QUIC J. Iyengar, Ed.
Internet-Draft Google
Intended status: Standards Track M. Thomson, Ed.
Expires: February 16, 2018 Mozilla
August 15, 2017
QUIC: A UDP-Based Multiplexed and Secure Transport
draft-ietf-quic-transport-05
Abstract
This document defines the core of the QUIC transport protocol. This
document describes connection establishment, packet format,
multiplexing and reliability. Accompanying documents describe the
cryptographic handshake and loss detection.
Note to Readers
Discussion of this draft takes place on the QUIC working group
mailing list (quic@ietf.org), which is archived at
https://mailarchive.ietf.org/arch/search/?email_list=quic.
Working Group information can be found at https://github.com/quicwg;
source code and issues list for this draft can be found at
https://github.com/quicwg/base-drafts/labels/transport.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on February 16, 2018.
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Copyright Notice
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document authors. All rights reserved.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Conventions and Definitions . . . . . . . . . . . . . . . . . 5
2.1. Notational Conventions . . . . . . . . . . . . . . . . . 5
3. A QUIC Overview . . . . . . . . . . . . . . . . . . . . . . . 6
3.1. Low-Latency Connection Establishment . . . . . . . . . . 6
3.2. Stream Multiplexing . . . . . . . . . . . . . . . . . . . 6
3.3. Rich Signaling for Congestion Control and Loss Recovery . 7
3.4. Stream and Connection Flow Control . . . . . . . . . . . 7
3.5. Authenticated and Encrypted Header and Payload . . . . . 7
3.6. Connection Migration and Resilience to NAT Rebinding . . 8
3.7. Version Negotiation . . . . . . . . . . . . . . . . . . . 8
4. Versions . . . . . . . . . . . . . . . . . . . . . . . . . . 8
5. Packet Types and Formats . . . . . . . . . . . . . . . . . . 9
5.1. Long Header . . . . . . . . . . . . . . . . . . . . . . . 9
5.2. Short Header . . . . . . . . . . . . . . . . . . . . . . 11
5.3. Version Negotiation Packet . . . . . . . . . . . . . . . 13
5.4. Cleartext Packets . . . . . . . . . . . . . . . . . . . . 14
5.4.1. Client Initial Packet . . . . . . . . . . . . . . . . 14
5.4.2. Server Stateless Retry Packet . . . . . . . . . . . . 15
5.4.3. Server Cleartext Packet . . . . . . . . . . . . . . . 16
5.4.4. Client Cleartext Packet . . . . . . . . . . . . . . . 16
5.5. Protected Packets . . . . . . . . . . . . . . . . . . . . 16
5.6. Connection ID . . . . . . . . . . . . . . . . . . . . . . 17
5.7. Packet Numbers . . . . . . . . . . . . . . . . . . . . . 17
5.7.1. Initial Packet Number . . . . . . . . . . . . . . . . 19
5.8. Handling Packets from Different Versions . . . . . . . . 19
6. Frames and Frame Types . . . . . . . . . . . . . . . . . . . 19
7. Life of a Connection . . . . . . . . . . . . . . . . . . . . 21
7.1. Version Negotiation . . . . . . . . . . . . . . . . . . . 22
7.1.1. Using Reserved Versions . . . . . . . . . . . . . . . 23
7.2. Cryptographic and Transport Handshake . . . . . . . . . . 23
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7.3. Transport Parameters . . . . . . . . . . . . . . . . . . 24
7.3.1. Transport Parameter Definitions . . . . . . . . . . . 26
7.3.2. Values of Transport Parameters for 0-RTT . . . . . . 27
7.3.3. New Transport Parameters . . . . . . . . . . . . . . 27
7.3.4. Version Negotiation Validation . . . . . . . . . . . 28
7.4. Stateless Retries . . . . . . . . . . . . . . . . . . . . 29
7.5. Proof of Source Address Ownership . . . . . . . . . . . . 29
7.5.1. Client Address Validation Procedure . . . . . . . . . 30
7.5.2. Address Validation on Session Resumption . . . . . . 31
7.5.3. Address Validation Token Integrity . . . . . . . . . 32
7.6. Connection Migration . . . . . . . . . . . . . . . . . . 32
7.6.1. Privacy Implications of Connection Migration . . . . 32
7.6.2. Address Validation for Migrated Connections . . . . . 33
7.7. Connection Termination . . . . . . . . . . . . . . . . . 34
7.8. Stateless Reset . . . . . . . . . . . . . . . . . . . . . 34
7.8.1. Detecting a Stateless Reset . . . . . . . . . . . . . 36
7.8.2. Calculating a Stateless Reset Token . . . . . . . . . 36
8. Frame Types and Formats . . . . . . . . . . . . . . . . . . . 37
8.1. PADDING Frame . . . . . . . . . . . . . . . . . . . . . . 37
8.2. RST_STREAM Frame . . . . . . . . . . . . . . . . . . . . 37
8.3. CONNECTION_CLOSE frame . . . . . . . . . . . . . . . . . 38
8.4. MAX_DATA Frame . . . . . . . . . . . . . . . . . . . . . 39
8.5. MAX_STREAM_DATA Frame . . . . . . . . . . . . . . . . . . 39
8.6. MAX_STREAM_ID Frame . . . . . . . . . . . . . . . . . . . 40
8.7. PING frame . . . . . . . . . . . . . . . . . . . . . . . 41
8.8. BLOCKED Frame . . . . . . . . . . . . . . . . . . . . . . 41
8.9. STREAM_BLOCKED Frame . . . . . . . . . . . . . . . . . . 41
8.10. STREAM_ID_NEEDED Frame . . . . . . . . . . . . . . . . . 42
8.11. NEW_CONNECTION_ID Frame . . . . . . . . . . . . . . . . . 42
8.12. STOP_SENDING Frame . . . . . . . . . . . . . . . . . . . 43
8.13. ACK Frame . . . . . . . . . . . . . . . . . . . . . . . . 43
8.13.1. ACK Block Section . . . . . . . . . . . . . . . . . 46
8.13.2. Timestamp Section . . . . . . . . . . . . . . . . . 46
8.13.3. ACK Frames and Packet Protection . . . . . . . . . . 48
8.14. STREAM Frame . . . . . . . . . . . . . . . . . . . . . . 49
9. Packetization and Reliability . . . . . . . . . . . . . . . . 51
9.1. Special Considerations for PMTU Discovery . . . . . . . . 53
10. Streams: QUIC's Data Structuring Abstraction . . . . . . . . 53
10.1. Stream Identifiers . . . . . . . . . . . . . . . . . . . 54
10.2. Life of a Stream . . . . . . . . . . . . . . . . . . . . 54
10.2.1. idle . . . . . . . . . . . . . . . . . . . . . . . . 56
10.2.2. open . . . . . . . . . . . . . . . . . . . . . . . . 56
10.2.3. half-closed (local) . . . . . . . . . . . . . . . . 57
10.2.4. half-closed (remote) . . . . . . . . . . . . . . . . 57
10.2.5. closed . . . . . . . . . . . . . . . . . . . . . . . 58
10.3. Solicited State Transitions . . . . . . . . . . . . . . 58
10.4. Stream Concurrency . . . . . . . . . . . . . . . . . . . 59
10.5. Sending and Receiving Data . . . . . . . . . . . . . . . 59
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10.6. Stream Prioritization . . . . . . . . . . . . . . . . . 60
11. Flow Control . . . . . . . . . . . . . . . . . . . . . . . . 61
11.1. Edge Cases and Other Considerations . . . . . . . . . . 62
11.1.1. Response to a RST_STREAM . . . . . . . . . . . . . . 63
11.1.2. Data Limit Increments . . . . . . . . . . . . . . . 63
11.2. Stream Limit Increment . . . . . . . . . . . . . . . . . 63
11.2.1. Blocking on Flow Control . . . . . . . . . . . . . . 64
11.3. Stream Final Offset . . . . . . . . . . . . . . . . . . 64
12. Error Handling . . . . . . . . . . . . . . . . . . . . . . . 65
12.1. Connection Errors . . . . . . . . . . . . . . . . . . . 65
12.2. Stream Errors . . . . . . . . . . . . . . . . . . . . . 66
12.3. Error Codes . . . . . . . . . . . . . . . . . . . . . . 66
13. Security and Privacy Considerations . . . . . . . . . . . . . 68
13.1. Spoofed ACK Attack . . . . . . . . . . . . . . . . . . . 68
13.2. Slowloris Attacks . . . . . . . . . . . . . . . . . . . 68
13.3. Stream Fragmentation and Reassembly Attacks . . . . . . 69
13.4. Stream Commitment Attack . . . . . . . . . . . . . . . . 69
14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 70
14.1. QUIC Transport Parameter Registry . . . . . . . . . . . 70
15. References . . . . . . . . . . . . . . . . . . . . . . . . . 71
15.1. Normative References . . . . . . . . . . . . . . . . . . 71
15.2. Informative References . . . . . . . . . . . . . . . . . 72
15.3. URIs . . . . . . . . . . . . . . . . . . . . . . . . . . 73
Appendix A. Contributors . . . . . . . . . . . . . . . . . . . . 73
