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Versions: (draft-hamilton-quic-transport-protocol) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14

QUIC                                                     J. Iyengar, Ed.
Internet-Draft                                                    Fastly
Intended status: Standards Track                         M. Thomson, Ed.
Expires: February 16, 2019                                       Mozilla
                                                         August 15, 2018


           QUIC: A UDP-Based Multiplexed and Secure Transport
                      draft-ietf-quic-transport-14

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 https://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, 2019.








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Copyright Notice

   Copyright (c) 2018 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (https://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   5
   2.  Conventions and Definitions . . . . . . . . . . . . . . . . .   6
     2.1.  Notational Conventions  . . . . . . . . . . . . . . . . .   7
   3.  Versions  . . . . . . . . . . . . . . . . . . . . . . . . . .   7
   4.  Packet Types and Formats  . . . . . . . . . . . . . . . . . .   8
     4.1.  Long Header . . . . . . . . . . . . . . . . . . . . . . .   8
     4.2.  Short Header  . . . . . . . . . . . . . . . . . . . . . .  11
     4.3.  Version Negotiation Packet  . . . . . . . . . . . . . . .  12
     4.4.  Retry Packet  . . . . . . . . . . . . . . . . . . . . . .  14
     4.5.  Cryptographic Handshake Packets . . . . . . . . . . . . .  16
     4.6.  Initial Packet  . . . . . . . . . . . . . . . . . . . . .  16
       4.6.1.  Connection IDs  . . . . . . . . . . . . . . . . . . .  18
       4.6.2.  Tokens  . . . . . . . . . . . . . . . . . . . . . . .  19
       4.6.3.  Starting Packet Numbers . . . . . . . . . . . . . . .  20
       4.6.4.  0-RTT Packet Numbers  . . . . . . . . . . . . . . . .  20
       4.6.5.  Minimum Packet Size . . . . . . . . . . . . . . . . .  20
     4.7.  Handshake Packet  . . . . . . . . . . . . . . . . . . . .  21
     4.8.  Protected Packets . . . . . . . . . . . . . . . . . . . .  21
     4.9.  Coalescing Packets  . . . . . . . . . . . . . . . . . . .  22
     4.10. Connection ID Encoding  . . . . . . . . . . . . . . . . .  23
     4.11. Packet Numbers  . . . . . . . . . . . . . . . . . . . . .  24
   5.  Frames and Frame Types  . . . . . . . . . . . . . . . . . . .  26
     5.1.  Extension Frames  . . . . . . . . . . . . . . . . . . . .  29
   6.  Life of a Connection  . . . . . . . . . . . . . . . . . . . .  29
     6.1.  Connection ID . . . . . . . . . . . . . . . . . . . . . .  30
     6.2.  Matching Packets to Connections . . . . . . . . . . . . .  31
       6.2.1.  Client Packet Handling  . . . . . . . . . . . . . . .  31
       6.2.2.  Server Packet Handling  . . . . . . . . . . . . . . .  32
     6.3.  Version Negotiation . . . . . . . . . . . . . . . . . . .  32
       6.3.1.  Sending Version Negotiation Packets . . . . . . . . .  33
       6.3.2.  Handling Version Negotiation Packets  . . . . . . . .  33



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       6.3.3.  Using Reserved Versions . . . . . . . . . . . . . . .  34
     6.4.  Cryptographic and Transport Handshake . . . . . . . . . .  34
     6.5.  Example Handshake Flows . . . . . . . . . . . . . . . . .  35
     6.6.  Transport Parameters  . . . . . . . . . . . . . . . . . .  37
       6.6.1.  Transport Parameter Definitions . . . . . . . . . . .  39
       6.6.2.  Values of Transport Parameters for 0-RTT  . . . . . .  41
       6.6.3.  New Transport Parameters  . . . . . . . . . . . . . .  42
       6.6.4.  Version Negotiation Validation  . . . . . . . . . . .  42
     6.7.  Stateless Retries . . . . . . . . . . . . . . . . . . . .  44
     6.8.  Using Explicit Congestion Notification  . . . . . . . . .  44
     6.9.  Proof of Source Address Ownership . . . . . . . . . . . .  46
       6.9.1.  Client Address Validation Procedure . . . . . . . . .  47
       6.9.2.  Address Validation for Future Connections . . . . . .  47
       6.9.3.  Address Validation Token Integrity  . . . . . . . . .  48
     6.10. Path Validation . . . . . . . . . . . . . . . . . . . . .  48
       6.10.1.  Initiation . . . . . . . . . . . . . . . . . . . . .  49
       6.10.2.  Response . . . . . . . . . . . . . . . . . . . . . .  49
       6.10.3.  Completion . . . . . . . . . . . . . . . . . . . . .  50
       6.10.4.  Abandonment  . . . . . . . . . . . . . . . . . . . .  50
     6.11. Connection Migration  . . . . . . . . . . . . . . . . . .  51
       6.11.1.  Probing a New Path . . . . . . . . . . . . . . . . .  51
       6.11.2.  Initiating Connection Migration  . . . . . . . . . .  52
       6.11.3.  Responding to Connection Migration . . . . . . . . .  52
       6.11.4.  Loss Detection and Congestion Control  . . . . . . .  54
       6.11.5.  Privacy Implications of Connection Migration . . . .  55
     6.12. Server's Preferred Address  . . . . . . . . . . . . . . .  55
       6.12.1.  Communicating A Preferred Address  . . . . . . . . .  56
       6.12.2.  Responding to Connection Migration . . . . . . . . .  56
       6.12.3.  Interaction of Client Migration and Preferred
                Address  . . . . . . . . . . . . . . . . . . . . . .  56
     6.13. Connection Termination  . . . . . . . . . . . . . . . . .  57
       6.13.1.  Closing and Draining Connection States . . . . . . .  57
       6.13.2.  Idle Timeout . . . . . . . . . . . . . . . . . . . .  58
       6.13.3.  Immediate Close  . . . . . . . . . . . . . . . . . .  59
       6.13.4.  Stateless Reset  . . . . . . . . . . . . . . . . . .  60
   7.  Frame Types and Formats . . . . . . . . . . . . . . . . . . .  64
     7.1.  Variable-Length Integer Encoding  . . . . . . . . . . . .  64
     7.2.  PADDING Frame . . . . . . . . . . . . . . . . . . . . . .  65
     7.3.  RST_STREAM Frame  . . . . . . . . . . . . . . . . . . . .  65
     7.4.  CONNECTION_CLOSE frame  . . . . . . . . . . . . . . . . .  66
     7.5.  APPLICATION_CLOSE frame . . . . . . . . . . . . . . . . .  67
     7.6.  MAX_DATA Frame  . . . . . . . . . . . . . . . . . . . . .  68
     7.7.  MAX_STREAM_DATA Frame . . . . . . . . . . . . . . . . . .  69
     7.8.  MAX_STREAM_ID Frame . . . . . . . . . . . . . . . . . . .  70
     7.9.  PING Frame  . . . . . . . . . . . . . . . . . . . . . . .  70
     7.10. BLOCKED Frame . . . . . . . . . . . . . . . . . . . . . .  71
     7.11. STREAM_BLOCKED Frame  . . . . . . . . . . . . . . . . . .  71
     7.12. STREAM_ID_BLOCKED Frame . . . . . . . . . . . . . . . . .  72



