that
has not yet been used can start sending new packets with those as a destination
IP address and port. Packets exchanged between endpoints can then follow the
new path.Due to variations in path latency or packet reordering, packets from different
source addresses might be reordered. The packet with the highest packet number
MUST be used to determine which path to use. Endpoints also need to be prepared
to receive packets from an older source address.An endpoint MUST validate that its peer can receive packets at the new address
before sending any significant quantity of data to that address, or it risks
being used for denial of service. See for details.Using a stable connection ID on multiple network paths allows a passive observer
to correlate activity between those paths. A client that moves between networks
might not wish to have their activity correlated by any entity other than a
server. The NEW_CONNECTION_ID message can be sent by a server to provide an
unlinkable connection ID for use in case the client wishes to explicitly break
linkability between two points of network attachment.A client might need to send packets on multiple networks without receiving any
response from the server. To ensure that the client is not linkable across each
of these changes, a new connection ID and packet number gap are needed for each
network. To support this, a server sends multiple NEW_CONNECTION_ID messages.
Each NEW_CONNECTION_ID is marked with a sequence number. Connection IDs MUST be
used in the order in which they are numbered.A client which wishes to break linkability upon changing networks MUST use the
connection ID provided by the server as well as incrementing the packet sequence
number by an externally unpredictable value computed as described in
. Packet number gaps are cumulative. A client might skip
connection IDs, but it MUST ensure that it applies the associated packet number
gaps for connection IDs that it skips in addition to the packet number gap
associated with the connection ID that it does use.A server that receives a packet that is marked with a new connection ID recovers
the packet number by adding the cumulative packet number gap to its expected
packet number. A server SHOULD discard packets that contain a smaller gap than
it advertised.For instance, a server might provide a packet number gap of 7 associated with a
new connection ID. If the server received packet 10 using the previous
connection ID, it should expect packets on the new connection ID to start at 18.
A packet with the new connection ID and a packet number of 17 is discarded as
being in error.In order to avoid linkage, the packet number gap MUST be externally
indistinguishable from random. The packet number gap for a connection
ID with sequence number is computed by encoding the sequence number
as a 32-bit integer in big-endian format, and then computing:The output of HKDF-Expand-Label is interpreted as a big-endian
number. “packet_number_secret” is derived from the TLS key exchange,
as described in Section 5.6 of .An endpoint that receives a packet from a new remote IP address and port (or
just a new remote port) on packets from its peer is likely seeing a connection
migration at the peer.However, it is also possible that the peer is spoofing its source address in
order to cause the endpoint to send excessive amounts of data to an unwilling
host. If the endpoint sends significantly more data than the peer, connection
migration might be used to amplify the volume of data that an attacker can
generate toward a victim.Thus, when seeing a new remote transport address, an endpoint MUST verify that
its peer can receive and respond to packets at that new address. By providing
copies of the data that it receives, the peer proves that it is receiving
packets at the new address and consents to receive data.Prior to validating the new remote address, and endpoint MUST limit the amount
of data and packets that it sends to its peer. At a minimum, this needs to
consider the possibility that packets are sent without congestion feedback.Once a connection is established, address validation is relatively simple (see
for the process that is used during the handshake). An
endpoint validates a remote address by sending a PING frame containing a payload
that is hard to guess. This frame MUST be sent in a packet that is sent to the
new address. Once a PONG frame containing the same payload is received, the
address is considered to be valid. The PONG frame can use any path on its
return. A PING frame containing 12 randomly generated octets is
sufficient to ensure that it is easier to receive the packet than it is to guess
the value correctly.If the PING frame is determined to be lost, a new PING frame SHOULD be
generated. This PING frame MUST include a new Data field that is similarly
difficult to guess.If validation of the new remote address fails, after allowing enough time for
possible loss and recovery of packets carrying PING and PONG frames, the
endpoint MUST terminate the connection. When setting this timer,
implementations are cautioned that the new path could have a longer round trip
time than the original. The endpoint MUST NOT send a CONNECTION_CLOSE frame in
this case; it has to assume that the remote peer does not want to receive any
more packets.If the remote address is validated successfully, the endpoint MAY increase the
rate that it sends on the new path using the state from the previous path. The
capacity available on the new path might not be the same as the old path. An
endpoint MUST NOT restore its send rate unless it is reasonably sure that the
path is the same as the previous path. For instance, a change in only 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 reduce
send rates appropriately.After verifying an address, the endpoint SHOULD update any address validation
tokens () that it has issued to its peer if those are no
longer valid based on the changed address.Address validation using the PING frame MAY be used at any time by either peer.