Appendix B. Acknowledgments . . . . . . . . . . . . . . . . . . 73
Appendix C. Change Log . . . . . . . . . . . . . . . . . . . . . 74
C.1. Since draft-ietf-quic-transport-04 . . . . . . . . . . . 74
C.2. Since draft-ietf-quic-transport-03 . . . . . . . . . . . 74
C.3. Since draft-ietf-quic-transport-02 . . . . . . . . . . . 75
C.4. Since draft-ietf-quic-transport-01 . . . . . . . . . . . 76
C.5. Since draft-ietf-quic-transport-00 . . . . . . . . . . . 78
C.6. Since draft-hamilton-quic-transport-protocol-01 . . . . . 78
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 78
1. Introduction
QUIC is a multiplexed and secure transport protocol that runs on top
of UDP. QUIC aims to provide a flexible set of features that allow
it to be a general-purpose 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
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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 [QUIC-RECOVERY], and the use of TLS 1.3 for key negotiation
[QUIC-TLS].
2. Conventions and Definitions
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 [RFC2119].
Definitions of terms that are used in this document:
Client: The endpoint initiating a QUIC connection.
Server: The endpoint accepting incoming QUIC connections.
Endpoint: The client or server end of a connection.
Stream: A logical, bi-directional channel of ordered bytes within a
QUIC connection.
Connection: A conversation between two QUIC endpoints with a single
encryption context that multiplexes streams within it.
Connection ID: The identifier for a QUIC connection.
QUIC packet: 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.
2.1. Notational Conventions
Packet and frame diagrams use the format described in [RFC2360]
Section 3.1, with the following additional conventions:
[x] Indicates that x is optional
{x} Indicates that x is encrypted
x (A) Indicates that x is A bits long
x (A/B/C) ... Indicates that x is one of A, B, or C bits long
x (*) ... Indicates that x is variable-length
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3. A QUIC Overview
This section briefly describes QUIC's key mechanisms and benefits.
Key strengths of QUIC include:
o Low-latency connection establishment
o Multiplexing without head-of-line blocking
o Authenticated and encrypted header and payload
o Rich signaling for congestion control and loss recovery
o Stream and connection flow control
o Connection migration and resilience to NAT rebinding
o Version negotiation
3.1. Low-Latency Connection Establishment
QUIC 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 [QUIC-TLS].
3.2. Stream Multiplexing
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.
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3.3. Rich Signaling for Congestion Control and Loss Recovery
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.
3.4. Stream and Connection Flow Control
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.
3.5. Authenticated and Encrypted Header and Payload
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
[RFC6824] 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
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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.
3.6. Connection Migration and Resilience to NAT Rebinding
QUIC connections are identified by a 64-bit Connection ID, randomly
generated by the server. QUIC's consistent connection ID allows
connections to survive changes to the client's IP and port, such as
those caused by NAT rebindings or by the client changing network
connectivity to a new address. QUIC provides automatic cryptographic
verification of a rebound client, since the client continues to use
the same session key for encrypting and decrypting packets. The
consistent connection ID can be used to allow migration of the
connection to a new server IP address as well, since the Connection
ID remains consistent across changes in the client's and the server's
network addresses.
3.7. Version Negotiation
QUIC version negotiation allows for multiple versions of the protocol
to be deployed and used concurrently. Version negotiation is
described in Section 7.1.
4. Versions
QUIC versions are identified using a 32-bit value.
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 [QUIC-TLS].
Versions with the most significant 16 bits of the version number
cleared are reserved for use in future IETF consensus documents.
Versions that follow the pattern 0x?a?a?a?a are reserved for use in
forcing version negotiation to be exercised. That is, any version
number where the low four bits of all octets is 1010 (in binary). A
client or server MAY advertise support for any of these reserved
versions.
Reserved version numbers will probably never represent a real
protocol; a client MAY use one of these version numbers with the
expectation that the server will initiate version negotiation; a
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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 [4].
5. Packet Types and Formats
We first describe QUIC's packet types and their formats, since some
are referenced in subsequent mechanisms.
All numeric values are encoded in network byte order (that is, big-
endian) and all field sizes are in bits. When discussing individual
bits of fields, the least significant bit is referred to as bit 0.
Hexadecimal notation is used for describing the value of fields.
Any QUIC packet has either a long or a short header, as indicated by
the Header Form bit. Long headers are expected to be used early in
the connection before version negotiation and establishment of 1-RTT
keys. Short headers are minimal version-specific headers, which can
be used after version negotiation and 1-RTT keys are established.
5.1. Long Header
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+
|1| Type (7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ Connection ID (64) +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Packet Number (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Version (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Payload (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 1: Long Header Format
Long headers are used for packets that are sent prior to the
completion of version negotiation and establishment of 1-RTT keys.
Once both conditions are met, a sender 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:
Header Form: The most significant bit (0x80) of octet 0 (the first
octet) is set to 1 for long headers.
Long Packet Type: The remaining seven bits of octet 0 contain the
packet type. This field can indicate one of 128 packet types.
The types specified for this version are listed in Table 1.
Connection ID: Octets 1 through 8 contain the connection ID.
Section 5.6 describes the use of this field in more detail.
Packet Number: Octets 9 to 12 contain the packet number.
Section 5.7 describes the use of packet numbers.
Version: 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.
Payload: Octets from 17 onwards (the rest of QUIC packet) are the
payload of the packet.
The following packet types are defined:
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+------+-------------------------------+---------------+
| Type | Name | Section |
+------+-------------------------------+---------------+
| 0x01 | Version Negotiation | Section 5.3 |
| | | |
| 0x02 | Client Initial | Section 5.4.1 |
| | | |
| 0x03 | Server Stateless Retry | Section 5.4.2 |
| | | |
| 0x04 | Server Cleartext | Section 5.4.3 |
| | | |
| 0x05 | Client Cleartext | Section 5.4.4 |
| | | |
| 0x06 | 0-RTT Protected | Section 5.5 |
| | | |
| 0x07 | 1-RTT Protected (key phase 0) | Section 5.5 |
| | | |
| 0x08 | 1-RTT Protected (key phase 1) | Section 5.5 |
+------+-------------------------------+---------------+
Table 1: Long Header Packet Types
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 Table 1 are version-specific. See
Section 5.8 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.
