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 words “MUST”, “MUST NOT”, “SHOULD”, and “MAY” are used in this document.
It’s not shouting; when they are capitalized, they have the special meaning
defined in .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. QUIC packet
size in this document refers to the UDP payload size.Packet and frame diagrams use the format described in Section 3.1,
with the following additional conventions:
Indicates that x is optional
Indicates that x is encrypted
Indicates that x is A bits long
Indicates that x is one of A, B, or C bits long
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 up
to 256 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
lient, 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 an invalid version. 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 can be 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 SHOULD switch to sending short-form headers. While inefficient,
long headers MAY be used for packets encrypted with 1-RTT keys. 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 packet number. describes the
use of packet numbers.
Octets 13 to 16 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 from 17 onwards (the rest of QUIC packet) are the payload of the
packet.The following packet types are defined:TypeNameSection0x01Version Negotiation0x02Client Initial0x03Server Stateless Retry0x04Server Cleartext0x05Client Cleartext0x060-RTT Protected0x071-RTT Protected (key phase 0)0x081-RTT Protected (key phase 1)The 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.(TODO: Should the list of packet types be version-independent?)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
present. If set to 1, then the Connection ID field is present; if set to 0,
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 Connection ID Flag is 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 Size0x011 octet0x022 octets0x034 octetsThe header form, 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 has long headers with a type value of 0x01 and is
sent only by servers. The Version Negotiation packet is a response to a client
packet that contains a version that is not supported by the server.The packet number, connection ID and version fields echo corresponding values
from the triggering client packet. This allows clients some assurance that the
server received the packet and that the Version Negotiation packet was not
carried in a packet with a spoofed source address.A Version Negotiation packet is never explicitly acknowledged in an ACK frame by
a client. Receiving another Client Initial packet implicitly acknowledges a
Version Negotiation packet.The payload of the Version Negotiation packet is a list of 32-bit versions which
the server supports, as shown below.See for a description of the version negotiation
process.Cleartext packets are sent during the handshake prior to key negotiation.All cleartext packets contain the current QUIC version in the version field.The payload of cleartext packets also includes an integrity check, which is
described in .The Client Initial packet uses long headers with a type value of 0x02.
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 packet number used for Client Initial packets is initialized with a random
value each time the new contents are created for the packet. Retransmissions of
the packet contents increment the packet number by one, see
().The payload of a Client 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 cyptographic handshake message MUST fit in a single packet (see
).The client uses the Client 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 Server Stateless Retry packet
().A Server Stateless Retry packet uses long headers with a type value of 0x03.
It carries cryptographic handshake messages and acknowledgments. It is used
by a server that wishes to perform a stateless retry (see
).The packet number and connection ID fields echo the corresponding fields from
the triggering client packet. This allows a client to verify that the server
received its packet.A Server Stateless Retry packet is never explicitly acknowledged in an ACK frame
by a client. Receiving another Client Initial packet implicitly acknowledges a
Server Stateless Retry packet.After receiving a Server Stateless Retry packet, the client uses a new Client
Initial packet containing the next cryptographic handshake message. The client
retains the state of its cryptographic handshake, but discards all transport
state. The new Client Initial packet includes a newly randomized packet number,
STREAM frames on stream 0 that start again at an offset of 0, and the original
connection ID.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 Server Stateless Retry packet contains STREAM frames and
could contain PADDING and ACK frames. A server can only send a single Server
Stateless Retry packet in response to each Client Initial packet that is
receives.A Server Cleartext packet uses long headers with a type value of 0x04. It is
used to carry acknowledgments and cryptographic handshake messages from the
server.The connection ID field in a Server Cleartext packet contains a connection ID
that is chosen by the server (see ).The first Server Cleartext packet contains a randomized packet number. This
value is increased for each subsequent packet sent by the server as described in
.The payload of this packet contains STREAM frames and could contain PADDING and
ACK frames.A Client Cleartext packet uses long headers with a type value of 0x05, and is
sent when the client has received a Server Cleartext packet from the server.The connection ID field in a Client Cleartext packet contains a server-selected
connection ID, see .The Client Cleartext packet includes a packet number that is one higher than the
last Client Initial, 0-RTT Protected or Client Cleartext packet that was sent.