For instance, an endpoint might check that a peer is still in possession of its
address after a period of quiescence.Upon seeing a connection migration, an endpoint that sees a new address MUST
abandon any address validation it is performing with other addresses on the
expectation that the validation is likely to fail. Abandoning address
validation primarily means not closing the connection when a PONG frame is not
received, but it could also mean ceasing retransmissions of the PING frame. An
endpoint that doesn’t retransmit a PING frame might receive a PONG frame, which
it MUST ignore.A connection migration could be triggered by an attacker that is able to capture
and forward a packet such that it arrives before the legitimate copy of that
packet. Such a packet will appear to be a legitimate connection migration and
the legitimate copy will be dropped as a duplicate.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 PING frame that is sent to it, even if it wanted to.
Such a spurious connection migration could result in the connection being
dropped when the source address validation fails. This grants an attacker the
ability to terminate the connection.Receipt of packets with higher packet numbers from the legitimate address will
trigger another connection migration. This will cause the validation of the
address of the spurious migration to be abandoned.To ensure that a peer sends packets from the legitimate address before the
validation of the new address can fail, an endpoint SHOULD attempt to validate
the old remote address before attempting to validate the new address. If the
connection migration is spurious, then the legitimate address will be used to
respond and the connection will migrate back to the old address.As with any address validation, packets containing retransmissions of the PING
frame validating an address MUST be sent to the address being validated.
Consequently, during a migration of a peer, an endpoint could be sending to
multiple remote addresses.An endpoint MAY abandon address validation for an address that it considers to
be already valid. That is, if successive connection migrations occur in quick
succession with the final remote address being identical to the initial remote
address, the endpoint MAY abandon address validation for that address.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 ()immediate close ()stateless reset ()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 .An endpoint enters a closing period after initiating an immediate close
() and optionally after an idle timeout ().
While closing, an endpoint MUST NOT send packets unless they contain a
CONNECTION_CLOSE or APPLICATION_CLOSE frame (see 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
() 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
connection 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 as though the address were not validated (see ).
A server in the closing state MAY instead choose to discard packets received
from a new source address.A connection that remains idle for longer than the idle timeout (see
) is closed. A connection enters the
draining state when the idle timeout expires.The time at which an idle timeout takes effect won’t be perfectly synchronized
on both endpoints. An endpoint that sends packets near the end of an idle
period could have those packets discarded if its peer enters the draining state
before the packet is received.An endpoint sends a closing frame, either CONNECTION_CLOSE or APPLICATION_CLOSE,
to terminate the connection immediately. Either 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.
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.A stateless reset is provided as an option of last resort for a server that does
not have access to the state of a connection. A server crash or outage might
result in clients continuing to send data to a server that is unable to properly
continue the connection. A server 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, the server sends a stateless_reset_token value during
the handshake in the transport parameters. This value is protected by
encryption, so only client and server know this value.A server that receives packets that it cannot process sends a packet in the
following layout:A server copies the connection ID field from the packet that triggers the
stateless reset. A server omits the connection ID if explicitly configured to
do so, or if the client packet did not include a connection ID.The Packet Number field is set to a randomized value. The server SHOULD send a
packet with a short header and a type of 0x1F. This produces the shortest
possible packet number encoding, which minimizes the perceived gap between the
last packet that the server sent and this packet. A server MAY use a different
short header type, indicating a different packet number length, but a longer
packet number encoding might allow this message to be identified as a stateless
reset more easily using heuristics.After the Packet Number, the server pads the message with an arbitrary
number of octets containing random values.Finally, the last 16 octets of the packet are set to the value of the Stateless
Reset Token.This design ensures that a stateless reset packet is - to the extent possible -
indistinguishable from a regular packet.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. A server that
supports multiple versions of QUIC needs to generate a stateless reset that will
be accepted by clients that support any version that the server 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.A client detects a potential stateless reset when a packet with a short header
either cannot be decrypted or is marked as a duplicate packet. The client then
compares the last 16 octets of the packet with the Stateless Reset Token
provided by the server in its transport parameters. If these values are
identical, the client MUST enter the draining period and not send any further
packets on this connection. If the comparison fails, the packet can be
discarded.The stateless reset token MUST be difficult to guess. In order to create a
Stateless Reset Token, a server could randomly generate a secret
for every connection that it creates. However, this presents a coordination
problem when there are multiple servers in a cluster or a storage problem for a
server 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 three inputs: the static key, a the connection ID for the connection
(see ), and an identifier for the server instance. A server
could use HMAC (for example, HMAC(static_key, server_id ||
connection_id)) or HKDF (for example, using the static key as input
keying material, with server and connection identifiers as salt). The output of
this function is truncated to 16 octets to produce the Stateless Reset Token
for that connection.A server that loses state can use the same method to generate a valid Stateless
Reset Secret. The connection ID comes from the packet that the server receives.This design relies on the client always sending a connection ID in its packets
so that the server can use the connection ID from a packet to reset the
connection. A server that uses this design cannot allow clients to omit a
connection ID (that is, it cannot use the truncate_connection_id transport
parameter ).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 server instance, connection
ID, and static key cannot occur for another connection. A connection ID from a
connection that is reset by revealing the Stateless Reset Token cannot be
reused for new connections at the same server without first changing to use a
different static key or server identifier.Note that Stateless Reset messages do not have any cryptographic protection.As described in , Regular packets contain one or more frames.