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     7.13. NEW_CONNECTION_ID Frame . . . . . . . . . . . . . . . . .  72
     7.14. STOP_SENDING Frame  . . . . . . . . . . . . . . . . . . .  74
     7.15. ACK Frame . . . . . . . . . . . . . . . . . . . . . . . .  74
       7.15.1.  ACK Block Section  . . . . . . . . . . . . . . . . .  76
       7.15.2.  Sending ACK Frames . . . . . . . . . . . . . . . . .  77
       7.15.3.  ACK Frames and Packet Protection . . . . . . . . . .  78
     7.16. ACK_ECN Frame . . . . . . . . . . . . . . . . . . . . . .  78
     7.17. PATH_CHALLENGE Frame  . . . . . . . . . . . . . . . . . .  79
     7.18. PATH_RESPONSE Frame . . . . . . . . . . . . . . . . . . .  80
     7.19. NEW_TOKEN frame . . . . . . . . . . . . . . . . . . . . .  80
     7.20. STREAM Frames . . . . . . . . . . . . . . . . . . . . . .  80
     7.21. CRYPTO Frame  . . . . . . . . . . . . . . . . . . . . . .  82
   8.  Packetization and Reliability . . . . . . . . . . . . . . . .  83
     8.1.  Packet Processing and Acknowledgment  . . . . . . . . . .  84
     8.2.  Retransmission of Information . . . . . . . . . . . . . .  84
     8.3.  Packet Size . . . . . . . . . . . . . . . . . . . . . . .  86
     8.4.  Path Maximum Transmission Unit  . . . . . . . . . . . . .  87
       8.4.1.  IPv4 PMTU Discovery . . . . . . . . . . . . . . . . .  87
       8.4.2.  Special Considerations for Packetization Layer PMTU
               Discovery . . . . . . . . . . . . . . . . . . . . . .  88
   9.  Streams: QUIC's Data Structuring Abstraction  . . . . . . . .  88
     9.1.  Stream Identifiers  . . . . . . . . . . . . . . . . . . .  89
     9.2.  Stream States . . . . . . . . . . . . . . . . . . . . . .  90
       9.2.1.  Send Stream States  . . . . . . . . . . . . . . . . .  91
       9.2.2.  Receive Stream States . . . . . . . . . . . . . . . .  93
       9.2.3.  Permitted Frame Types . . . . . . . . . . . . . . . .  96
       9.2.4.  Bidirectional Stream States . . . . . . . . . . . . .  96
     9.3.  Solicited State Transitions . . . . . . . . . . . . . . .  97
     9.4.  Stream Concurrency  . . . . . . . . . . . . . . . . . . .  98
     9.5.  Sending and Receiving Data  . . . . . . . . . . . . . . .  99
     9.6.  Stream Prioritization . . . . . . . . . . . . . . . . . .  99
   10. Flow Control  . . . . . . . . . . . . . . . . . . . . . . . . 100
     10.1.  Edge Cases and Other Considerations  . . . . . . . . . . 101
       10.1.1.  Response to a RST_STREAM . . . . . . . . . . . . . . 102
       10.1.2.  Data Limit Increments  . . . . . . . . . . . . . . . 102
     10.2.  Stream Limit Increment . . . . . . . . . . . . . . . . . 103
       10.2.1.  Blocking on Flow Control . . . . . . . . . . . . . . 103
     10.3.  Stream Final Offset  . . . . . . . . . . . . . . . . . . 103
     10.4.  Flow Control for Cryptographic Handshake . . . . . . . . 104
   11. Error Handling  . . . . . . . . . . . . . . . . . . . . . . . 104
     11.1.  Connection Errors  . . . . . . . . . . . . . . . . . . . 104
     11.2.  Stream Errors  . . . . . . . . . . . . . . . . . . . . . 105
     11.3.  Transport Error Codes  . . . . . . . . . . . . . . . . . 105
     11.4.  Application Protocol Error Codes . . . . . . . . . . . . 107
   12. Security Considerations . . . . . . . . . . . . . . . . . . . 107
     12.1.  Handshake Denial of Service  . . . . . . . . . . . . . . 107
     12.2.  Spoofed ACK Attack . . . . . . . . . . . . . . . . . . . 108
     12.3.  Optimistic ACK Attack  . . . . . . . . . . . . . . . . . 109