The packet number is incremented for each subsequent packet, see
.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. Packets
that are protected with 1-RTT keys MAY be sent with long 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 0x06. The connection ID
field for a 0-RTT packet is selected by the client.The client can send 0-RTT packets after having received a packet from the server
if that packet does not complete the handshake. Even if the client receives a
different connection ID from the server, it MUST NOT update the connection ID it
uses for 0-RTT packets. This enables consistent routing for all 0-RTT packets.Packets protected with 1-RTT keys that use long headers use a type value of 0x07
for key phase 0 and 0x08 for key phase 1; see for more details on
the use of key phases. The connection ID field for these packet types MUST
contain the value selected by the server, 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 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 Client Initial
packets () and 0-RTT packets ().
If the client has received any packet from the server, it uses the connection ID
it received from the server for all packets other than 0-RTT packets.When the server receives a Client Initial packet and decides to proceed with the
handshake, it chooses a new value for the connection ID and sends that in a
Server Cleartext 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 uses
this for all subsequent packets that it sends, except for any 0-RTT packets,
which all have the same connection ID.The packet number is a 64-bit unsigned number and is used as part of a
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 2^64 - 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.To reduce the number of bits required to represent the packet number over the
wire, only the least significant bits of the packet number are transmitted. 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 Server Stateless Retry
() packets have special rules for populating the
packet number field.The initial value for packet number MUST be selected from an uniform random
distribution between 0 and 2^31-1. That is, the lower 31 bits of the packet
number are randomized. provides guidance on the generation of
random values.The first set of packets sent by an endpoint MUST include the low 32-bits of the
packet number. Once any packet has been acknowledged, subsequent packets can
use a shorter packet number encoding.A client that receives a Version Negotiation () or Server
Stateless Retry packet () MUST generate a new initial
packet number. This ensures that the first transmission attempt for a Client
Initial packet () always contains a randomized packet
number, but packets that contain retransmissions increment the packet number.A client MUST NOT generate a new initial packet number if it discards the server
packet. This might happen if the information the client retransmits its Client
Initial packet.Between different versions the following things are guaranteed to remain
constant:the location of the header form flag,the location of the 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, andthe location and size of the Packet Number field in long headers.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 cleartext packets and the plaintext after decryption of protected
payloads consists of a sequence of frames, as shown in .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_CLOSE0x04MAX_DATA0x05MAX_STREAM_DATA0x06MAX_STREAM_ID0x07PING0x08BLOCKED0x09STREAM_BLOCKED0x0aSTREAM_ID_BLOCKED0x0bNEW_CONNECTION_ID0x0cSTOP_SENDING0xa0 - 0xbfACK0xc0 - 0xffSTREAMA 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 .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 a handshake packet. The details
of the handshake mechanisms are described in , but all of the
initial packets sent from the client to the server MUST use the long header
format and MUST specify the version of the protocol being used.When the server receives a packet from a client with the long header format, it
compares the client’s version to the versions it supports.If the version selected by the client is not acceptable to the server, the
server discards the incoming packet and responds with a Version Negotiation
packet (). This includes a list of versions that the server
will accept.To avoid packet amplification attacks a server MUST NOT send a Version
Negotiation packet that is larger than the packet it responds to. It is
anticipated that this is ample space for all QUIC versions that a single server
might need to advertise.A server sends a Version Negotiation packet for every packet that it receives
with an unacceptable version. This allows a server to process packets with
unsupported versions without retaining state. Though either the initial client
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.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.When the client receives a Version Negotiation packet from the server, it should
select an acceptable protocol version. If the server lists an acceptable
version, the client selects that version and reattempts to create a connection
using that version. Though the contents of a packet 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 ignore 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 uses unprotected data. 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. However, when the server generates
a Version Negotiation packet, it cannot randomly generate a reserved version
number. This is because the server is required to include the same value in its
transport parameters (see ). To avoid the selected
version number changing during connection establishment, the reserved version
SHOULD be generated as a function of values that will be available to the server
when later generating its handshake packets.A pseudorandom function that takes client address information (IP and port) and
the client selected version as input would ensure that there is sufficient
variability in the values that a server uses.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 negotiated_version;
QuicVersion initial_version;
case encrypted_extensions:
QuicVersion supported_versions<4..2^8-4>;
case new_session_ticket:
struct {};
};
TransportParameter parameters<30..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
.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 1024 octets. That is, the value
here is multiplied by 1024 to determine the actual maximum value. This is
equivalent to sending a MAX_DATA () for the connection
immediately after completing the handshake.