We now describe the various QUIC frame types that can be present in a Regular
packet. The use of these frames and various frame header bits are described in
subsequent sections.QUIC frames use a common variable-length encoding for all 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. summarizes the
encoding properties.2BitLengthUsable BitsRange00160-63012140-16383104300-1073741823118620-4611686018427387903For 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 () are described using integers, but do not use this
encoding.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.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.The RST_STREAM frame is as follows:The fields are:
A variable-length integer encoding of the Stream ID of the stream being
terminated.
A 16-bit application protocol error code (see ) which
indicates why the stream is being closed.
A variable-length integer indicating the absolute byte offset of the end of
data written on this stream by the RST_STREAM sender.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:The fields of a CONNECTION_CLOSE frame are as follows:
A 16-bit error code which indicates the reason for closing this connection.
CONNECTION_CLOSE uses codes from the space defined in
(APPLICATION_CLOSE uses codes from the application protocol error code space,
see ).
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.
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 .An APPLICATION_CLOSE frame (type=0x03) uses the same format as the
CONNECTION_CLOSE frame (), except that it uses error
codes from the application protocol error code space ()
instead of the transport error code space.Other than the error code space, the format and semantics of the
APPLICATION_CLOSE frame are identical to the CONNECTION_CLOSE frame.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:The fields in the MAX_DATA frame are as follows:
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, with the exception of
data on stream 0. The sum of the largest received offsets on all streams -
including streams in terminal states, but excluding stream 0 - 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 ).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.The frame is as follows:The fields in the MAX_STREAM_DATA frame are as follows:
The stream ID of the stream that is affected encoded as a variable-length
integer.
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 ).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:The fields in the MAX_STREAM_ID frame are as follows:
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
).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 a variable-length payload.
This 8-bit value describes the length of the Data field.
This variable-length field contains arbitrary data.A PING frame with an empty Data field causes the packet containing it to be
acknowledged as normal. No other action is required of the recipient.An empty 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.If the Data field is not empty, the recipient of this frame MUST generate a PONG
frame () containing the same Data. A PING frame with data is not
appropriate for use in keeping a connection alive, because the PONG frame
elicits an acknowledgement, causing the sender of the original PING to send two
packets.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
). However, state in middleboxes might time out earlier than
that. Though REQ-5 in 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.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 ).
BLOCKED frames can be used as input to tuning of flow control algorithms (see
).The BLOCKED frame is as follows:The BLOCKED frame contains a single field.
A variable-length integer indicating the connection-level offset at which
the blocking occurred.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 ().The STREAM_BLOCKED frame is as follows:The STREAM_BLOCKED frame contains two fields:
A variable-length integer indicating the stream which is flow control blocked.
A variable-length integer indicating the offset of the stream at which the
blocking occurred.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
). 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:The STREAM_ID_BLOCKED frame contains a single field.
A variable-length integer indicating the highest stream ID that the sender
was permitted to open.A server sends a NEW_CONNECTION_ID frame (type=0x0b) to provide the client with
alternative connection IDs that can be used to break linkability when migrating
connections (see ).The NEW_CONNECTION_ID is as follows:The fields are:
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.
A 64-bit connection ID.
A 128-bit value that will be used to for a stateless reset when the associated
connection ID is used (see ).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.The STOP_SENDING frame is as follows:The fields are:
A variable-length integer carrying the Stream ID of the stream being ignored.
A 16-bit, application-specified reason the sender is ignoring the stream (see
).The PONG frame (type=0x0d) is sent in response to a PING frame that contains
data. Its format is identical to the PING frame ().An endpoint that receives an unsolicited PONG frame - that is, a PONG frame
containing a payload that is empty MUST generate a connection error of type
FRAME_ERROR, indicating the PONG frame (that is, 0x10d). If the content of a
PONG frame does not match the content of a PING frame previously sent by the
endpoint, the endpoint MAY generate a connection error of type UNSOLICITED_PONG.Receivers send ACK frames (type=0xe) to inform senders which packets they have
received and processed. A sent packet that has never been acknowledged is
missing. The ACK frame contains any number of ACK blocks. ACK blocks are
ranges of acknowledged packets.Unlike TCP SACKs, QUIC acknowledgements are irrevocable. Once a packet has
been acknowledged, even if it does not appear in a future ACK frame,
it remains acknowledged.A client MUST NOT acknowledge Version Negotiation or Retry packets. These
packet types contain packet numbers selected by the client, not the server.A sender MAY intentionally skip packet numbers to introduce entropy into the
connection, to avoid opportunistic acknowledgement attacks. The sender SHOULD
close the connection if an unsent packet number is acknowledged. The format of
the ACK frame is efficient at expressing even long blocks of missing packets,
allowing for large, unpredictable gaps.An ACK frame is shown below.The fields in the ACK frame are as follows:
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.