5.2. Short Header
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+
|0|C|K| Type (5)|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ [Connection ID (64)] +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Packet Number (8/16/32) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Protected Payload (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2: Short Header Format
The short header can be used after the version and 1-RTT keys are
negotiated. This header form has the following fields:
Header Form: The most significant bit (0x80) of octet 0 is set to 0
for the short header.
Connection ID Flag: 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 (Section 7.3.1) is
specified by the intended recipient of the packet.
Key Phase Bit: 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
[QUIC-TLS] for details.
Short Packet Type: The remaining 5 bits of octet 0 include one of 32
packet types. Table 2 lists the types that are defined for short
packets.
Connection ID: If the Connection ID Flag is set, a connection ID
occupies octets 1 through 8 of the packet. See Section 5.6 for
more details.
Packet Number: 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.
Protected Payload: Packets with a short header always include a
1-RTT protected payload.
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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.
+------+--------------------+
| Type | Packet Number Size |
+------+--------------------+
| 0x01 | 1 octet |
| | |
| 0x02 | 2 octets |
| | |
| 0x03 | 4 octets |
+------+--------------------+
Table 2: Short Header Packet Types
The 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 Section 5.8 for details
on how packets from different versions of QUIC are interpreted.
5.3. Version Negotiation Packet
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.
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Supported Version 1 (32) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Supported Version 2 (32)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Supported Version N (32)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 3: Version Negotiation Packet
See Section 7.1 for a description of the version negotiation process.
5.4. Cleartext Packets
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 [QUIC-TLS].
5.4.1. Client Initial Packet
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 (Section 5.7).
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 Section 9). The stream in this packet always starts at an
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offset of 0 (see Section 7.4) and the complete cyptographic handshake
message MUST fit in a single packet (see Section 7.2).
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 (Section 5.3) or Server Stateless
Retry packet (Section 5.4.2).
5.4.2. 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 Section 7.4).
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. In effect, the next
cryptographic handshake message is sent on a new connection. The new
Client Initial packet is sent in a packet with a newly randomized
packet number and starting at a stream offset of 0.
Continuing the cryptographic handshake is necessary to ensure that an
attacker cannot force a downgrade of any cryptographic parameters.
In addition to continuing the cryptographic handshake, the client
MUST remember the results of any version negotiation that occurred
(see Section 7.1). 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.
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5.4.3. Server Cleartext Packet
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 Section 5.6).
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 Section 5.7.
The payload of this packet contains STREAM frames and could contain
PADDING and ACK frames.
5.4.4. Client Cleartext Packet
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 Section 5.6.
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 Section 5.7.
The payload of this packet contains STREAM frames and could contain
PADDING and ACK frames.
5.5. Protected Packets
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
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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
[QUIC-TLS] 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 Section 5.6.
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 Section 5.7 for details.
The payload is protected using authenticated encryption. [QUIC-TLS]
describes packet protection in detail. After decryption, the
plaintext consists of a sequence of frames, as described in
Section 6.
5.6. Connection ID
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 (Section 5.4.1) and 0-RTT packets (Section 5.5). 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.
5.7. Packet Numbers
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
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number for sending MUST increase by at least one after sending any
packet, unless otherwise specified (see Section 5.7.1).
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; the sender MAY send a Public Reset packet 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 (Section 5.3) and Server Stateless Retry
(Section 5.4.2) packets have special rules for populating the packet
number field.
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5.7.1. Initial Packet Number
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. [RFC4086] 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 (Section 5.3) or Server
Stateless Retry packet (Section 5.4.2) MUST generate a new initial
packet number. This ensures that the first transmission attempt for
a Client Initial packet (Section 5.4.1) 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.
5.8. Handling Packets from Different Versions
Between different versions the following things are guaranteed to
remain constant:
o the location of the header form flag,
o the location of the Connection ID flag in short headers,
o the location and size of the Connection ID field in both header
forms,
o the location and size of the Version field in long headers, and
o the 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.
6. Frames and Frame Types
The payload of cleartext packets and the plaintext after decryption
of protected payloads consists of a sequence of frames, as shown in
Figure 4.
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Frame 1 (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Frame 2 (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Frame N (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 4: Contents of Protected Payload
Protected payloads MUST contain at least one frame, and MAY contain
multiple frames and multiple frame types.
Frames MUST fit within a single QUIC packet and MUST NOT span a QUIC
packet boundary. Each frame begins with a Frame Type byte,
indicating its type, followed by additional type-dependent fields:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Type (8) | Type-Dependent Fields (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 5: Generic Frame Layout
Frame types are listed in Table 3. 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.
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+-------------+-------------------+--------------+
| Type Value | Frame Type Name | Definition |
+-------------+-------------------+--------------+
| 0x00 | PADDING | Section 8.1 |
| | | |
| 0x01 | RST_STREAM | Section 8.2 |
| | | |
| 0x02 | CONNECTION_CLOSE | Section 8.3 |
| | | |
| 0x04 | MAX_DATA | Section 8.4 |
| | | |
| 0x05 | MAX_STREAM_DATA | Section 8.5 |
| | | |
| 0x06 | MAX_STREAM_ID | Section 8.6 |
| | | |
| 0x07 | PING | Section 8.7 |
| | | |
| 0x08 | BLOCKED | Section 8.8 |
| | | |
| 0x09 | STREAM_BLOCKED | Section 8.9 |
| | | |
| 0x0a | STREAM_ID_NEEDED | Section 8.10 |
| | | |
| 0x0b | NEW_CONNECTION_ID | Section 8.11 |
| | | |
| 0x0c | STOP_SENDING | Section 8.12 |
| | | |
| 0xa0 - 0xbf | ACK | Section 8.13 |
| | | |
| 0xc0 - 0xff | STREAM | Section 8.14 |
+-------------+-------------------+--------------+
Table 3: Frame Types
7. Life of a Connection
A QUIC connection is a single conversation between two QUIC
endpoints. QUIC's connection establishment intertwines version
negotiation with the cryptographic and transport handshakes to reduce
connection establishment latency, as described in Section 7.2. Once
established, a connection may migrate to a different IP or port at
either endpoint, due to NAT rebinding or mobility, as described in
Section 7.6. Finally a connection may be terminated by either
endpoint, as described in Section 7.7.
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7.1. Version Negotiation
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 Section 7.2,
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 (Section 5.3). This includes a list of
versions that the server will accept.
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 (Section 7.2). 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
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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 Section 7.3.4).
7.1.1. Using Reserved Versions
For a server to use a new version in the future, clients must
correctly handle unsupported versions. To help ensure this, a server
SHOULD include a reserved version (see Section 4) 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 Section 7.3.4). 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.
7.2. Cryptographic and Transport Handshake
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 [QUIC-TLS]; 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:
o authenticated key exchange, where
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* a server is always authenticated,
* a client is optionally authenticated,
* every connection produces distinct and unrelated keys,
* keying material is usable for packet protection for both 0-RTT
and 1-RTT packets, and
* 1-RTT keys have forward secrecy
o authenticated values for the transport parameters of the peer (see
Section 7.3)
o authenticated confirmation of version negotiation (see
Section 7.3.4)
o authenticated negotiation of an application protocol (TLS uses
ALPN [RFC7301] for this purpose)
o 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 Section 7.5)
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 [QUIC-TLS].