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     12.4.  Slowloris Attacks  . . . . . . . . . . . . . . . . . . . 109
     12.5.  Stream Fragmentation and Reassembly Attacks  . . . . . . 109
     12.6.  Stream Commitment Attack . . . . . . . . . . . . . . . . 110
     12.7.  Explicit Congestion Notification Attacks . . . . . . . . 110
     12.8.  Stateless Reset Oracle . . . . . . . . . . . . . . . . . 110
   13. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 111
     13.1.  QUIC Transport Parameter Registry  . . . . . . . . . . . 111
     13.2.  QUIC Frame Type Registry . . . . . . . . . . . . . . . . 112
     13.3.  QUIC Transport Error Codes Registry  . . . . . . . . . . 113
   14. References  . . . . . . . . . . . . . . . . . . . . . . . . . 115
     14.1.  Normative References . . . . . . . . . . . . . . . . . . 115
     14.2.  Informative References . . . . . . . . . . . . . . . . . 116
   Appendix A.  Sample Packet Number Decoding Algorithm  . . . . . . 117
   Appendix B.  Change Log . . . . . . . . . . . . . . . . . . . . . 118
     B.1.  Since draft-ietf-quic-transport-13  . . . . . . . . . . . 118
     B.2.  Since draft-ietf-quic-transport-12  . . . . . . . . . . . 119
     B.3.  Since draft-ietf-quic-transport-11  . . . . . . . . . . . 120
     B.4.  Since draft-ietf-quic-transport-10  . . . . . . . . . . . 120
     B.5.  Since draft-ietf-quic-transport-09  . . . . . . . . . . . 121
     B.6.  Since draft-ietf-quic-transport-08  . . . . . . . . . . . 121
     B.7.  Since draft-ietf-quic-transport-07  . . . . . . . . . . . 122
     B.8.  Since draft-ietf-quic-transport-06  . . . . . . . . . . . 123
     B.9.  Since draft-ietf-quic-transport-05  . . . . . . . . . . . 123
     B.10. Since draft-ietf-quic-transport-04  . . . . . . . . . . . 124
     B.11. Since draft-ietf-quic-transport-03  . . . . . . . . . . . 124
     B.12. Since draft-ietf-quic-transport-02  . . . . . . . . . . . 125
     B.13. Since draft-ietf-quic-transport-01  . . . . . . . . . . . 126
     B.14. Since draft-ietf-quic-transport-00  . . . . . . . . . . . 128
     B.15. Since draft-hamilton-quic-transport-protocol-01 . . . . . 128
   Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . . 128
   Contributors  . . . . . . . . . . . . . . . . . . . . . . . . . . 128
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . . 129

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 secure transport for multiple
   applications.

   o  Version negotiation

   o  Low-latency connection establishment

   o  Authenticated and encrypted header and payload

   o  Stream multiplexing




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   o  Stream and connection-level flow control

   o  Connection migration and resilience to NAT rebinding

   QUIC implements techniques learned from experience with TCP, SCTP and
   other transport protocols.  QUIC uses UDP as substrate so as to not
   require changes to legacy client operating systems and middleboxes to
   be deployable.  QUIC authenticates all of its headers and encrypts
   most of the data it exchanges, including its signaling.  This allows
   the protocol to evolve without incurring a dependency on upgrades to
   middleboxes.  This document describes the core QUIC protocol,
   including the conceptual design, wire format, and mechanisms of the
   QUIC protocol for connection establishment, stream multiplexing,
   stream and connection-level flow control, connection migration, and
   data reliability.

   Accompanying documents describe QUIC's loss detection and congestion
   control [QUIC-RECOVERY], and the use of TLS 1.3 for key negotiation
   [QUIC-TLS].

   QUIC version 1 conforms to the protocol invariants in
   [QUIC-INVARIANTS].

2.  Conventions and Definitions

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
   "OPTIONAL" in this document are to be interpreted as described in BCP
   14 [RFC2119] [RFC8174] when, and only when, they appear in all
   capitals, as shown here.

   Definitions of terms that are used in this document:

   Client:  The endpoint initiating a QUIC connection.

   Server:  The endpoint accepting incoming QUIC connections.

   Endpoint:  The client or server end of a connection.

   Stream:  A logical, bi-directional channel of ordered bytes within a
      QUIC connection.

   Connection:  A conversation between two QUIC endpoints with a single
      encryption context that multiplexes streams within it.

   Connection ID:  An opaque identifier that is used to identify a QUIC
      connection at an endpoint.  Each endpoint sets a value that its
      peer includes in packets.



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   QUIC packet:  A well-formed UDP payload that can be parsed by a QUIC
      receiver.

   QUIC is a name, not an acronym.

2.1.  Notational Conventions

   Packet and frame diagrams use the format described in Section 3.1 of
   [RFC2360], with the following additional conventions:

   [x]  Indicates that x is optional

   x (A)  Indicates that x is A bits long

   x (A/B/C) ...  Indicates that x is one of A, B, or C bits long

   x (i) ...  Indicates that x uses the variable-length encoding in
      Section 7.1

   x (*) ...  Indicates that x is variable-length

3.  Versions

   QUIC versions are identified using a 32-bit unsigned number.

   The version 0x00000000 is reserved to represent version negotiation.
   This version of the specification is identified by the number
   0x00000001.

   Other versions of QUIC might have different properties to this
   version.  The properties of QUIC that are guaranteed to be consistent
   across all versions of the protocol are described in
   [QUIC-INVARIANTS].

   Version 0x00000001 of QUIC uses TLS as a cryptographic handshake
   protocol, as described in [QUIC-TLS].

   Versions with the most significant 16 bits of the version number
   cleared are reserved for use in future IETF consensus documents.

   Versions that follow the pattern 0x?a?a?a?a are reserved for use in
   forcing version negotiation to be exercised.  That is, any version
   number where the low four bits of all octets is 1010 (in binary).  A
   client or server MAY advertise support for any of these reserved
   versions.

   Reserved version numbers will probably never represent a real
   protocol; a client MAY use one of these version numbers with the



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   expectation that the server will initiate version negotiation; a
   server MAY advertise support for one of these versions and can expect
   that clients ignore the value.

   [[RFC editor: please remove the remainder of this section before
   publication.]]

   The version number for the final version of this specification
   (0x00000001), is reserved for the version of the protocol that is
   published as an RFC.

   Version numbers used to identify IETF drafts are created by adding
   the draft number to 0xff000000.  For example, draft-ietf-quic-
   transport-13 would be identified as 0xff00000D.

   Implementors are encouraged to register version numbers of QUIC that
   they are using for private experimentation on the GitHub wiki at
   <https://github.com/quicwg/base-drafts/wiki/QUIC-Versions>.

4.  Packet Types and Formats

   We first describe QUIC's packet types and their formats, since some
   are referenced in subsequent mechanisms.

   All numeric values are encoded in network byte order (that is, big-
   endian) and all field sizes are in bits.  When discussing individual
   bits of fields, the least significant bit is referred to as bit 0.
   Hexadecimal notation is used for describing the value of fields.