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
). Scaling in this fashion allows for a
larger range of values with a shorter encoding at the cost of lower
resolution.
The number of Additional ACK Block (and Gap) fields after the First ACK Block.
Contains one or more blocks of packet numbers which have been successfully
received, see .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.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: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: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_ERROR
indicating an error in an ACK frame (that is, 0x10d).The fields in the ACK Block Section are:
A variable-length integer indicating the number of contiguous packets
preceding the Largest Acknowledged that are being acknowledged.
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.
A variable-length integer indicating the number of contiguous acknowledged
packets preceding the largest packet number, as determined by the
preceding Gap.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 ACK
frame 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.A receiver that is only sending ACK frames will not receive acknowledgments for
its packets. Sending an occasional MAX_DATA or MAX_STREAM_DATA frame as data is
received will ensure that acknowledgements are generated by a peer. Otherwise,
an endpoint MAY send a PING frame once per RTT to solicit an acknowledgment.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
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 that acknowledge protected packets MUST be carried in a packet that
has an equivalent or greater level of packet protection.Packets that are protected with 1-RTT keys MUST be acknowledged in packets that
are also protected with 1-RTT keys.A packet that is not protected and claims to acknowledge a packet number that
was sent with packet protection is not valid. An unprotected packet that
carries acknowledgments for protected packets MUST be discarded in its entirety.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.Unprotected packets, such as those that carry the initial cryptographic
handshake messages, MAY be acknowledged in unprotected packets. Unprotected
packets are vulnerable to falsification or modification. Unprotected packets
can be acknowledged along with protected packets in a protected packet.An endpoint SHOULD acknowledge packets containing cryptographic handshake
messages in the next unprotected packet that it sends, unless it is able to
acknowledge those packets in later packets protected by 1-RTT keys. At the
completion of the cryptographic handshake, both peers send unprotected packets
containing cryptographic handshake messages followed by packets protected by
1-RTT keys. An endpoint SHOULD acknowledge the unprotected packets that complete
the cryptographic handshake in a protected packet, because its peer is
guaranteed to have access to 1-RTT packet protection keys.For instance, a server acknowledges a TLS ClientHello in the packet that carries
the TLS ServerHello; similarly, a client can acknowledge a TLS HelloRetryRequest
in the packet containing a second TLS ClientHello. The complete set of server
handshake messages (TLS ServerHello through to Finished) might be acknowledged
by a client in protected packets, because it is certain that the server is able
to decipher the packet.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.A STREAM frame is shown below.The STREAM frame contains the following fields:
A variable-length integer indicating the stream ID of the stream (see
).
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.
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.
The bytes from the designated stream to be delivered.A stream frame’s Stream Data MUST NOT be empty, unless the FIN bit is set. When
the FIN flag is sent on an empty STREAM frame, 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.A sender bundles one or more frames in a Regular QUIC packet (see ).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 heuristics about expected application sending behavior 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.Regular QUIC packets are “containers” of frames; a packet is never retransmitted
whole. How an endpoint handles the loss of the frame depends on the type of the
frame. Some frames are simply retransmitted, some have their contents moved to
new frames, and others are never retransmitted.When a packet is detected as lost, the sender re-sends any frames as necessary:All application data sent in STREAM frames MUST be retransmitted, unless the
endpoint has sent a RST_STREAM for that stream. When an endpoint sends a
RST_STREAM frame, data outstanding on that stream SHOULD NOT be retransmitted,
since subsequent data on this stream is expected to not be delivered by the
receiver.ACK and PADDING frames MUST NOT be retransmitted. ACK frames
containing updated information will be sent as described in .STOP_SENDING frames MUST be retransmitted until the receive stream enters
either a “Data Recvd” or “Reset Recvd” state. See
.The most recent MAX_STREAM_DATA frame for a stream MUST be retransmitted until
the receive stream enters a “Size Known” state. Any previous unacknowledged
MAX_STREAM_DATA frame for the same stream SHOULD NOT be retransmitted since a
newer MAX_STREAM_DATA frame for a stream obviates the need for delivering
older ones. Similarly, the most recent MAX_DATA frame MUST be retransmitted;
previous unacknowledged ones SHOULD NOT be retransmitted.BLOCKED, STREAM_BLOCKED, and STREAM_ID_BLOCKED frames SHOULD be retransmitted
if the sender is still blocked on the same limit. If the limit has been
increased since the frame was originally sent, the frame SHOULD NOT be
retransmitted.All other frames MUST be retransmitted.Upon detecting losses, a sender MUST take appropriate congestion control action.