The initial maximum stream ID parameter contains the initial maximum stream
number the peer may initiate, encoded as an unsigned 32-bit integer. This is
equivalent to sending a MAX_STREAM_ID () 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 omit connection identifier parameter indicates that packets sent to the
endpoint that advertises this parameter can omit the connection ID. 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 this parameter indicates that the endpoint relies on the
connection ID being present in every packet.
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
().Transport parameters from the server MUST be remembered by the client for use
with 0-RTT data. If the TLS NewSessionTicket message includes the
quic_transport_parameters extension, then those values are used for the server
values when establishing a new connection using that ticket. Otherwise, the
transport parameters that the server advertises during connection establishment
are used.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.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
.The transport parameters include three fields that encode version information.
These retroactively authenticate the version negotiation (see
) that is performed prior to the cryptographic handshake.The cryptographic handshake provides integrity protection for the negotiated
version as part of the transport parameters (see ). As
a result, modification of version negotiation packets by an attacker can be
detected.The client includes two fields in the transport parameters:The negotiated_version is the version that was finally selected for use. This
MUST be identical to the value that is on the packet that carries the
ClientHello. A server 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 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.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 and negotiated
versions are the same, a stateless server can accept the value.If the initial version is different from the negotiated_version, 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 value of negotiated_version, the server MUST terminate the connection with a
VERSION_NEGOTIATION_ERROR error.The server includes a list of versions that it would send in any version
negotiation packet () in supported_versions. The server
populates this field even if it did not send a version negotiation packet. This
field is absent if the parameters are included in a NewSessionTicket message.The client can validate 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
negotiated_version value is not included 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 Server Stateless 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 Server Stateless 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 Server Stateless 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 1280 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.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.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.TODO: see issue #161Connections 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 four ways:application close ()idle timeout ()immediate close ()stateless reset ()After a connection is closed for any reason, an endpoint might receive packets
from its peer. These packets might have been sent prior to receiving any close
signal, or they might be retransmissions of packets for which acknowledgments
were lost.The draining period persists for three times the current Retransmission Timeout
(RTO) interval as defined in . During this period, new packets
can be acknowledged, but no new application data can be sent on the connection.Different treatment is given to packets that are received while a connection is
in the draining period depending on how the connection was closed. In all
cases, it is possible to acknowledge packets that are received as normal, but
other reactions might be preferable depending on how the connection was closed.
An endpoint that is in a draining period MUST NOT send packets containing frames
other than ACK, PADDING, or CONNECTION_CLOSE.Once the draining period has ended, an endpoint SHOULD discard per-connection
state. This results in new packets on the connection being discarded. An
endpoint MAY send a stateless reset in response to any further incoming packets.The draining period does not apply when a stateless reset ()
is used.An application protocol can arrange to close a connection. This might be after
negotiating 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.
A negotiated shutdown might not result in exchanging messages that are visible
to the transport.In the draining period, an endpoint that has been closed by an application
SHOULD generate and send ACK frames as normal. This allows the peer to receive
acknowledgements where previous acknowledgements were lost.A connection that remains idle for longer than the idle timeout (see
) becomes closed. Either peer removes
connection state if they have neither sent nor received a packet for this time.The time at which an idle timeout takes effect won’t be perfectly synchronized
on peers. A connection enters the draining period when the idle timeout
expires. During this time, an endpoint that receives new packets MAY choose to
restore the connection. Alternatively, an endpoint that receives packets MAY
signal the timeout using an immediate close.An endpoint sends a CONNECTION_CLOSE frame to terminate the connection
immediately. A CONNECTION_CLOSE causes all open streams to immediately become
closed; open streams can be assumed to be implicitly reset. After sending or
receiving a CONNECTION_CLOSE frame, endpoints immediately enter a draining
period.During the draining period, an endpoint that sends a CONNECTION_CLOSE frame
SHOULD respond to any subsequent packet that it receives with another packet
containing a CONNECTION_CLOSE frame. To reduce the state that an endpoint
maintains in this case, it MAY send the exact same packet. However, endpoints
SHOULD limit the number of CONNECTION_CLOSE messages they generate. For
instance, an endpoint could progressively increase the number of packets that it
receives before sending additional CONNECTION_CLOSE frames.