   Any QUIC packet has either a long or a short header, as indicated by
   the Header Form bit.  Long headers are expected to be used early in
   the connection before version negotiation and establishment of 1-RTT
   keys.  Short headers are minimal version-specific headers, which are
   used after version negotiation and 1-RTT keys are established.

4.1.  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)  |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         Version (32)                          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |DCIL(4)|SCIL(4)|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |               Destination Connection ID (0/32..144)         ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                 Source Connection ID (0/32..144)            ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                           Length (i)                        ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                     Packet Number (8/16/32)                   |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                          Payload (*)                        ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                    Figure 1: Long Header Packet Format

   Long headers are used for packets that are sent prior to the
   completion of version negotiation and establishment of 1-RTT keys.
   Once both conditions are met, a sender switches to sending packets
   using the short header (Section 4.2).  The long form allows for
   special packets - such as the Version Negotiation packet - to be
   represented in this uniform fixed-length packet format.  Packets that
   use the long header contain the following fields:

   Header Form:  The most significant bit (0x80) of octet 0 (the first
      octet) is set to 1 for long headers.

   Long Packet Type:  The remaining seven bits of octet 0 contain the
      packet type.  This field can indicate one of 128 packet types.
      The types specified for this version are listed in Table 1.

   Version:  The QUIC Version is a 32-bit field that follows the Type.
      This field indicates which version of QUIC is in use and
      determines how the rest of the protocol fields are interpreted.

   DCIL and SCIL:  The octet following the version contains the lengths
      of the two connection ID fields that follow it.  These lengths are
      encoded as two 4-bit unsigned integers.  The Destination
      Connection ID Length (DCIL) field occupies the 4 high bits of the
      octet and the Source Connection ID Length (SCIL) field occupies
      the 4 low bits of the octet.  An encoded length of 0 indicates
      that the connection ID is also 0 octets in length.  Non-zero



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      encoded lengths are increased by 3 to get the full length of the
      connection ID, producing a length between 4 and 18 octets
      inclusive.  For example, an octet with the value 0x50 describes an
      8-octet Destination Connection ID and a zero-length Source
      Connection ID.

   Destination Connection ID:  The Destination Connection ID field
      follows the connection ID lengths and is either 0 octets in length
      or between 4 and 18 octets.  Section 4.10 describes the use of
      this field in more detail.

   Source Connection ID:  The Source Connection ID field follows the
      Destination Connection ID and is either 0 octets in length or
      between 4 and 18 octets.  Section 4.10 describes the use of this
      field in more detail.

   Length:  The length of the remainder of the packet (that is, the
      Packet Number and Payload fields) in octets, encoded as a
      variable-length integer (Section 7.1).

   Packet Number:  The packet number field is 1, 2, or 4 octets long.
      The packet number has confidentiality protection separate from
      packet protection, as described in Section 5.3 of [QUIC-TLS].  The
      length of the packet number field is encoded in the plaintext
      packet number.  See Section 4.11 for details.

   Payload:  The payload of the packet.

   The following packet types are defined:

                 +------+-----------------+-------------+
                 | Type | Name            | Section     |
                 +------+-----------------+-------------+
                 | 0x7F | Initial         | Section 4.6 |
                 |      |                 |             |
                 | 0x7E | Retry           | Section 4.4 |
                 |      |                 |             |
                 | 0x7D | Handshake       | Section 4.7 |
                 |      |                 |             |
                 | 0x7C | 0-RTT Protected | Section 4.8 |
                 +------+-----------------+-------------+

                     Table 1: Long Header Packet Types

   The header form, type, connection ID lengths octet, destination and
   source connection IDs, and version fields of a long header packet are
   version-independent.  The packet number and values for packet types
   defined in Table 1 are version-specific.  See [QUIC-INVARIANTS] for



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   details on how packets from different versions of QUIC are
   interpreted.

   The interpretation of the fields and the payload are specific to a
   version and packet type.  Type-specific semantics for this version
   are described in the following sections.

   The end of the packet is determined by the Length field.  The Length
   field covers the both the Packet Number and Payload fields, both of
   which are confidentiality protected and initially of unknown length.
   The size of the Payload field is learned once the packet number
   protection is removed.

   Senders can sometimes coalesce multiple packets into one UDP
   datagram.  See Section 4.9 for more details.

4.2.  Short Header

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+
   |0|K|1|1|0|R R R|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                Destination Connection ID (0..144)           ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                      Packet Number (8/16/32)                ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                     Protected Payload (*)                   ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                   Figure 2: Short Header Packet Format

   The short header can be used after the version and 1-RTT keys are
   negotiated.  Packets that use the short header contain the following
   fields:

   Header Form:  The most significant bit (0x80) of octet 0 is set to 0
      for the short header.

   Key Phase Bit:  The second bit (0x40) of octet 0 indicates the key
      phase, which allows a recipient of a packet to identify the packet
      protection keys that are used to protect the packet.  See
      [QUIC-TLS] for details.

   [[Editor's Note: this section should be removed and the bit
   definitions changed before this draft goes to the IESG.]]

   Third Bit:  The third bit (0x20) of octet 0 is set to 1.



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   [[Editor's Note: this section should be removed and the bit
   definitions changed before this draft goes to the IESG.]]

   Fourth Bit:  The fourth bit (0x10) of octet 0 is set to 1.

   [[Editor's Note: this section should be removed and the bit
   definitions changed before this draft goes to the IESG.]]

   Google QUIC Demultipexing Bit:  The fifth bit (0x8) of octet 0 is set
      to 0.  This allows implementations of Google QUIC to distinguish
      Google QUIC packets from short header packets sent by a client
      because Google QUIC servers expect the connection ID to always be
      present.  The special interpretation of this bit SHOULD be removed
      from this specification when Google QUIC has finished
      transitioning to the new header format.

   Reserved:  The sixth, seventh, and eighth bits (0x7) of octet 0 are
      reserved for experimentation.  Endpoints MUST ignore these bits on
      packets they receive unless they are participating in an
      experiment that uses these bits.  An endpoint not actively using
      these bits SHOULD set the value randomly on packets they send to
      protect against unwanted inference about particular values.

   Destination Connection ID:  The Destination Connection ID is a
      connection ID that is chosen by the intended recipient of the
      packet.  See Section 6.1 for more details.