The details of loss detection and congestion control are described in
.A packet MUST NOT be acknowledged until packet protection has been successfully
removed and all frames contained in the packet have been processed. For STREAM
frames, this means the data has been queued (but not necessarily delivered to
the application). This also means that any stream state transitions triggered
by STREAM or RST_STREAM frames have occurred. 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.Strategies and implications of the frequency of generating acknowledgments are
discussed in more detail in .The QUIC packet size includes the QUIC header and integrity check, but not the
UDP or IP header.Clients MUST pad any Initial packet it sends to have a QUIC packet size of at
least 1200 octets. Sending an Initial packet of this size ensures that the
network path supports a reasonably sized packet, and helps reduce the amplitude
of amplification attacks caused by server responses toward an unverified client
address.An Initial packet MAY exceed 1200 octets if the client knows 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 an Initial packet 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.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 (). Endpoints
MAY use PMTU Discovery (, ) 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.Traditional ICMP-based path MTU discovery in IPv4 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.The PADDING frame provides a useful option for PMTU probe packets that does not
exist in other transports. PADDING frames generate acknowledgements, but their
content need not be delivered reliably. PADDING frames may delay the delivery of
application data, as they consume the congestion window. However, by definition
their likely loss in a probe packet does not require delay-inducing
retransmission of application data.When implementing the algorithm in Section 7.2 of , 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 discusses tradeoffs 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 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 ).Either type of stream can be created by either endpoint, can concurrently send
data interleaved with other streams, and can be cancelled.Data that is received on a stream is delivered in order within that stream, but
there is no particular delivery order across streams. Transmit ordering among
streams is left to the implementation.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
, which may be a more appealing description
for some applications.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
.Low BitsStream Type0x0Client-Initiated, Bidirectional0x1Server-Initiated, Bidirectional0x2Client-Initiated, Unidirectional0x3Server-Initiated, UnidirectionalStream ID 0 (0x0) is a client-initiated, bidirectional stream that is used for
the cryptographic handshake. Stream 0 MUST NOT be used for application data.A QUIC endpoint MUST NOT reuse a Stream ID. Open streams can be used in any
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 ).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 (); another
for streams from which an endpoint receives data ().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 causes the stream to open
in both directions.Opening a stream causes all lower-numbered streams of the same type to
implicitly open. This includes both send and receive streams if the stream is
bidirectional. For bidirectional streams, an endpoint can send data on an
implicitly opened stream. On both unidirectional and bidirectional streams, an
endpoint MAY send MAX_STREAM_DATA or STOP_SENDING on implicitly opened streams.
An endpoint SHOULD NOT implicitly open streams that it initiates, instead
opening streams in order.
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. shows the states for the part of a stream that sends
data to a peer.|
+-------+ |
| |
| Send STREAM + FIN |
v v
+-------+ +-------+
| Data | Send RST_STREAM | Reset |
| Sent +------------------>| Sent |
+-------+ +-------+
| |
| Recv All ACKs | Recv ACK
v v
+-------+ +-------+
| Data | | Reset |
| Recvd | | Recvd |
+-------+ +-------+
]]>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 “Open” 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 “Open” 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 “Open”, “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. 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 “Open” state;
instead, receive streams track the delivery of data to the application or
application protocol some of which cannot be observed by the sender.|
+-------+ |
| |
| 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 |
+-------+ +-------+
]]>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 “Open” 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.In the “Recv” state, the endpoint receives STREAM and STREAM_BLOCKED frames.
Incoming data is buffered and 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
) 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.The sender of a stream sends just three frame types that affect the state of a
stream at either sender or receiver: STREAM (), STREAM_BLOCKED
(), and 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 () and
STOP_SENDING frames ().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.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. shows a more complex mapping of bidirectional stream
states that loosely correspond to the stream states in HTTP/2
. 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 StreamReceive StreamComposite StateNo Stream/OpenNo Stream/Recv *1idleOpen/Send/Data SentRecv/Size KnownopenOpen/Send/Data SentData Recvd/Data Readhalf-closed (remote)Open/Send/Data SentReset Recvd/Reset Readhalf-closed (remote)Data RecvdRecv/Size Knownhalf-closed (local)Reset Sent/Reset RecvdRecv/Size Knownhalf-closed (local)Data RecvdRecv/Size Knownhalf-closed (local)Reset Sent/Reset RecvdData Recvd/Data ReadclosedReset Sent/Reset RecvdReset Recvd/Reset ReadclosedData RecvdData Recvd/Data ReadclosedData RecvdReset Recvd/Reset Readclosed
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.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.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 ) and is subsequently
increased by MAX_STREAM_ID frames (see ).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 (), unless this
is a result of a change in the initial offsets (see ).A receiver MUST NOT renege on an advertisement; that is, once a receiver
advertises a stream ID via a MAX_STREAM_ID frame, it MUST NOT subsequently
advertise a smaller maximum ID. A sender may receive MAX_STREAM_ID frames out
of order; a sender MUST therefore ignore any MAX_STREAM_ID that does not
increase the maximum.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 byte of data that is sent on a stream has the stream offset 0.