7.3. Transport Parameters
During connection establishment, both endpoints make authenticated
declarations of their transport parameters. These declarations are
made unilaterally by each endpoint. Endpoints are required to comply
with the restrictions implied by these parameters; the description of
each parameter includes rules for its handling.
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The format of the transport parameters is the TransportParameters
struct from Figure 6. This is described using the presentation
language from Section 3 of [I-D.ietf-tls-tls13].
uint32 QuicVersion;
enum {
initial_max_stream_data(0),
initial_max_data(1),
initial_max_stream_id(2),
idle_timeout(3),
omit_connection_id(4),
max_packet_size(5),
stateless_reset_token(6),
(65535)
} TransportParameterId;
struct {
TransportParameterId parameter;
opaque value<0..2^16-1>;
} TransportParameter;
struct {
select (Handshake.msg_type) {
case client_hello:
QuicVersion negotiated_version;
QuicVersion initial_version;
case encrypted_extensions:
QuicVersion supported_versions<2..2^8-4>;
case new_session_ticket:
struct {};
};
TransportParameter parameters<30..2^16-1>;
} TransportParameters;
Figure 6: Definition of TransportParameters
The "extension_data" field of the quic_transport_parameters extension
defined in [QUIC-TLS] contains a TransportParameters value. TLS
encoding rules are therefore used to encode the transport parameters.
QUIC encodes transport parameters into a sequence of octets, which
are then included in the cryptographic handshake. Once the handshake
completes, the transport parameters declared by the peer are
available. Each endpoint validates the value provided by its peer.
In particular, version negotiation MUST be validated (see
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Section 7.3.4) before the connection establishment is considered
properly complete.
Definitions for each of the defined transport parameters are included
in Section 7.3.1.
7.3.1. Transport Parameter Definitions
An endpoint MUST include the following parameters in its encoded
TransportParameters:
initial_max_stream_data (0x0000): 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 (Section 8.5) being sent on
all streams immediately after opening.
initial_max_data (0x0001): The initial maximum data parameter
contains the initial value for the maximum amount of data that can
be sent on the connection. This parameter is encoded as an
unsigned 32-bit integer in units of 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 (Section 8.4) for
the connection immediately after completing the handshake.
initial_max_stream_id (0x0002): 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 (Section 8.6) immediately
after completing the handshake.
idle_timeout (0x0003): The idle timeout is a value in seconds that
is encoded as an unsigned 16-bit integer. The maximum value is
600 seconds (10 minutes).
stateless_reset_token (0x0005): The Stateless Reset Token is used in
verifying a stateless reset, see Section 7.8. This parameter is a
sequence of 16 octets.
An endpoint MAY use the following transport parameters:
omit_connection_id (0x0004): 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
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that the endpoint relies on the connection ID being present in
every packet.
max_packet_size (0x0005): The maximum packet size parameter places a
limit on the size of packets that the endpoint is willing to
receive, encoded as an unsigned 16-bit integer. This indicates
that packets larger than this limit will be dropped. The default
for this parameter is the maximum permitted UDP payload of 65527.
Values below 1200 are invalid. This limit only applies to
protected packets (Section 5.5).
7.3.2. Values of Transport Parameters for 0-RTT
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.
7.3.3. New Transport Parameters
New transport parameters can be used to negotiate new protocol
behavior. An endpoint MUST ignore transport parameters that it does
not support. Absence of a transport parameter therefore disables any
optional protocol feature that is negotiated using the parameter.
New transport parameters can be registered according to the rules in
Section 14.1.
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7.3.4. Version Negotiation Validation
The transport parameters include three fields that encode version
information. These retroactively authenticate the version
negotiation (see Section 7.1) 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
Section 7.3). As a result, modification of version negotiation
packets by an attacker can be detected.
The client includes two fields in the transport parameters:
o 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.
o The initial_version is the version that the client initially
attempted to use. If the server did not send a version
negotiation packet Section 5.3, 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 (Section 5.3) 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.
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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.
7.4. Stateless Retries
A server can process an initial cryptographic handshake messages from
a client without committing any state. This allows a server to
perform address validation (Section 7.5, 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
(Section 5.4.2). 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.
7.5. Proof of Source Address Ownership
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.
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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.
7.5.1. Client Address Validation Procedure
QUIC uses token-based address validation. Any time the server wishes
to validate a client address, it provides the client with a token.
As long as the token cannot be easily guessed (see Section 7.5.3), 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
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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.
7.5.2. Address Validation on Session Resumption
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 Section 7.6). 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.
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7.5.3. Address Validation Token Integrity
An address validation token MUST be difficult to guess. Including a
large enough random value in the token would be sufficient, but this
depends on the server remembering the value it sends to clients.
A token-based scheme allows the server to offload any state
associated with validation to the client. For this design to work,
the token MUST be covered by integrity protection against
modification or falsification by clients. Without integrity
protection, malicious clients could generate or guess values for
tokens that would be accepted by the server. Only the server
requires access to the integrity protection key for tokens.
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.
7.6. Connection Migration
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.
7.6.1. Privacy Implications of Connection Migration
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
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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 Section 7.6.1.1. 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.
7.6.1.1. Packet Number Gap
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:
Gap = HKDF-Expand-Label(packet_number_secret,
"QUIC packet sequence gap", sequence, 4)
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 [QUIC-TLS] Section 5.6.
7.6.2. Address Validation for Migrated Connections
TODO: see issue #161
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7.7. Connection Termination
Connections should remain open until they become idle for a pre-
negotiated period of time. A QUIC connection, once established, can
be terminated in one of three ways:
1. Explicit Shutdown: An endpoint sends a CONNECTION_CLOSE frame to
terminate the connection. An endpoint MAY use application-layer
mechanisms prior to a CONNECTION_CLOSE to indicate that the
connection will soon be terminated. On termination of the active
streams, a CONNECTION_CLOSE may be sent. If an endpoint sends a
CONNECTION_CLOSE frame while unterminated streams are active (no
FIN bit or RST_STREAM frames have been sent or received for one
or more streams), then the peer must assume that the streams were
incomplete and were abnormally terminated.
2. Implicit Shutdown: The default idle timeout is a required
parameter in connection negotiation. The maximum is 10 minutes.
If there is no network activity for the duration of the idle
timeout, the connection is closed. By default a CONNECTION_CLOSE
frame will be sent. A silent close option can be enabled when it
is expensive to send an explicit close, such as mobile networks
that must wake up the radio.
3. Stateless Reset: An endpoint that loses state can use this
procedure to cause the connection to terminate early, see
Section 7.8 for details.
After receiving either a CONNECTION_CLOSE frame or a Public Reset, an
endpoint MUST NOT send additional packets on that connection. After
sending either a CONNECTION_CLOSE frame or a Public Reset packet,
implementations MUST NOT send any non-closing packets on that
connection. If additional packets are received after this time and
before idle_timeout seconds has passed, implementations SHOULD
respond to them by sending a CONNECTION_CLOSE (which MAY just be a
duplicate of the previous CONNECTION_CLOSE packet) but MAY also send
a Public Reset packet. Implementations SHOULD throttle these
responses, for instance by exponentially backing off the number of
packets which are received before sending a response. After this
time, implementations SHOULD respond to unexpected packets with a
Public Reset packet.
7.8. Stateless Reset
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.
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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:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+
|0|C|K| 00001 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ [Connection ID (64)] +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ +
| |
+ Stateless Reset Token (128) +
| |
+ +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Random Octets (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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.
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After the Stateless Reset Token, the endpoint 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.
7.8.1. Detecting a Stateless Reset
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.