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.An endpoint can cease sending CONNECTION_CLOSE frames if it receives either a
CONNECTION_CLOSE or an acknowledgement for a packet that contained a
CONNECTION_CLOSE.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 CONNECTION_CLOSE 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:This packet SHOULD use the short header form with the shortest possible packet
number encoding. This minimizes the perceived gap between the last packet that
the server sent and this packet. The leading octet of the Stateless Reset Token
will be interpreted as a packet number. A server MAY use a different short
header type, indicating a different packet number length, but this allows for
the message to be identified as a stateless reset more easily using heuristics.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.After the first short header octet and optional connection ID, the server
includes the value of the Stateless Reset Token that it included in its
transport parameters.After the Stateless Reset Token, the server pads the message with an arbitrary
number of octets containing random values.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 frame if it has sufficient state to do so.A client detects a potential stateless reset when a packet with a short header
cannot be decrypted. The client then performs a constant-time comparison of the
16 octets that follow the Connection ID with the Stateless Reset Token provided
by the server in its transport parameters. If this comparison is successful,
the connection MUST be terminated immediately. Otherwise, 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.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.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:
The 32-bit Stream ID of the stream being terminated.
A 32-bit error code which indicates why the stream is being closed.
A 64-bit unsigned 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. If there are open streams that haven’t been
explicitly closed, they are implicitly closed when the connection is closed.
The frame is as follows:The fields of a CONNECTION_CLOSE frame are as follows:
A 32-bit error code which indicates the reason for closing this connection.
A 16-bit unsigned number 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 .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 64-bit unsigned integer indicating the maximum amount of data that can be
sent on the entire connection, in units of 1024 octets. That is, the updated
connection-level data limit is determined by multiplying the encoded value by
1024.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 closed streams, 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.
A 64-bit unsigned 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 peer-initiated stream ID for the connection.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 no
additional fields. The receiver of a PING frame simply needs to acknowledge the
packet containing this frame.A PING frame has no additional fields.The PING frame can be used to keep a connection alive when an application or
application protocol wishes to prevent the connection from timing out. An
application protocol SHOULD provide guidance about the conditions under which
generating a PING is recommended. This guidance SHOULD indicate whether it is
the client or the server that is expected to send the PING. Having both
endpoints send PING frames without coordination can produce an excessive number
of packets and poor performance.A connection will time out if no packets are sent or received for a period
longer than the time specified in the idle_timeout transport parameter (see
). 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 sends 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 does not contain a payload.A sender sends 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 a single field:
A 32-bit unsigned number indicating the stream which is flow control blocked.A sender MAY send a STREAM_ID_BLOCKED frame (type=0x0a) when it wishes to open a
stream, but is unable to due to the maximum stream ID limit set by its peer (see
). 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 does not contain a payload.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 16-bit sequence number. This value starts at 0 and increases by 1 for each
connection ID that is provided by the server. The sequence value can wrap;
the value 65535 is followed by 0. When wrapping the sequence field, the
server MUST ensure that a value with the same sequence has been received and
acknowledged by the client. The connection ID that is assigned during the
handshake is assumed to have a sequence of 65535.
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 frame is as
follows:The fields are:
The 32-bit Stream ID of the stream being ignored.
The application-specified reason the sender is ignoring the stream.Receivers send ACK frames to inform senders which packets they have received and
processed, as well as which packets are considered missing. The ACK frame
contains between 1 and 256 ACK blocks. ACK blocks are ranges of acknowledged
packets. Implementations SHOULD NOT generate ACK packets in response to packets
which only contain ACKs. However, they SHOULD ACK those packets when sending
ACKs for other packets.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. When this is necessary, the receiver
SHOULD acknowledge newly received packets and stop acknowledging packets
received in the past.Unlike TCP SACKs, QUIC ACK blocks 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 Server Stateless Retry
packets. These packet types contain packet numbers selected by the client, not
the server.QUIC ACK frames contain a timestamp section with up to 255 timestamps.
Timestamps enable better congestion control, but are not required for correct
loss recovery, and old timestamps are less valuable, so it is not guaranteed
every timestamp will be received by the sender. A receiver SHOULD send a
timestamp exactly once for each received packet containing retransmittable
frames. A receiver MAY send timestamps for non-retransmittable packets.