   Packet Number:  The packet number field is 1, 2, or 4 octets long.
      The packet number has confidentiality protection separate from
      packet protection, as described in Section 5.3 of [QUIC-TLS].  The
      length of the packet number field is encoded in the plaintext
      packet number.  See Section 4.11 for details.

   Protected Payload:  Packets with a short header always include a
      1-RTT protected payload.

   The header form and connection ID field of a short header packet are
   version-independent.  The remaining fields are specific to the
   selected QUIC version.  See [QUIC-INVARIANTS] for details on how
   packets from different versions of QUIC are interpreted.

4.3.  Version Negotiation Packet

   A Version Negotiation packet is inherently not version-specific, and
   does not use the long packet header (see Section 4.1.  Upon receipt
   by a client, it will appear to be a packet using the long header, but
   will be identified as a Version Negotiation packet based on the
   Version field having a value of 0.



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   The Version Negotiation packet is a response to a client packet that
   contains a version that is not supported by the server, and is only
   sent by servers.

   The layout of a Version Negotiation packet is:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+
   |1|  Unused (7) |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                          Version (32)                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |DCIL(4)|SCIL(4)|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |               Destination Connection ID (0/32..144)         ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                 Source Connection ID (0/32..144)            ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                    Supported Version 1 (32)                 ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                   [Supported Version 2 (32)]                ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
                                  ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                   [Supported Version N (32)]                ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                   Figure 3: Version Negotiation Packet

   The value in the Unused field is selected randomly by the server.

   The Version field of a Version Negotiation packet MUST be set to
   0x00000000.

   The server MUST include the value from the Source Connection ID field
   of the packet it receives in the Destination Connection ID field.
   The value for Source Connection ID MUST be copied from the
   Destination Connection ID of the received packet, which is initially
   randomly selected by a client.  Echoing both connection IDs gives
   clients some assurance that the server received the packet and that
   the Version Negotiation packet was not generated by an off-path
   attacker.

   The remainder of the Version Negotiation packet is a list of 32-bit
   versions which the server supports.





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   A Version Negotiation packet cannot be explicitly acknowledged in an
   ACK frame by a client.  Receiving another Initial packet implicitly
   acknowledges a Version Negotiation packet.

   The Version Negotiation packet does not include the Packet Number and
   Length fields present in other packets that use the long header form.
   Consequently, a Version Negotiation packet consumes an entire UDP
   datagram.

   See Section 6.3 for a description of the version negotiation process.

4.4.  Retry Packet

   A Retry packet uses a long packet header with a type value of 0x7E.
   It carries an address validation token created by the server.  It is
   used by a server that wishes to perform a stateless retry (see
   Section 6.7).

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+
   |1|    0x7e     |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         Version (32)                          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |DCIL(4)|SCIL(4)|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |               Destination Connection ID (0/32..144)         ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                 Source Connection ID (0/32..144)            ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |    ODCIL(8)   |      Original Destination Connection ID (*)   |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        Retry Token (*)                      ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                          Figure 4: Retry Packet

   A Retry packet (shown in Figure 4) only uses the invariant portion of
   the long packet header [QUIC-INVARIANTS]; that is, the fields up to
   and including the Destination and Source Connection ID fields.  The
   contents of the Retry packet are not protected.  Like Version
   Negotiation, a Retry packet contains the long header including the
   connection IDs, but omits the Length, Packet Number, and Payload
   fields.  These are replaced with:

   ODCIL:  The length of the Original Destination Connection ID field.
      The length is encoded in the least significant 4 bits of the



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      octet, using the same encoding as the DCIL and SCIL fields.  The
      most significant 4 bits of this octet are reserved.  Unless a use
      for these bits has been negotiated, endpoints SHOULD send
      randomized values and MUST ignore any value that it receives.

   Original Destination Connection ID:  The Original Destination
      Connection ID contains the value of the Destination Connection ID
      from the Initial packet that this Retry is in response to.  The
      length of this field is given in ODCIL.

   Retry Token:  An opaque token that the server can use to validate the
      client's address.

   The server populates the Destination Connection ID with the
   connection ID that the client included in the Source Connection ID of
   the Initial packet.

   The server includes a connection ID of its choice in the Source
   Connection ID field.  The client MUST use this connection ID in the
   Destination Connection ID of subsequent packets that it sends.

   A Retry packet does not include a packet number and cannot be
   explicitly acknowledged by a client.

   A server MUST NOT send a Retry in response to packets other than
   Initial or 0-RTT packets.  A server MAY choose to only send Retry in
   response to Initial packets and discard or buffer 0-RTT packets
   corresponding to unvalidated client addresses.

   If the Original Destination Connection ID field does not match the
   Destination Connection ID from the most recent Initial packet it
   sent, clients MUST discard the packet.  This prevents an off-path
   attacker from injecting a Retry packet.

   The client responds to a Retry packet with an Initial packet that
   includes the provided Retry Token to continue connection
   establishment.

   A client MAY attempt 0-RTT after receiving a Retry packet by sending
   0-RTT packets to the connection ID provided by the server.  A client
   that sends additional 0-RTT packets MUST NOT reset the packet number
   to 0 after a Retry packet, see Section 4.6.4.

   A server that might send another Retry packet in response to a
   subsequent Initial packet MUST set the Source Connection ID to a new
   value of at least 8 octets in length.  This allows clients to
   distinguish between Retry packets when the server sends multiple
   rounds of Retry packets.  Consequently, a valid Retry packet will



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   always have an Original Destination Connection ID that is at least 8
   octets long; clients MUST discard Retry packets that include a
   shorter value.  A server that will not send additional Retry packets
   can set the Source Connection ID to any value.

4.5.  Cryptographic Handshake Packets

   Once version negotiation is complete, the cryptographic handshake is
   used to agree on cryptographic keys.  The cryptographic handshake is
   carried in Initial (Section 4.6) and Handshake (Section 4.7) packets.

   All these packets use the long header and contain the current QUIC
   version in the version field.

   In order to prevent tampering by version-unaware middleboxes, Initial
   packets are protected with connection- and version-specific keys
   (Initial keys) as described in [QUIC-TLS].  This protection does not
   provide confidentiality or integrity against on-path attackers, but
   provides some level of protection against off-path attackers.