7.8.2. Calculating a Stateless Reset Token
The stateless reset token MUST be difficult to guess. In order to
create a Stateless Reset Token, a server could randomly generate
[RFC4086] 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 Section 5.6), and an
identifier for the server instance. A server could use HMAC
[RFC2104] (for example, HMAC(static_key, server_id || connection_id))
or HKDF [RFC5869] (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
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cannot allow clients to omit a connection ID (that is, it cannot use
the truncate_connection_id transport parameter Section 7.3.1).
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.
8. Frame Types and Formats
As described in Section 6, 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.
8.1. PADDING Frame
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.
8.2. RST_STREAM 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 of STREAM
frames on the identified stream. A receiver of RST_STREAM can
discard any data that it already received on that stream. An
endpoint sends a RST_STREAM in response to a RST_STREAM unless the
stream is already closed.
The RST_STREAM frame is as follows:
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Error Code (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ Final Offset (64) +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The fields are:
Stream ID: The 32-bit Stream ID of the stream being terminated.
Error code: A 32-bit error code which indicates why the stream is
being closed.
Final offset: A 64-bit unsigned integer indicating the absolute byte
offset of the end of data written on this stream by the RST_STREAM
sender.
8.3. CONNECTION_CLOSE frame
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:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Error Code (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Reason Phrase Length (16) | [Reason Phrase (*)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The fields of a CONNECTION_CLOSE frame are as follows:
Error Code: A 32-bit error code which indicates the reason for
closing this connection.
Reason Phrase Length: 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.
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Reason Phrase: A human-readable explanation for why the connection
was closed. This can be zero length if the sender chooses to not
give details beyond the Error Code. This SHOULD be a UTF-8
encoded string [RFC3629].
8.4. MAX_DATA Frame
The MAX_DATA frame (type=0x04) is used in flow control to inform the
peer of the maximum amount of data that can be sent on the connection
as a whole.
The frame is as follows:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ Maximum Data (64) +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The fields in the MAX_DATA frame are as follows:
Maximum Data: 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 Section 7.3.2).
8.5. MAX_STREAM_DATA Frame
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:
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ Maximum Stream Data (64) +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The fields in the MAX_STREAM_DATA frame are as follows:
Stream ID: The stream ID of the stream that is affected.
Maximum Stream Data: 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 Section 7.3.2).
8.6. MAX_STREAM_ID Frame
The MAX_STREAM_ID frame (type=0x06) informs the peer of the maximum
stream ID that they are permitted to open.
The frame is as follows:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Maximum Stream ID (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The fields in the MAX_STREAM_ID frame are as follows:
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Maximum Stream ID: 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
Section 7.3.2).
8.7. PING frame
Endpoints can use PING frames (type=0x07) to verify that their peers
are still alive or to check reachability to the peer. The PING frame
contains no additional fields. The receiver of a PING frame simply
needs to acknowledge the packet containing this frame. The PING
frame SHOULD be used to keep a connection alive when a stream is
open. The default is to send a PING frame after 15 seconds of
quiescence. A PING frame has no additional fields.
8.8. BLOCKED Frame
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
Section 11.2.1). BLOCKED frames can be used as input to tuning of
flow control algorithms (see Section 11.1.2).
The BLOCKED frame does not contain a payload.
8.9. STREAM_BLOCKED Frame
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 (Section 8.8).
The STREAM_BLOCKED frame is as follows:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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The STREAM_BLOCKED frame contains a single field:
Stream ID: A 32-bit unsigned number indicating the stream which is
flow control blocked.
An endpoint MAY send a STREAM_BLOCKED frame for a stream that exceeds
the maximum stream ID set by its peer (see Section 8.6). 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.
8.10. STREAM_ID_NEEDED Frame
A sender sends a STREAM_ID_NEEDED frame (type=0x0a) when it wishes to
open a stream, but is unable to due to the maximum stream ID limit.
The STREAM_ID_NEEDED frame does not contain a payload.
8.11. NEW_CONNECTION_ID Frame
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 Section 7.6.1).
The NEW_CONNECTION_ID is as follows:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Sequence (16) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ Connection ID (64) +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ +
| |
+ Stateless Reset Token (128) +
| |
+ +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The fields are:
Sequence: 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
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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.
Connection ID: A 64-bit connection ID.
Stateless Reset Token: A 128-bit value that will be used to for a
stateless reset when the associated connection ID is used (see
Section 7.8).
8.12. STOP_SENDING Frame
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:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Error Code (32) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The fields are:
Stream ID: The 32-bit Stream ID of the stream being ignored.
Error Code: The application-specified reason the sender is ignoring
the stream.
8.13. ACK Frame
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.
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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:
o The first three bits must be set to 101 indicating that this is an
ACK frame.
o The "N" bit indicates whether the frame contains a Num Blocks
field.
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o 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.
o 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.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|[Num Blocks(8)]| NumTS (8) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Largest Acknowledged (8/16/32/64) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ACK Delay (16) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ACK Block Section (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Timestamp Section (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 7: ACK Frame Format
The fields in the ACK frame are as follows:
Num Blocks (opt): 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.
Num Timestamps: An unsigned 8-bit number specifying the total number
of <packet number, timestamp> pairs in the Timestamp Section.
Largest Acknowledged: 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.)
ACK Delay: 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.
ACK Block Section: Contains one or more blocks of packet numbers
which have been successfully received, see Section 8.13.1.
Timestamp Section: Contains zero or more timestamps reporting
transit delay of received packets. See Section 8.13.2.
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8.13.1. ACK Block Section
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.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| First ACK Block Length (8/16/32/64) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Gap 1 (8)] | [ACK Block 1 Length (8/16/32/64)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Gap 2 (8)] | [ACK Block 2 Length (8/16/32/64)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Gap N (8)] | [ACK Block N Length (8/16/32/64)] ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 8: ACK Block Section
The fields in the ACK Block Section are:
First ACK Block Length: An unsigned packet number delta that
indicates the number of contiguous additional packets being
acknowledged starting at the Largest Acknowledged.
Gap To Next Block (opt, repeated): 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.
ACK Block Length (opt, repeated): 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.
8.13.2. Timestamp Section
The Timestamp Section contains between zero and 255 measurements of
packet receive times relative to the beginning of the connection.
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+
| [Delta LA (8)]|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [First Timestamp (32)] |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|[Delta LA 1(8)]| [Time Since Previous 1 (16)] |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|[Delta LA 2(8)]| [Time Since Previous 2 (16)] |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|[Delta LA N(8)]| [Time Since Previous N (16)] |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 9: Timestamp Section
The fields in the Timestamp Section are:
Delta Largest Acknowledged (opt): 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.)
First Timestamp (opt): 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.
Delta Largest Acked 1..N (opt, repeated): This field has the same
semantics and format as "Delta Largest Acknowledged". Repeated
"Num Timestamps - 1" times.
Time Since Previous Timestamp 1..N(opt, repeated): 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.
8.13.2.1. Time Format
DISCUSS_AND_REPLACE: Perhaps make this format simpler.
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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.
8.13.3. ACK Frames and Packet Protection
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
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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.
8.14. STREAM Frame
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:
o The first two bits must be set to 11, indicating that this is a
STREAM frame.
o "F" is the FIN bit, which is used for stream termination.
o 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.
o 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.
o 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.
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream ID (8/16/24/32) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Offset (0/16/32/64) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| [Data Length (16)] | Stream Data (*) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 10: STREAM Frame Format
The STREAM frame contains the following fields:
Stream ID: The stream ID of the stream (see Section 10.1).