The largest offset delivered on a stream MUST be less than 2^62. A receiver
MUST ensure that received stream data is delivered to the application as an
ordered byte-stream. Data received out of order MUST be buffered for later
delivery, as long as it is not in violation of the receiver’s flow control
limits.An endpoint MUST NOT send data on any stream without ensuring that it is within
the data limits set by its peer. The cryptographic handshake stream, Stream 0,
is exempt from the connection-level data limits established by MAX_DATA. Data on
stream 0 other than the initial cryptographic handshake message is still subject
to stream-level data limits and MAX_STREAM_DATA. This message is exempt
from flow control because it needs to be sent in a single packet regardless of
the server’s flow control state. This rule applies even for 0-RTT handshakes
where the remembered value of MAX_STREAM_DATA would not permit sending a full
initial cryptographic handshake message.Flow control is described in detail in , and congestion control
is described in the companion document .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 , 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.Stream 0 MUST 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 frames that are 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.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
. 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 the
maximum absolute byte offset of a stream, while MAX_DATA sends the
maximum sum of the absolute byte offsets of all streams other than
stream 0.A receiver MAY advertise a larger offset at any point by sending MAX_DATA or
MAX_STREAM_DATA frames. A receiver MUST NOT renege on an advertisement; that
is, once a receiver advertises an offset, it MUST NOT subsequently advertise a
smaller offset. A sender could receive MAX_DATA or MAX_STREAM_DATA frames out
of order; a sender MUST therefore ignore any flow control offset that does not
move the window forward.A receiver MUST close the connection with a FLOW_CONTROL_ERROR error
() 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 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.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.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.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
tradeoff 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 roundtrip time estimate and the rate at
which the receiving application consumes data, similar to common TCP
implementations.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.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 possibiity of loss, a receiver should
send a MAX_DATA or MAX_STREAM_DATA frame at least two roundtrips 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.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 ). 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.An endpoint that detects an error SHOULD signal the existence of that error to
its peer. Errors can affect an entire connection (see ),
or a single stream (see ).The most appropriate error code () 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 () 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.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
(, ). 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
().An endpoint that receives an invalid CONNECTION_CLOSE or APPLICATION_CLOSE frame
MUST NOT signal the existence of the error to its peer.If the error affects a single stream, but otherwise leaves the connection in a
recoverable state, the endpoint can send a RST_STREAM frame
() with an appropriate error code to terminate just the
affected stream.Stream 0 is critical to the functioning of the entire connection. If stream 0
is closed with either a RST_STREAM or STREAM frame bearing the FIN flag, an
endpoint MUST generate a connection error of type PROTOCOL_VIOLATION.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.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.
An endpoint uses this with CONNECTION_CLOSE to signal that the connection is
being closed abruptly in the absence of any error.
The endpoint encountered an internal error and cannot continue with the
connection.
An endpoint received more data than it permitted in its advertised data limits
(see ).
An endpoint received a frame for a stream identifier that exceeded its
advertised maximum stream ID.
An endpoint received a frame for a stream that was not in a state that
permitted that frame (see ).
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.
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. This is a
generic error code; an endpoint SHOULD use the more specific frame format
error codes (0x1XX) if possible.
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.
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.
An endpoint detected an error with protocol compliance that was not covered by
more specific error codes.
An endpoint received a PONG frame that did not correspond to any PING frame
that it previously sent.
An endpoint detected an error in a specific frame type. The frame type is
included as the last octet of the error code. For example, an error in a
MAX_STREAM_ID frame would be indicated with the code (0x106).See for details of registering new 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
() and 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.An attacker might be able to receive an address validation token
() 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.The attacks commonly known as 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.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.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 .
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 . If chosen judisciously, 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.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 . Values with the first byte 0xff are
reserved for Private Use .Registrations MUST include the following fields:
The numeric value of the assignment (registrations will be between 0x0000 and
0xfeff).
A short mnemonic for the parameter.
A reference to a publicly available specification for the value.The nominated expert(s) verify that a specification exists and is readily
accessible. The 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 .ValueParameter NameSpecification0x0000initial_max_stream_data0x0001initial_max_data0x0002initial_max_stream_id_bidi0x0003idle_timeout0x0004omit_connection_id0x0005max_packet_size0x0006stateless_reset_token0x0007ack_delay_exponent0x0008initial_max_stream_id_uniIANA [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 . Values with the first byte 0xff are
reserved for Private Use .Registrations MUST include the following fields:
The numeric value of the assignment (registrations will be between 0x0000 and
0xfeff).