4.6.  Initial Packet

   The Initial packet uses long headers with a type value of 0x7F.  It
   carries the first CRYPTO frames sent by the client and server to
   perform key exchange, and carries ACKs in either direction.  The
   Initial packet is protected by Initial keys as described in
   [QUIC-TLS].

   The Initial packet (shown in Figure 5) has two additional header
   fields that are added to the Long Header before the Length field.





















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   +-+-+-+-+-+-+-+-+
   |1|    0x7f     |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         Version (32)                          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |DCIL(4)|SCIL(4)|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |               Destination Connection ID (0/32..144)         ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                 Source Connection ID (0/32..144)            ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         Token Length (i)                    ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                            Token (*)                        ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                           Length (i)                        ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                     Packet Number (8/16/32)                   |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                          Payload (*)                        ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                         Figure 5: Initial Packet

   These fields include the token that was previously provided in a
   Retry packet or NEW_TOKEN frame:

   Token Length:  A variable-length integer specifying the length of the
      Token field, in bytes.  This value is zero if no token is present.
      Initial packets sent by the server MUST set the Token Length field
      to zero; clients that receive an Initial packet with a non-zero
      Token Length field MUST either discard the packet or generate a
      connection error of type PROTOCOL_VIOLATION.

   Token:  The value of the token.

   The client and server use the Initial packet type for any packet that
   contains an initial cryptographic handshake message.  This includes
   all cases where a new packet containing the initial cryptographic
   message needs to be created, such as the packets sent after receiving
   a Version Negotiation (Section 4.3) or Retry packet (Section 4.4).

   A server sends its first Initial packet in response to a client
   Initial.  A server may send multiple Initial packets.  The
   cryptographic key exchange could require multiple round trips or
   retransmissions of this data.





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   The payload of an Initial packet includes a CRYPTO frame (or frames)
   containing a cryptographic handshake message, ACK frames, or both.
   PADDING frames are also permitted.  The first CRYPTO frame sent
   always begins at an offset of 0 (see Section 6.4).

   The first packet sent by a client always includes a CRYPTO frame that
   contains the entirety of the first cryptographic handshake message.
   This packet, and the cryptographic handshake message, MUST fit in a
   single UDP datagram (see Section 6.4).

   Note that if the server sends a HelloRetryRequest, the client will
   send a second Initial packet.  This Initial packet will continue the
   cryptographic handshake and will contain a CRYPTO frame with an
   offset matching the size of the CRYPTO frame sent in the first
   Initial packet.  Cryptographic handshake messages subsequent to the
   first do not need to fit within a single UDP datagram.

4.6.1.  Connection IDs

   When an Initial packet is sent by a client which has not previously
   received a Retry packet from the server, it populates the Destination
   Connection ID field with an unpredictable value.  This MUST be at
   least 8 octets in length.  Until a packet is received from the
   server, the client MUST use the same value unless it abandons the
   connection attempt and starts a new one.  The initial Destination
   Connection ID is used to determine packet protection keys for Initial
   packets.

   The client populates the Source Connection ID field with a value of
   its choosing and sets the SCIL field to match.

   The Destination Connection ID field in the server's Initial packet
   contains a connection ID that is chosen by the recipient of the
   packet (i.e., the client); the Source Connection ID includes the
   connection ID that the sender of the packet wishes to use (see
   Section 6.1).  The server MUST use consistent Source Connection IDs
   during the handshake.

   On first receiving an Initial or Retry packet from the server, the
   client uses the Source Connection ID supplied by the server as the
   Destination Connection ID for subsequent packets.  Once a client has
   received an Initial packet from the server, it MUST discard any
   packet it receives with a different Source Connection ID.








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4.6.2.  Tokens

   If the client has a token received in a NEW_TOKEN frame on a previous
   connection to what it believes to be the same server, it can include
   that value in the Token field of its Initial packet.

   A token allows a server to correlate activity between connections.
   Specifically, the connection where the token was issued, and any
   connection where it is used.  Clients that want to break continuity
   of identity with a server MAY discard tokens provided using the
   NEW_TOKEN frame.  Tokens obtained in Retry packets MUST NOT be
   discarded.

   A client SHOULD NOT reuse a token.  Reusing a token allows
   connections to be linked by entities on the network path (see
   Section 6.11.5).  A client MUST NOT reuse a token if it believes that
   its point of network attachment has changed since the token was last
   used; that is, if there is a change in its local IP address or
   network interface.  A client needs to start the connection process
   over if it migrates prior to completing the handshake.

   If the client received a Retry packet from the server and sends an
   Initial packet in response, then it sets the Destination Connection
   ID to the value from the Source Connection ID in the Retry packet.
   Changing Destination Connection ID also results in a change to the
   keys used to protect the Initial packet.  It also sets the Token
   field to the token provided in the Retry.  The client MUST NOT change
   the Source Connection ID because the server could include the
   connection ID as part of its token validation logic.

   When a server receives an Initial packet with an address validation
   token, it SHOULD attempt to validate it.  If the token is invalid
   then the server SHOULD proceed as if the client did not have a
   validated address, including potentially sending a Retry.  If the
   validation succeeds, the server SHOULD then allow the handshake to
   proceed (see Section 6.7).

   Note:  The rationale for treating the client as unvalidated rather
      than discarding the packet is that the client might have received
      the token in a previous connection using the NEW_TOKEN frame, and
      if the server has lost state, it might be unable to validate the
      token at all, leading to connection failure if the packet is
      discarded.  A server MAY encode tokens provided with NEW_TOKEN
      frames and Retry packets differently, and validate the latter more
      strictly.






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4.6.3.  Starting Packet Numbers

   The first Initial packet sent by either endpoint contains a packet
   number of 0.  The packet number MUST increase monotonically
   thereafter.  Initial packets are in a different packet number space
   to other packets (see Section 4.11).

4.6.4.  0-RTT Packet Numbers

   Packet numbers for 0-RTT protected packets use the same space as
   1-RTT protected packets.

   After a client receives a Retry or Version Negotiation packet, 0-RTT
   packets are likely to have been lost or discarded by the server.  A
   client MAY attempt to resend data in 0-RTT packets after it sends a
   new Initial packet.