Offset: 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.
Data Length: 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.
Stream Data: 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.
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9. Packetization and Reliability
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
([RFC4821]) and MAY use PMTU Discovery ([RFC1191], [RFC1981]) 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
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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
Section 6).
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:
o 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.
o ACK and PADDING frames MUST NOT be retransmitted. ACK frames
containing updated information will be sent as described in
Section 8.13.
o STOP_SENDING frames MUST be retransmitted, unless the stream has
become closed in the appropriate direction. See Section 10.3.
o 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 [QUIC-RECOVERY].
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
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(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
generate an ACK frame in response to a packet containing only ACK or
PADDING frames.
Strategies and implications of the frequency of generating
acknowledgments are discussed in more detail in [QUIC-RECOVERY].
9.1. Special Considerations for PMTU Discovery
Traditional ICMP-based path MTU discovery in IPv4 [RFC1191] 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:
o 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.
o 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.
o 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.
10. Streams: QUIC's Data Structuring Abstraction
Streams in QUIC provide a lightweight, ordered, and bidirectional
byte-stream abstraction modeled closely on HTTP/2 streams [RFC7540].
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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
[SST], which may be a more appealing description for some
applications.
10.1. Stream Identifiers
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 (Section 8.14) permits a shorter encoding when the leading bits
of the stream ID are zero.
10.2. Life of a Stream
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 [RFC7540], with some differences to accommodate the
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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.
+--------+
| |
| idle |
| |
+--------+
|
send/recv STREAM/RST
recv MSD/SB
|
v
recv FIN/ +--------+ send FIN/
recv RST | | send RST
,---------| open |-----------.
/ | | \
v +--------+ v
+----------+ +----------+
| half | | half |
| closed | | closed |
| (remote) | | (local) |
+----------+ +----------+
| |
| send FIN/ +--------+ recv FIN/ |
\ send RST | | recv RST /
`----------->| 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
Figure 11: Lifecycle of a stream
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.
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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 Section 12).
10.2.1. idle
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 Section 10.1. 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
Section 8.6).
10.2.2. open
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
Section 11).
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.
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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.
10.2.3. half-closed (local)
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 Section 10.2.5 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.
10.2.4. half-closed (remote)
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.
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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
Section 11.3 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 Section 11).
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.
10.2.5. closed
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 Section 10.2.4.
10.3. Solicited State Transitions
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.
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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 Section 9.
10.4. Stream Concurrency
An endpoint limits the number of concurrently active incoming streams
by adjusting the maximum stream ID. An initial value is set in the
transport parameters (see Section 7.3.1) and is subsequently
increased by MAX_STREAM_ID frames (see Section 8.6).
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
(Section 12), unless this is a result of a change in the initial
offsets (see Section 7.3.2).
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.
10.5. Sending and Receiving Data
Once a stream is created, endpoints may use the stream to send and
receive data. Each endpoint may send a series of STREAM frames
encapsulating data on a stream until the stream is terminated in that
direction. Streams are an ordered byte-stream abstraction, and they
have no other structure within them. STREAM frame boundaries are not
expected to be preserved in retransmissions from the sender or during
delivery to the application at the receiver.
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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 Section 11, and congestion
control is described in the companion document [QUIC-RECOVERY].
10.6. Stream Prioritization
Stream multiplexing has a significant effect on application
performance if resources allocated to streams are correctly
prioritized. Experience with other multiplexed protocols, such as
HTTP/2 [RFC7540], 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.
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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.
11. Flow Control
It is necessary to limit the amount of data that a sender may have
outstanding at any time, so as to prevent a fast sender from
overwhelming a slow receiver, or to prevent a malicious sender from
consuming significant resources at a receiver. This section
describes QUIC's flow-control mechanisms.
QUIC employs a credit-based flow-control scheme similar to HTTP/2's
flow control [RFC7540]. 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 receiver sends MAX_DATA or MAX_STREAM_DATA frames to the sender to
advertise additional credit by sending the absolute byte offset in
the connection or stream which it is willing to receive.
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
(Section 12) if the peer violates the advertised connection or stream
data limits.
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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.
11.1. Edge Cases and Other Considerations
There are some edge cases which must be considered when dealing with
stream and connection level flow control. Given enough time, both
endpoints must agree on flow control state. If one end believes it
can send more than the other end is willing to receive, the
connection will be torn down when too much data arrives.
Conversely if a sender believes it is blocked, while endpoint B
expects more data can be received, then the connection can be in a
deadlock, with the sender waiting for a MAX_DATA or MAX_STREAM_DATA
frame which will never come.
On receipt of a RST_STREAM frame, an endpoint will tear down state
for the matching stream and ignore further data arriving on that
stream. This could result in the endpoints getting out of sync,
since the RST_STREAM frame may have arrived out of order and there
may be further bytes in flight. The data sender would have counted
the data against its connection level flow control budget, but a
receiver that has not received these bytes would not know to include
them as well. The receiver must learn the number of bytes that were
sent on the stream to make the same adjustment in its connection flow
controller.
To avoid this de-synchronization, a RST_STREAM sender MUST include
the final byte offset sent on the stream in the RST_STREAM frame. On
receiving a RST_STREAM frame, a receiver definitively knows how many
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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.
11.1.1. Response to a RST_STREAM
RST_STREAM terminates one direction of a stream abruptly. Whether
any action or response can or should be taken on the data already
received is an application-specific issue, but it will often be the
case that upon receipt of a RST_STREAM an endpoint will choose to
stop sending data in its own direction. If the sender of a
RST_STREAM wishes to explicitly state that no future data will be
processed, that endpoint MAY send a STOP_SENDING frame at the same
time.
11.1.2. Data Limit Increments
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.
11.2. Stream Limit Increment
As with flow control, this document leaves when and how many streams
to make available to a peer via MAX_STREAM_ID to implementations, but
offers a few considerations. MAX_STREAM_ID frames constitute minimal
overhead, while withholding MAX_STREAM_ID frames can prevent the peer
from using the available parallelism.
Implementations will likely want to increase the maximum stream ID as
peer-initiated streams close. A receiver MAY also advance the
maximum stream ID based on current activity, system conditions, and
other environmental factors.
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11.2.1. Blocking on Flow Control
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.
11.3. Stream Final Offset
The final offset is the count of the number of octets that are
transmitted on a stream. For a stream that is reset, the final
offset is carried explicitly in 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 Section 12). 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
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that the endpoint needs to maintain the final offset state for closed
streams, which could mean a significant state commitment.
12. Error Handling
An endpoint that detects an error SHOULD signal the existence of that
error to its peer. Errors can affect an entire connection (see
Section 12.1), or a single stream (see Section 12.2).
The most appropriate error code (Section 12.3) 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 (Section 7.8) 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.
12.1. Connection Errors
Errors that result in the connection being unusable, such as an
obvious violation of protocol semantics or corruption of state that
affects an entire connection, MUST be signaled using a
CONNECTION_CLOSE frame (Section 8.3). 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 (Section 7.8).
An endpoint that receives an invalid CONNECTION_CLOSE frame MUST NOT
signal the existence of the error to its peer.
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12.2. Stream Errors
If the error affects a single stream, but otherwise leaves the
connection in a recoverable state, the endpoint can send a RST_STREAM
frame (Section 8.2) 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.
12.3. Error Codes
Error codes are 32 bits long, with the first two bits indicating the
source of the error code:
0x00000000-0x3FFFFFFF: Application-specific error codes. Defined by
each application-layer protocol.
0x40000000-0x7FFFFFFF: 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.