A short mnemonic for the parameter.
A brief description of the error code semantics, which MAY be a summary if a
specification reference is provided.
A reference to a publicly available specification for the value.The initial contents of this registry are shown in . Note
that FRAME_ERROR takes the range from 0x100 to 0x1FF and private use occupies
the range from 0xFE00 to 0xFFFF.ValueErrorDescriptionSpecification0x0NO_ERRORNo error0x1INTERNAL_ERRORImplementation error0x3FLOW_CONTROL_ERRORFlow control error0x4STREAM_ID_ERRORInvalid stream ID0x5STREAM_STATE_ERRORFrame received in invalid stream state0x6FINAL_OFFSET_ERRORChange to final stream offset0x7FRAME_FORMAT_ERRORGeneric frame format error0x8TRANSPORT_PARAMETER_ERRORError in transport parameters0x9VERSION_NEGOTIATION_ERRORVersion negotiation failure0xAPROTOCOL_VIOLATIONGeneric protocol violation0xBUNSOLICITED_PONGUnsolicited PONG frame0x100-0x1FFFRAME_ERRORSpecific frame format errorQUIC Loss Detection and Congestion ControlGoogleGoogleUsing Transport Layer Security (TLS) to Secure QUICMozillasn3rdKey words for use in RFCs to Indicate Requirement LevelsIn many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.Ambiguity of Uppercase vs Lowercase in RFC 2119 Key WordsRFC 2119 specifies common key words that may be used in protocol specifications. This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the defined special meanings.Randomness Requirements for SecuritySecurity systems are built on strong cryptographic algorithms that foil pattern analysis attempts. However, the security of these systems is dependent on generating secret quantities for passwords, cryptographic keys, and similar quantities. The use of pseudo-random processes to generate secret quantities can result in pseudo-security. A sophisticated attacker may find it easier to reproduce the environment that produced the secret quantities and to search the resulting small set of possibilities than to locate the quantities in the whole of the potential number space.Choosing random quantities to foil a resourceful and motivated adversary is surprisingly difficult. This document points out many pitfalls in using poor entropy sources or traditional pseudo-random number generation techniques for generating such quantities. It recommends the use of truly random hardware techniques and shows that the existing hardware on many systems can be used for this purpose. It provides suggestions to ameliorate the problem when a hardware solution is not available, and it gives examples of how large such quantities need to be for some applications. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.The Transport Layer Security (TLS) Protocol Version 1.3This document specifies version 1.3 of the Transport Layer Security (TLS) protocol. TLS allows client/server applications to communicate over the Internet in a way that is designed to prevent eavesdropping, tampering, and message forgery.UTF-8, a transformation format of ISO 10646ISO/IEC 10646-1 defines a large character set called the Universal Character Set (UCS) which encompasses most of the world's writing systems. The originally proposed encodings of the UCS, however, were not compatible with many current applications and protocols, and this has led to the development of UTF-8, the object of this memo. UTF-8 has the characteristic of preserving the full US-ASCII range, providing compatibility with file systems, parsers and other software that rely on US-ASCII values but are transparent to other values. This memo obsoletes and replaces RFC 2279.Packetization Layer Path MTU DiscoveryThis document describes a robust method for Path MTU Discovery (PMTUD) that relies on TCP or some other Packetization Layer to probe an Internet path with progressively larger packets. This method is described as an extension to RFC 1191 and RFC 1981, which specify ICMP-based Path MTU Discovery for IP versions 4 and 6, respectively. [STANDARDS-TRACK]Path MTU discoveryThis memo describes a technique for dynamically discovering the maximum transmission unit (MTU) of an arbitrary internet path. It specifies a small change to the way routers generate one type of ICMP message. For a path that passes through a router that has not been so changed, this technique might not discover the correct Path MTU, but it will always choose a Path MTU as accurate as, and in many cases more accurate than, the Path MTU that would be chosen by current practice. [STANDARDS-TRACK]Path MTU Discovery for IP version 6This document describes Path MTU Discovery (PMTUD) for IP version 6. It is largely derived from RFC 1191, which describes Path MTU Discovery for IP version 4. It obsoletes RFC 1981.Path MTU discoveryThis memo describes a technique for dynamically discovering the maximum transmission unit (MTU) of an arbitrary internet path. It specifies a small change to the way routers generate one type of ICMP message. For a path that passes through a router that has not been so changed, this technique might not discover the correct Path MTU, but it will always choose a Path MTU as accurate as, and in many cases more accurate than, the Path MTU that would be chosen by current practice. [STANDARDS-TRACK]Packetization Layer Path MTU DiscoveryThis document describes a robust method for Path MTU Discovery (PMTUD) that relies on TCP or some other Packetization Layer to probe an Internet path with progressively larger packets. This method is described as an extension to RFC 1191 and RFC 1981, which specify ICMP-based Path MTU Discovery for IP versions 4 and 6, respectively. [STANDARDS-TRACK]Guidelines for Writing an IANA Considerations Section in RFCsMany protocols make use of points of extensibility that use constants to identify various protocol parameters. To ensure that the values in these fields do not have conflicting uses and to promote interoperability, their allocations are often