A receiver MUST not send timestamps in unprotected packets.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 blocks of missing packets; skipping
packet numbers between 1 and 255 effectively provides up to 8 bits of efficient
entropy on demand, which should be adequate protection against most
opportunistic acknowledgement attacks.The type byte for a ACK frame contains embedded flags, and is formatted as
101NLLMM. These bits are parsed as follows:The first three bits must be set to 101 indicating that this is an ACK frame.The N bit indicates whether the frame contains a Num Blocks field.The two LL bits encode the length of the Largest Acknowledged field.
The values 00, 01, 02, and 03 indicate lengths of 8, 16, 32, and 64
bits respectively.The two MM bits encode the length of the ACK Block Length fields.
The values 00, 01, 02, and 03 indicate lengths of 8, 16, 32, and 64
bits respectively.An ACK frame is shown below.The fields in the ACK frame are as follows:
An optional 8-bit unsigned value specifying the number of additional ACK
blocks (besides the required First ACK Block) in this ACK frame. Only present
if the ‘N’ flag bit is 1.
An unsigned 8-bit number specifying the total number of <packet number,
timestamp> pairs in the Timestamp Section.
A variable-sized unsigned value representing the largest packet number the
peer is acknowledging in this packet (typically the largest that the peer has
seen thus far.)
The time from when the largest acknowledged packet, as indicated in the
Largest Acknowledged field, was received by this peer to when this ACK was
sent.
Contains one or more blocks of packet numbers which have been successfully
received, see .
Contains zero or more timestamps reporting transit delay of received packets.
See .The ACK Block Section contains between one and 256 blocks of packet numbers
which have been successfully received. If the Num Blocks field is absent, only
the First ACK Block length is present in this section. Otherwise, the Num Blocks
field indicates how many additional blocks follow the First ACK Block Length
field.The fields in the ACK Block Section are:
An unsigned packet number delta that indicates the number of contiguous
additional packets being acknowledged starting at the Largest Acknowledged.
An unsigned number specifying the number of contiguous missing packets from
the end of the previous ACK block to the start of the next. Repeated “Num
Blocks” times.
An unsigned packet number delta that indicates the number of contiguous
packets being acknowledged starting after the end of the previous gap.
Repeated “Num Blocks” times.The Timestamp Section contains between zero and 255 measurements of packet
receive times relative to the beginning of the connection.The fields in the Timestamp Section are:
An optional 8-bit unsigned packet number delta specifying the delta between
the largest acknowledged and the first packet whose timestamp is being
reported. In other words, this first packet number may be computed as
(Largest Acknowledged - Delta Largest Acknowledged.)
An optional 32-bit unsigned value specifying the time delta in microseconds,
from the beginning of the connection to the arrival of the packet indicated by
Delta Largest Acknowledged.
This field has the same semantics and format as “Delta Largest Acknowledged”.
Repeated “Num Timestamps - 1” times.
An optional 16-bit unsigned value specifying time delta from the previous
reported timestamp. It is encoded in the same format as the ACK Delay.
Repeated “Num Timestamps - 1” times.The timestamp section lists packet receipt timestamps ordered by timestamp.DISCUSS_AND_REPLACE: Perhaps make this format simpler.The time format used in the ACK frame above is a 16-bit unsigned float with 11
explicit bits of mantissa and 5 bits of explicit exponent, specifying time in
microseconds. The bit format is loosely modeled after IEEE 754. For example, 1
microsecond is represented as 0x1, which has an exponent of zero, presented in
the 5 high order bits, and mantissa of 1, presented in the 11 low order bits.
When the explicit exponent is greater than zero, an implicit high-order 12th bit
of 1 is assumed in the mantissa. For example, a floating value of 0x800 has an
explicit exponent of 1, as well as an explicit mantissa of 0, but then has an
effective mantissa of 4096 (12th bit is assumed to be 1). Additionally, the
actual exponent is one-less than the explicit exponent, and the value represents
4096 microseconds. Any values larger than the representable range are clamped
to 0xFFFF.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 type byte
for a STREAM frame contains embedded flags, and is formatted as 11FSSOOD.
These bits are parsed as follows:The first two bits must be set to 11, indicating that this is a STREAM frame.F is the FIN bit, which is used for stream termination.The SS bits encode the length of the Stream ID header field.
The values 00, 01, 02, and 03 indicate lengths of 8, 16, 24, and 32 bits
long respectively.The OO bits encode the length of the Offset header field.