0x80000000-0xBFFFFFFF: QUIC transport error codes, including packet
protection errors. Applicable to all uses of QUIC.
0xC0000000-0xFFFFFFFF: 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.
NO_ERROR (0x80000000): 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
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signal that the stream is no longer wanted or in response to the
receipt of a RST_STREAM for that stream.
INTERNAL_ERROR (0x80000001): The endpoint encountered an internal
error and cannot continue with the connection.
CANCELLED (0x80000002): 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.
FLOW_CONTROL_ERROR (0x80000003): An endpoint received more data than
it permitted in its advertised data limits (see Section 11).
STREAM_ID_ERROR (0x80000004): An endpoint received a frame for a
stream identifier that exceeded its advertised maximum stream ID.
STREAM_STATE_ERROR (0x80000005): An endpoint received a frame for a
stream that was not in a state that permitted that frame (see
Section 10.2).
FINAL_OFFSET_ERROR (0x80000006): An endpoint received a STREAM frame
containing data that exceeded the previously established final
offset. Or an endpoint received a RST_STREAM frame containing a
final offset that was lower than the maximum offset of data that
was already received. Or an endpoint received a RST_STREAM frame
containing a different final offset to the one already
established.
FRAME_FORMAT_ERROR (0x80000007): 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.
TRANSPORT_PARAMETER_ERROR (0x80000008): An endpoint received
transport parameters that were badly formatted, included an
invalid value, was absent even though it is mandatory, was present
though it is forbidden, or is otherwise in error.
VERSION_NEGOTIATION_ERROR (0x80000009): 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.
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PROTOCOL_VIOLATION (0x8000000A): An endpoint detected an error with
protocol compliance that was not covered by more specific error
codes.
QUIC_RECEIVED_RST (0x80000035): Terminating stream because peer sent
a RST_STREAM or STOP_SENDING.
FRAME_ERROR (0x800001XX): 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).
13. Security and Privacy Considerations
13.1. Spoofed ACK Attack
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.
13.2. Slowloris Attacks
The attacks commonly known as Slowloris [SLOWLORIS] try to keep many
connections to the target endpoint open and hold them open as long as
possible. These attacks can be executed against a QUIC endpoint by
generating the minimum amount of activity necessary to avoid being
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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.
13.3. Stream Fragmentation and Reassembly Attacks
An adversarial endpoint might intentionally fragment the data on
stream buffers in order to cause disproportionate memory commitment.
An adversarial endpoint could open a stream and send some STREAM
frames containing arbitrary fragments of the stream content.
The attack is mitigated if flow control windows correspond to
available memory. However, some receivers will over-commit memory
and advertise flow control offsets in the aggregate that exceed
actual available memory. The over-commitment strategy can lead to
better performance when endpoints are well behaved, but renders
endpoints vulnerable to the stream fragmentation attack.
QUIC deployments SHOULD provide mitigations against the stream
fragmentation attack. Mitigations could consist of avoiding over-
committing memory, delaying reassembly of STREAM frames, implementing
heuristics based on the age and duration of reassembly holes, or some
combination.
13.4. Stream Commitment Attack
An adversarial endpoint can open lots of streams, exhausting state on
an endpoint. The adversarial endpoint could repeat the process on a
large number of connections, in a manner similar to SYN flooding
attacks in TCP.
Normally, clients will open streams sequentially, as explained in
Section 10.1. 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.
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The number of active streams is limited by the concurrent stream
limit transport parameter, as explained in Section 10.4. 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.
14. IANA Considerations
14.1. QUIC Transport Parameter Registry
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
[RFC5226]. Values with the first byte 0xff are reserved for Private
Use [RFC5226].
Registrations MUST include the following fields:
Value: The numeric value of the assignment (registrations will be
between 0x0000 and 0xfeff).
Parameter Name: A short mnemonic for the parameter.
Specification: A reference to a publicly available specification for
the value.
The nominated expert(s) verify that a specification exists and is
readily accessible. 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 Table 4.
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+--------+-------------------------+---------------+
| Value | Parameter Name | Specification |
+--------+-------------------------+---------------+
| 0x0000 | initial_max_stream_data | Section 7.3.1 |
| | | |
| 0x0001 | initial_max_data | Section 7.3.1 |
| | | |
| 0x0002 | initial_max_stream_id | Section 7.3.1 |
| | | |
| 0x0003 | idle_timeout | Section 7.3.1 |
| | | |
| 0x0004 | omit_connection_id | Section 7.3.1 |
| | | |
| 0x0005 | max_packet_size | Section 7.3.1 |
| | | |
| 0x0006 | stateless_reset_token | Section 7.3.1 |
+--------+-------------------------+---------------+
Table 4: Initial QUIC Transport Parameters Entries
15. References
15.1. Normative References
[I-D.ietf-tls-tls13]
Rescorla, E., "The Transport Layer Security (TLS) Protocol
Version 1.3", draft-ietf-tls-tls13-21 (work in progress),
July 2017.
[QUIC-RECOVERY]
Iyengar, J., Ed. and I. Swett, Ed., "QUIC Loss Detection
and Congestion Control", draft-ietf-quic-recovery (work in
progress), August 2017.
[QUIC-TLS]
Thomson, M., Ed. and S. Turner, Ed., "Using Transport
Layer Security (TLS) to Secure QUIC", draft-ietf-quic-tls
(work in progress), August 2017.
[RFC1191] Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191,
DOI 10.17487/RFC1191, November 1990,
<http://www.rfc-editor.org/info/rfc1191>.
[RFC1981] McCann, J., Deering, S., and J. Mogul, "Path MTU Discovery
for IP version 6", RFC 1981, DOI 10.17487/RFC1981, August
1996, <http://www.rfc-editor.org/info/rfc1981>.
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[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>.
[RFC3629] Yergeau, F., "UTF-8, a transformation format of ISO
10646", STD 63, RFC 3629, DOI 10.17487/RFC3629, November
2003, <http://www.rfc-editor.org/info/rfc3629>.
[RFC4086] Eastlake 3rd, D., Schiller, J., and S. Crocker,
"Randomness Requirements for Security", BCP 106, RFC 4086,
DOI 10.17487/RFC4086, June 2005,
<http://www.rfc-editor.org/info/rfc4086>.
[RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU
Discovery", RFC 4821, DOI 10.17487/RFC4821, March 2007,
<http://www.rfc-editor.org/info/rfc4821>.
[RFC5226] Narten, T. and H. Alvestrand, "Guidelines for Writing an
IANA Considerations Section in RFCs", RFC 5226,
DOI 10.17487/RFC5226, May 2008,
<http://www.rfc-editor.org/info/rfc5226>.
15.2. Informative References
[EARLY-DESIGN]
Roskind, J., "QUIC: Multiplexed Transport Over UDP",
December 2013, <https://goo.gl/dMVtFi>.
[RFC2104] Krawczyk, H., Bellare, M., and R. Canetti, "HMAC: Keyed-
Hashing for Message Authentication", RFC 2104,
DOI 10.17487/RFC2104, February 1997,
<http://www.rfc-editor.org/info/rfc2104>.
[RFC2360] Scott, G., "Guide for Internet Standards Writers", BCP 22,
RFC 2360, DOI 10.17487/RFC2360, June 1998,
<http://www.rfc-editor.org/info/rfc2360>.
[RFC5869] Krawczyk, H. and P. Eronen, "HMAC-based Extract-and-Expand
Key Derivation Function (HKDF)", RFC 5869,
DOI 10.17487/RFC5869, May 2010,
<http://www.rfc-editor.org/info/rfc5869>.