coordinated by a central record keeper. For IETF protocols, that role is filled by the Internet Assigned Numbers Authority (IANA).To make assignments in a given registry prudently, guidance describing the conditions under which new values should be assigned, as well as when and how modifications to existing values can be made, is needed. This document defines a framework for the documentation of these guidelines by specification authors, in order to assure that the provided guidance for the IANA Considerations is clear and addresses the various issues that are likely in the operation of a registry.This is the third edition of this document; it obsoletes RFC 5226.QUIC: Multiplexed Transport Over UDPWelcome to Slowloris...Guide for Internet Standards WritersThis document is a guide for Internet standard writers. It defines those characteristics that make standards coherent, unambiguous, and easy to interpret. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.TCP Extensions for Multipath Operation with Multiple AddressesTCP/IP communication is currently restricted to a single path per connection, yet multiple paths often exist between peers. The simultaneous use of these multiple paths for a TCP/IP session would improve resource usage within the network and, thus, improve user experience through higher throughput and improved resilience to network failure.Multipath TCP provides the ability to simultaneously use multiple paths between peers. This document presents a set of extensions to traditional TCP to support multipath operation. The protocol offers the same type of service to applications as TCP (i.e., reliable bytestream), and it provides the components necessary to establish and use multiple TCP flows across potentially disjoint paths. This document defines an Experimental Protocol for the Internet community.Transport Layer Security (TLS) Application-Layer Protocol Negotiation ExtensionThis document describes a Transport Layer Security (TLS) extension for application-layer protocol negotiation within the TLS handshake. For instances in which multiple application protocols are supported on the same TCP or UDP port, this extension allows the application layer to negotiate which protocol will be used within the TLS connection.HMAC: Keyed-Hashing for Message AuthenticationThis document describes HMAC, a mechanism for message authentication using cryptographic hash functions. HMAC can be used with any iterative cryptographic hash function, e.g., MD5, SHA-1, in combination with a secret shared key. The cryptographic strength of HMAC depends on the properties of the underlying hash function. This memo provides information for the Internet community. This memo does not specify an Internet standard of any kindHMAC-based Extract-and-Expand Key Derivation Function (HKDF)This document specifies a simple Hashed Message Authentication Code (HMAC)-based key derivation function (HKDF), which can be used as a building block in various protocols and applications. The key derivation function (KDF) is intended to support a wide range of applications and requirements, and is conservative in its use of cryptographic hash functions. This document is not an Internet Standards Track specification; it is published for informational purposes.Network Address Translation (NAT) Behavioral Requirements for Unicast UDPThis document defines basic terminology for describing different types of Network Address Translation (NAT) behavior when handling Unicast UDP and also defines a set of requirements that would allow many applications, such as multimedia communications or online gaming, to work consistently. Developing NATs that meet this set of requirements will greatly increase the likelihood that these applications will function properly. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.Structured streamsHypertext Transfer Protocol Version 2 (HTTP/2)This specification describes an optimized expression of the semantics of the Hypertext Transfer Protocol (HTTP), referred to as HTTP version 2 (HTTP/2). HTTP/2 enables a more efficient use of network resources and a reduced perception of latency by introducing header field compression and allowing multiple concurrent exchanges on the same connection. It also introduces unsolicited push of representations from servers to clients.This specification is an alternative to, but does not obsolete, the HTTP/1.1 message syntax. HTTP's existing semantics remain unchanged.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 . 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.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.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.Clarified requirements for BLOCKED usage (#65, #924)BLOCKED frame now includes reason for blocking (#452, #924, #927, #928)Cleartext integrity as version independent (#568)GAP limitation in ACK Frame (#613)Improved PMTUD description (#614, #1036)Clarified stream state machine (#634, #662, #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)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-directinal 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)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)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_BLOCKEDClarify Keep-alive requirements (#729)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)Change STREAM and RST_STREAM layoutAdd MAX_STREAM_ID settingsThe 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)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)Replaced DIVERSIFICATION_NONCE flag with KEY_PHASE flagDefined versioningReworked description of packet and frame layoutError code space is divided into regions for each componentUse big endian for all numeric valuesAdopted as base for draft-ietf-quic-tlsUpdated authors/editors listAdded IANA Considerations sectionMoved Contributors and Acknowledgments to appendices