   A client MUST NOT reset the packet number it uses for 0-RTT packets.
   The keys used to protect 0-RTT packets will not change as a result of
   responding to a Retry or Version Negotiation packet unless the client
   also regenerates the cryptographic handshake message.  Sending
   packets with the same packet number in that case is likely to
   compromise the packet protection for all 0-RTT packets because the
   same key and nonce could be used to protect different content.

   Receiving a Retry or Version Negotiation packet, especially a Retry
   that changes the connection ID used for subsequent packets, indicates
   a strong possibility that 0-RTT packets could be lost.  A client only
   receives acknowledgments for its 0-RTT packets once the handshake is
   complete.  Consequently, a server might expect 0-RTT packets to start
   with a packet number of 0.  Therefore, in determining the length of
   the packet number encoding for 0-RTT packets, a client MUST assume
   that all packets up to the current packet number are in flight,
   starting from a packet number of 0.  Thus, 0-RTT packets could need
   to use a longer packet number encoding.

   A client MAY instead generate a fresh cryptographic handshake message
   and start packet numbers from 0.  This ensures that new 0-RTT packets
   will not use the same keys, avoiding any risk of key and nonce reuse;
   this also prevents 0-RTT packets from previous handshake attempts
   from being accepted as part of the connection.

4.6.5.  Minimum Packet Size

   The payload of a UDP datagram carrying the Initial packet MUST be
   expanded to at least 1200 octets (see Section 8), by adding PADDING
   frames to the Initial packet and/or by combining the Initial packet
   with a 0-RTT packet (see Section 4.9).



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4.7.  Handshake Packet

   A Handshake packet uses long headers with a type value of 0x7D.  It
   is used to carry acknowledgments and cryptographic handshake messages
   from the server and client.

   A server sends its cryptographic handshake in one or more Handshake
   packets in response to an Initial packet if it does not send a Retry
   packet.  Once a client has received a Handshake packet from a server,
   it uses Handshake packets to send subsequent cryptographic handshake
   messages and acknowledgments to the server.

   The Destination Connection ID field in a Handshake packet contains a
   connection ID that is chosen by the recipient of the packet; the
   Source Connection ID includes the connection ID that the sender of
   the packet wishes to use (see Section 4.10).

   The first Handshake packet sent by a server contains a packet number
   of 0.  Handshake packets are their own packet number space.  Packet
   numbers are incremented normally for other Handshake packets.

   Servers MUST NOT send more than three datagrams including Initial and
   Handshake packets without receiving a packet from a verified source
   address.  Source addresses can be verified through an address
   validation token (delivered via a Retry packet or a NEW_TOKEN frame)
   or by receiving any message from the client encrypted using the
   Handshake keys.

   The payload of this packet contains CRYPTO frames and could contain
   PADDING, or ACK frames.  Handshake packets MAY contain
   CONNECTION_CLOSE frames if the handshake is unsuccessful.

4.8.  Protected Packets

   All QUIC packets use packet protection.  Packets that are protected
   with the static handshake keys or the 0-RTT keys are sent with long
   headers; all packets protected with 1-RTT keys are sent with short
   headers.  The different packet types explicitly indicate the
   encryption level and therefore the keys that are used to remove
   packet protection.  0-RTT and 1-RTT protected packets share a single
   packet number space.

   Packets protected with handshake keys only use packet protection to
   ensure that the sender of the packet is on the network path.  This
   packet protection is not effective confidentiality protection; any
   entity that receives the Initial packet from a client can recover the
   keys necessary to remove packet protection or to generate packets
   that will be successfully authenticated.



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   Packets protected with 0-RTT and 1-RTT keys are expected to have
   confidentiality and data origin authentication; the cryptographic
   handshake ensures that only the communicating endpoints receive the
   corresponding keys.

   Packets protected with 0-RTT keys use a type value of 0x7C.  The
   connection ID fields for a 0-RTT packet MUST match the values used in
   the Initial packet (Section 4.6).

   The client can send 0-RTT packets after receiving an Initial
   Section 4.6 or Handshake (Section 4.7) packet, if that packet does
   not complete the handshake.  Even if the client receives a different
   connection ID in the Handshake packet, it MUST continue to use the
   same Destination Connection ID for 0-RTT packets, see Section 4.10.

   The version field for protected packets is the current QUIC version.

   The packet number field contains a packet number, which has
   additional confidentiality protection that is applied after packet
   protection is applied (see [QUIC-TLS] for details).  The underlying
   packet number increases with each packet sent, see Section 4.11 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 5.

4.9.  Coalescing Packets

   A sender can coalesce multiple QUIC packets (typically a
   Cryptographic Handshake packet and a Protected packet) into one UDP
   datagram.  This can reduce the number of UDP datagrams needed to send
   application data during the handshake and immediately afterwards.  It
   is not necessary for senders to coalesce packets, though failing to
   do so will require sending a significantly larger number of datagrams
   during the handshake.  Receivers MUST be able to process coalesced
   packets.

   Senders SHOULD coalesce packets in order of increasing encryption
   levels (Initial, Handshake, 0-RTT, 1-RTT), as this makes it more
   likely the receiver will be able to process all the packets in a
   single pass.  A packet with a short header does not include a length,
   so it will always be the last packet included in a UDP datagram.

   Senders MUST NOT coalesce QUIC packets with different Destination
   Connection IDs into a single UDP datagram.  Receivers SHOULD ignore




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   any subsequent packets with a different Destination Connection ID
   than the first packet in the datagram.

   Every QUIC packet that is coalesced into a single UDP datagram is
   separate and complete.  Though the values of some fields in the
   packet header might be redundant, no fields are omitted.  The
   receiver of coalesced QUIC packets MUST individually process each
   QUIC packet and separately acknowledge them, as if they were received
   as the payload of different UDP datagrams.  If one or more packets in
   a datagram cannot be processed yet (because the keys are not yet
   available) or processing fails (decryption failure, unknown type,
   etc.), the receiver MUST still attempt to process the remaining
   packets.  The skipped packets MAY either be discarded or buffered for
   later processing, just as if the packets were received out-of-order
   in separate datagrams.

   Retry (Section 4.4) and Version Negotiation (Section 4.3) packets
   cannot be coalesced.