[RFC6824] Ford, A., Raiciu, C., Handley, M., and O. Bonaventure,
"TCP Extensions for Multipath Operation with Multiple
Addresses", RFC 6824, DOI 10.17487/RFC6824, January 2013,
<http://www.rfc-editor.org/info/rfc6824>.
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[RFC7301] Friedl, S., Popov, A., Langley, A., and E. Stephan,
"Transport Layer Security (TLS) Application-Layer Protocol
Negotiation Extension", RFC 7301, DOI 10.17487/RFC7301,
July 2014, <http://www.rfc-editor.org/info/rfc7301>.
[RFC7540] Belshe, M., Peon, R., and M. Thomson, Ed., "Hypertext
Transfer Protocol Version 2 (HTTP/2)", RFC 7540,
DOI 10.17487/RFC7540, May 2015,
<http://www.rfc-editor.org/info/rfc7540>.
[SLOWLORIS]
RSnake Hansen, R., "Welcome to Slowloris...", June 2009,
<https://web.archive.org/web/20150315054838/
http://ha.ckers.org/slowloris/>.
[SST] Ford, B., "Structured streams", ACM SIGCOMM Computer
Communication Review Vol. 37, pp. 361,
DOI 10.1145/1282427.1282421, October 2007.
15.3. URIs
[1] https://github.com/quicwg/base-drafts/wiki/QUIC-Versions
Appendix A. Contributors
The original authors of this specification were Ryan Hamilton, Jana
Iyengar, Ian Swett, and Alyssa Wilk.
The original design and rationale behind this protocol draw
significantly from work by Jim Roskind [EARLY-DESIGN]. In
alphabetical order, the contributors to the pre-IETF QUIC project at
Google are: Britt Cyr, Jeremy Dorfman, Ryan Hamilton, Jana Iyengar,
Fedor Kouranov, Charles Krasic, Jo Kulik, Adam Langley, Jim Roskind,
Robbie Shade, Satyam Shekhar, Cherie Shi, Ian Swett, Raman Tenneti,
Victor Vasiliev, Antonio Vicente, Patrik Westin, Alyssa Wilk, Dale
Worley, Fan Yang, Dan Zhang, Daniel Ziegler.
Appendix B. Acknowledgments
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.
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Appendix C. Change Log
*RFC Editor's Note:* Please remove this section prior to
publication of a final version of this document.
Issue and pull request numbers are listed with a leading octothorp.
C.1. Since draft-ietf-quic-transport-04
o Introduce STOP_SENDING frame, RST_STREAM only resets in one
direction (#165)
o Removed GOAWAY; application protocols are responsible for graceful
shutdown (#696)
o Reduced the number of error codes (#96, #177, #184, #211)
o Version validation fields can't move or change (#121)
o Removed versions from the transport parameters in a
NewSessionTicket message (#547)
o Clarify the meaning of "bytes in flight" (#550)
o Public reset is now stateless reset and not visible to the path
(#215)
o Reordered bits and fields in STREAM frame (#620)
o Clarifications to the stream state machine (#572, #571)
o Increased the maximum length of the Largest Acknowledged field in
ACK frames to 64 bits (#629)
o truncate_connection_id is renamed to omit_connection_id (#659)
o CONNECTION_CLOSE terminates the connection like TCP RST (#330,
#328)
o Update labels used in HKDF-Expand-Label to match TLS 1.3 (#642)
C.2. Since draft-ietf-quic-transport-03
o Change STREAM and RST_STREAM layout
o Add MAX_STREAM_ID settings
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C.3. Since draft-ietf-quic-transport-02
o The size of the initial packet payload has a fixed minimum (#267,
#472)
o Define when Version Negotiation packets are ignored (#284, #294,
#241, #143, #474)
o The 64-bit FNV-1a algorithm is used for integrity protection of
unprotected packets (#167, #480, #481, #517)
o Rework initial packet types to change how the connection ID is
chosen (#482, #442, #493)
o No timestamps are forbidden in unprotected packets (#542, #429)
o Cryptographic handshake is now on stream 0 (#456)
o Remove congestion control exemption for cryptographic handshake
(#248, #476)
o Version 1 of QUIC uses TLS; a new version is needed to use a
different handshake protocol (#516)
o STREAM frames have a reduced number of offset lengths (#543, #430)
o 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)
o A NEW_CONNECTION_ID frame supports connection migration without
linkability (#232, #491, #496)
o 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)
o Expanded security considerations (#440, #444, #445, #448)
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C.4. Since draft-ietf-quic-transport-01
o Defined short and long packet headers (#40, #148, #361)
o Defined a versioning scheme and stable fields (#51, #361)
o Define reserved version values for "greasing" negotiation (#112,
#278)
o The initial packet number is randomized (#35, #283)
o Narrow the packet number encoding range requirement (#67, #286,
#299, #323, #356)
o Defined client address validation (#52, #118, #120, #275)
o Define transport parameters as a TLS extension (#49, #122)
o SCUP and COPT parameters are no longer valid (#116, #117)
o Transport parameters for 0-RTT are either remembered from before,
or assume default values (#126)
o The server chooses connection IDs in its final flight (#119, #349,
#361)
o The server echoes the Connection ID and packet number fields when
sending a Version Negotiation packet (#133, #295, #244)
o Defined a minimum packet size for the initial handshake packet
from the client (#69, #136, #139, #164)
o Path MTU Discovery (#64, #106)
o The initial handshake packet from the client needs to fit in a
single packet (#338)
o Forbid acknowledgment of packets containing only ACK and PADDING
(#291)
o Require that frames are processed when packets are acknowledged
(#381, #341)
o Removed the STOP_WAITING frame (#66)
o Don't require retransmission of old timestamps for lost ACK frames
(#308)
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o Clarified that frames are not retransmitted, but the information
in them can be (#157, #298)
o Error handling definitions (#335)
o Split error codes into four sections (#74)
o Forbid the use of Public Reset where CONNECTION_CLOSE is possible
(#289)
o Define packet protection rules (#336)
o Require that stream be entirely delivered or reset, including
acknowledgment of all STREAM frames or the RST_STREAM, before it
closes (#381)
o Remove stream reservation from state machine (#174, #280)
o Only stream 1 does not contribute to connection-level flow control
(#204)
o Stream 1 counts towards the maximum concurrent stream limit (#201,
#282)
o Remove connection-level flow control exclusion for some streams
(except 1) (#246)
o RST_STREAM affects connection-level flow control (#162, #163)
o Flow control accounting uses the maximum data offset on each
stream, rather than bytes received (#378)
o Moved length-determining fields to the start of STREAM and ACK
(#168, #277)
o Added the ability to pad between frames (#158, #276)
o Remove error code and reason phrase from GOAWAY (#352, #355)
o GOAWAY includes a final stream number for both directions (#347)
o Error codes for RST_STREAM and CONNECTION_CLOSE are now at a
consistent offset (#249)
o Defined priority as the responsibility of the application protocol
(#104, #303)
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C.5. Since draft-ietf-quic-transport-00
o Replaced DIVERSIFICATION_NONCE flag with KEY_PHASE flag
o Defined versioning
o Reworked description of packet and frame layout
o Error code space is divided into regions for each component
o Use big endian for all numeric values
C.6. Since draft-hamilton-quic-transport-protocol-01
o Adopted as base for draft-ietf-quic-tls
o Updated authors/editors list
o Added IANA Considerations section
o Moved Contributors and Acknowledgments to appendices
Authors' Addresses
Jana Iyengar (editor)
Google
Email: jri@google.com
Martin Thomson (editor)
Mozilla
Email: martin.thomson@gmail.com
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