The values 00, 01, 02, and 03 indicate lengths of 0, 16, 32, and
64 bits long respectively.The D bit indicates whether a Data Length field is present in the STREAM
header. When set to 0, this field indicates that the Stream Data field
extends to the end of the packet. When set to 1, this field indicates that
Data Length field contains the length (in bytes) of the Stream Data field.
The option to omit the length should only be used when the packet is a
“full-sized” packet, to avoid the risk of corruption via padding.A STREAM frame is shown below.The STREAM frame contains the following fields:
The stream ID of the stream (see ).
A variable-sized unsigned number specifying the byte offset in the stream for
the data in this STREAM frame. When the offset length is 0, the offset is 0.
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 2^64.
An optional 16-bit unsigned number specifying the length of the Stream Data
field in this STREAM frame. This field is present when the D bit is set to
1.
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.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.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 () and 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.QUIC endpoints that implement any kind of PMTU discovery SHOULD maintain an
estimate for each combination of local and remote IP addresses (as each pairing
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 and therefore also supported by most modern IPv4 networks.
An endpoint MUST NOT reduce their MTU below this number, even if it receives
signals that indicate a smaller limit might exist.Clients MUST ensure that the first packet in a connection, and any
retransmissions of those octets, has a QUIC packet size of least 1200 octets.
The packet size for a QUIC packet includes the QUIC header and integrity check,
but not the UDP or IP header.The initial client packet SHOULD be padded to exactly 1200 octets unless the
client has a reasonable assurance that the PMTU is larger. Sending a packet of
this size ensures that the network path supports an MTU of this size and helps
reduce the amplitude of amplification attacks caused by server responses toward
an unverified client address.Servers MUST ignore an initial plaintext packet from a client if its total size
is less than 1200 octets.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.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, unless the stream has become closed
in the appropriate direction. See .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 .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.Streams in QUIC provide a lightweight, ordered, and bidirectional byte-stream
abstraction modeled closely on HTTP/2 streams .Streams can be created either by the client or the server, 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 32-bit integer, referred to as the Stream
ID. To avoid Stream ID collision, clients initiate streams using odd-numbered
Stream IDs; streams initiated by the server use even-numbered Stream IDs.Stream ID 0 (0x0) is reserved for the cryptographic handshake. Stream 0 MUST
NOT be used for application data, and is the first client-initiated stream.A QUIC endpoint cannot reuse a Stream ID. Streams MUST be created in sequential
order. Open streams can be used in any order. Streams that are used out of
order result in lower-numbered streams in the same direction being counted as
open.Stream IDs are usually encoded as a 32-bit integer, though the STREAM frame
() permits a shorter encoding when the leading bits of the
stream ID are zero.The semantics of QUIC streams is based on HTTP/2 streams, and the lifecycle of a
QUIC stream therefore closely follows that of an HTTP/2 stream ,
with some differences to accommodate the possibility of out-of-order delivery
due to the use of multiple streams in QUIC. The lifecycle of a QUIC stream is
shown in the following figure and described below.| closed |<-------------'
| |
+--------+
send: endpoint sends this frame
recv: endpoint receives this frame
STREAM: a STREAM frame
FIN: FIN flag in a STREAM frame
RST: RST_STREAM frame
MSD: MAX_STREAM_DATA frame
SB: STREAM_BLOCKED frame
]]>Note that this diagram shows stream state transitions and the frames and flags
that affect those transitions only. It is possible for a single frame to cause
two transitions: receiving a RST_STREAM frame, or a STREAM frame with the FIN
flag cause the stream state to move from “idle” to “open” and then immediately
to one of the “half-closed” states.The recipient of a frame that changes stream state will have a delayed view of
the state of a stream while the frame is in transit. Endpoints do not
coordinate the creation of streams; they are created unilaterally by either
endpoint. Endpoints can use acknowledgments to understand the peer’s subjective
view of stream state at any given time.In the absence of more specific guidance elsewhere in this document,
implementations SHOULD treat the receipt of a frame that is not expressly
permitted in the description of a state as a connection error (see
).All streams start in the “idle” state.The following transitions are valid from this state:Sending or receiving a STREAM or RST_STREAM frame causes the identified stream
to become “open”. The stream identifier for a new stream is selected as
described in . A RST_STREAM frame, or a STREAM frame with the FIN
flag set also causes a stream to become “half-closed”.An endpoint might receive MAX_STREAM_DATA or STREAM_BLOCKED frames on
peer-initiated streams that are “idle” if there is loss or reordering of
packets. Receiving these frames also causes the stream to become “open”.An endpoint MUST NOT send a STREAM or RST_STREAM frame for a stream ID that is
higher than the peers advertised maximum stream ID (see
).A stream in the “open” state may be used by both peers to send frames of any
type. In this state, endpoints can send MAX_STREAM_DATA and MUST observe the
value advertised by its receiving peer (see ).Opening a stream causes all lower-numbered streams in the same direction to
become open. Thus, opening an odd-numbered stream causes all “idle”,
odd-numbered streams with a lower identifier to become open and the same applies
to even numbered streams. Endpoints open streams in increasing numeric order,
but loss or reordering can cause packets that open streams to arrive out of
order.From the “open” state, either endpoint can send a frame with the FIN flag set,
which causes the stream to transition into one of the “half-closed” states. This
flag can be set on the frame that opens the stream, which causes the stream to
immediately become “half-closed”. Once an endpoint has completed sending all
stream data and a STREAM frame with a FIN flag, the stream state becomes
“half-closed (local)”. When an endpoint receives all stream data and a FIN flag
the stream state becomes “half-closed (remote)”. An endpoint MUST NOT consider
the stream state to have changed until all data has been sent or received.A RST_STREAM frame on an “open” stream also causes the stream to become
“half-closed”. A stream that becomes “open” as a result of sending or receiving
RST_STREAM immediately becomes “half-closed”. Sending a RST_STREAM frame causes
the stream to become “half-closed (local)”; receiving RST_STREAM causes the
stream to become “half-closed (remote)”.Any frame type that mentions a stream ID can be sent in this state.A stream that is in the “half-closed (local)” state MUST NOT be used for sending
on new STREAM frames. Retransmission of data that has already been sent on
STREAM frames is permitted. An endpoint MAY also send MAX_STREAM_DATA and
STOP_SENDING in this state.An endpoint that closes a stream MUST NOT send data beyond the final offset that
it has chosen, see for details.A stream transitions from this state to “closed” when a STREAM frame that
contains a FIN flag is received and all prior data has arrived, or when a
RST_STREAM frame is received.An endpoint can receive any frame that mentions a stream ID in this state.
Providing flow-control credit using MAX_STREAM_DATA frames is necessary to
continue receiving flow-controlled frames. In this state, a receiver MAY ignore
MAX_STREAM_DATA frames for this stream, which might arrive for a short period
after a frame bearing the FIN flag is sent.A stream is “half-closed (remote)” when the stream is no longer being used by
the peer to send any data. An endpoint will have either received all data that
a peer has sent or will have received a RST_STREAM frame and discarded any
received data.Once all data has been either received or discarded, a sender is no longer
obligated to update the maximum received data for the connection.Due to reordering, an endpoint could continue receiving frames for the stream
even after the stream is closed for sending. Frames received after a peer
closes a stream SHOULD be discarded. An endpoint MAY choose to limit the period
over which it ignores frames and treat frames that arrive after this time as
being in error.An endpoint will know the final offset of the data it receives on a stream when
it reaches the “half-closed (remote)” state, see for details.A stream in this state can be used by the endpoint to send any frame that
mentions a stream ID. In this state, the endpoint MUST observe advertised
stream and connection data limits (see ).A stream transitions from this state to “closed” by completing transmission of
all data. This includes sending all data carried in STREAM frames including
the terminal STREAM frame that contains a FIN flag.A stream also becomes “closed” when the endpoint sends a RST_STREAM frame.The “closed” state is the terminal state for a stream.Once a stream reaches this state, no frames can be sent that mention the stream.
Reordering might cause frames to be received after closing, see
.If an endpoint is no longer interested in the data being received, it MAY send a
STOP_SENDING frame on a stream in the “open” or “half-closed (local)” state to
prompt closure of the stream in the opposite direction. This typically
indicates that the receiving application is no longer reading 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.Upon receipt of a STOP_SENDING frame on a stream in the “open” or “half-closed
(remote)” states, an endpoint MUST send a RST_STREAM with an error code of
QUIC_RECEIVED_RST. If the STOP_SENDING frame is received on a stream that is
already in the “half-closed (local)” or “closed” states, a RST_STREAM frame MAY
still be sent in order to cancel retransmission of previously-sent STREAM
frames.While STOP_SENDING frames are retransmittable, an implementation MAY choose not
to retransmit a lost STOP_SENDING frame if the stream has already been closed
in the appropriate direction since the frame was first generated.
See .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 2^64. 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. Stream
0 is still subject to stream-level data limits and MAX_STREAM_DATA.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 frames 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 MUST send BLOCKED frames to indicate it has data to write but is
blocked by lack of connection or stream flow control credit. BLOCKED 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 MUST 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 MUST 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
the RST_STREAM frame. Otherwise, the final offset is the offset of the end of
the data carried in STREAM frame marked with a FIN flag.An endpoint will know the final offset for a stream when the stream enters the
“half-closed (remote)” state. However, if there is reordering or loss, an
endpoint might learn the final offset prior to entering this state if it is
carried on a STREAM frame.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 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 frame
(). An endpoint MAY close the connection in this
manner, even if the error only affects a single stream.A CONNECTION_CLOSE frame could be sent in a packet that is lost. An endpoint
SHOULD be prepared to retransmit a packet containing a CONNECTION_CLOSE frame if
it receives more packets on a terminated connection. Limiting the number of
retransmissions and the time over which this final packet is sent limits the
effort expended on terminated connections.An endpoint that chooses not to retransmit packets containing CONNECTION_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 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.Some application protocols make other streams critical to that protocol. An
application protocol does not need to inform the transport that a stream is
critical; it can instead generate appropriate errors in response to being
notified that the critical stream is closed.An endpoint MAY send a RST_STREAM frame in the same packet as a CONNECTION_CLOSE
frame.Error codes are 32 bits long, with the first two bits indicating the source of
the error code:
Application-specific error codes. Defined by each application-layer protocol.
Reserved for host-local error codes. These codes MUST NOT be sent to a peer,
but MAY be used in API return codes and logs.
QUIC transport error codes, including packet protection errors. Applicable to
all uses of QUIC.
Cryptographic error codes. Defined by the cryptographic handshake protocol
in use.This section lists the defined QUIC transport error codes that may be used in a
CONNECTION_CLOSE or RST_STREAM frame. Error codes share a common code space.
Some error codes apply only to either streams or the entire connection and have
no defined semantics in the other context.
An endpoint uses this with CONNECTION_CLOSE to signal that the connection is
being closed abruptly in the absence of any error. An endpoint uses this with
RST_STREAM to signal that the stream is no longer wanted or in response to the
receipt of a RST_STREAM for that stream.
The endpoint encountered an internal error and cannot continue with the
connection.
An endpoint sends this with RST_STREAM to indicate that the stream is not
wanted and that no application action was taken for the stream. This error
code is not valid for use with CONNECTION_CLOSE.
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 (0x800001XX) 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.
Terminating stream because peer sent a RST_STREAM or STOP_SENDING.
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 (0x80000106).An attacker receives an STK from the server and then releases the IP address on
which it received the STK. 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 then spoofs
ACK frames to the server which cause the server to potentially drown the victim
in data.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_id0x0003idle_timeout0x0004omit_connection_id0x0005max_packet_size0x0006stateless_reset_tokenQUIC 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.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 for IP version 6. It is largely derived from RFC 1191, which describes Path MTU Discovery for IP version 4. [STANDARDS-TRACK]Guidelines for Writing an IANA Considerations Section in RFCsMany protocols make use of identifiers consisting of constants and other well-known values. Even after a protocol has been defined and deployment has begun, new values may need to be assigned (e.g., for a new option type in DHCP, or a new encryption or authentication transform for IPsec). To ensure that such quantities have consistent values and interpretations across all implementations, their assignment must be administered by a central authority. For IETF protocols, that role is provided by the Internet Assigned Numbers Authority (IANA).In order for IANA to manage a given namespace prudently, it needs guidelines describing the conditions under which new values can be assigned or when modifications to existing values can be made. If IANA is expected to play a role in the management of a namespace, IANA must be given clear and concise instructions describing that role. This document discusses issues that should be considered in formulating a policy for assigning values to a namespace and provides guidelines for authors on the specific text that must be included in documents that place demands on IANA.This document obsoletes RFC 2434. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.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.Hypertext 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.Structured streamsThe 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.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