4.10.  Connection ID Encoding

   A connection ID is used to ensure consistent routing of packets, as
   described in Section 6.1.  The long header contains two connection
   IDs: the Destination Connection ID is chosen by the recipient of the
   packet and is used to provide consistent routing; the Source
   Connection ID is used to set the Destination Connection ID used by
   the peer.

   During the handshake, packets with the long header are used to
   establish the connection ID that each endpoint uses.  Each endpoint
   uses the Source Connection ID field to specify the connection ID that
   is used in the Destination Connection ID field of packets being sent
   to them.  Upon receiving a packet, each endpoint sets the Destination
   Connection ID it sends to match the value of the Source Connection ID
   that they receive.

   During the handshake, an endpoint might receive multiple packets with
   the long header, and thus be given multiple opportunities to update
   the Destination Connection ID it sends.  A client MUST only change
   the value it sends in the Destination Connection ID in response to
   the first packet of each type it receives from the server (Retry or
   Initial); a server MUST set its value based on the Initial packet.
   Any additional changes are not permitted; if subsequent packets of
   those types include a different Source Connection ID, they MUST be
   discarded.  This avoids problems that might arise from stateless
   processing of multiple Initial packets producing different connection
   IDs.




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   Short headers only include the Destination Connection ID and omit the
   explicit length.  The length of the Destination Connection ID field
   is expected to be known to endpoints.

   Endpoints using a connection-ID based load balancer could agree with
   the load balancer on a fixed or minimum length and on an encoding for
   connection IDs.  This fixed portion could encode an explicit length,
   which allows the entire connection ID to vary in length and still be
   used by the load balancer.

   The very first packet sent by a client includes a random value for
   Destination Connection ID.  The same value MUST be used for all 0-RTT
   packets sent on that connection (Section 4.8).  This randomized value
   is used to determine the packet protection keys for Initial packets
   (see Section 5.1.1 of [QUIC-TLS]).

   A Version Negotiation (Section 4.3) packet MUST use both connection
   IDs selected by the client, swapped to ensure correct routing toward
   the client.

   The connection ID can change over the lifetime of a connection,
   especially in response to connection migration (Section 6.11).
   NEW_CONNECTION_ID frames (Section 7.13) are used to provide new
   connection ID values.

4.11.  Packet Numbers

   The packet number is an integer in the range 0 to 2^62-1.  The value
   is used in determining the cryptographic nonce for packet encryption.
   Each endpoint maintains a separate packet number for sending and
   receiving.

   Packet numbers are divided into 3 spaces in QUIC:

   o  Initial space: All Initial packets Section 4.6 are in this space.

   o  Handshake space: All Handshake packets Section 4.7 are in this
      space.

   o  Application data space: All 0-RTT and 1-RTT encrypted packets
      Section 4.8 are in this space.

   As described in [QUIC-TLS], each packet type uses different
   encryption keys.

   Conceptually, a packet number space is the encryption context in
   which a packet can be processed and ACKed.  Initial packets can only
   be sent with Initial encryption keys and ACKed in packets which are



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   also Initial packets.  Similarly, Handshake packets can only be sent
   and acknowledged in Handshake packets.

   This enforces cryptographic separation between the data sent in the
   different packet sequence number spaces.  Each packet number space
   starts at packet number 0.  Subsequent packets sent in the same
   packet number space MUST increase the packet number by at least one.

   0-RTT and 1-RTT data exist in the same packet number space to make
   loss recovery algorithms easier to implement between the two packet
   types.

   A QUIC endpoint MUST NOT reuse a packet number within the same packet
   number space in one connection (that is, under the same cryptographic
   keys).  If the packet number for sending reaches 2^62 - 1, the sender
   MUST close the connection without sending a CONNECTION_CLOSE frame or
   any further packets; an endpoint MAY send a Stateless Reset
   (Section 6.13.4) in response to further packets that it receives.

   In the QUIC long and short packet headers, the number of bits
   required to represent the packet number is reduced by including only
   a variable number of the least significant bits of the packet number.
   One or two of the most significant bits of the first octet determine
   how many bits of the packet number are provided, as shown in Table 2.

          +---------------------+----------------+--------------+
          | First octet pattern | Encoded Length | Bits Present |
          +---------------------+----------------+--------------+
          | 0b0xxxxxxx          | 1 octet        | 7            |
          |                     |                |              |
          | 0b10xxxxxx          | 2              | 14           |
          |                     |                |              |
          | 0b11xxxxxx          | 4              | 30           |
          +---------------------+----------------+--------------+

            Table 2: Packet Number Encodings for Packet Headers

   Note that these encodings are similar to those in Section 7.1, but
   use different values.

   The encoded packet number is protected as described in Section 5.3
   [QUIC-TLS].  Protection of the packet number is removed prior to
   recovering the full packet number.  The full packet number is
   reconstructed at the receiver based on the number of significant bits
   present, the content of those bits, and the largest packet number
   received on a successfully authenticated packet.  Recovering the full
   packet number is necessary to successfully remove packet protection.




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   Once packet number protection is removed, the packet number is
   decoded by finding the packet number value that is closest to the
   next expected packet.  The next expected packet is the highest
   received packet number plus one.  For example, if the highest
   successfully authenticated packet had a packet number of 0xaa82f30e,
   then a packet containing a 14-bit value of 0x9b3 will be decoded as
   0xaa8309b3.  Example pseudo-code for packet number decoding can be
   found in Appendix A.

   The sender MUST use a packet number size able to represent more than
   twice as large a range than the difference between the largest
   acknowledged packet and packet number being sent.  A peer receiving
   the packet will then correctly decode the packet number, unless the
   packet is delayed in transit such that it arrives after many higher-
   numbered packets have been received.  An endpoint SHOULD use a large
   enough packet number encoding to allow the packet number to be
   recovered even if the packet arrives after packets that are sent
   afterwards.

   As a result, the size of the packet number encoding is at least one
   more than the base 2 logarithm of the number of contiguous
   unacknowledged packet numbers, including the new packet.

   For example, if an endpoint has received an acknowledgment for packet
   0x6afa2f, sending a packet with a number of 0x6b2d79 requires a
   packet number encoding with 14 bits or more; whereas the 30-bit
   packet number encoding is needed to send a packet with a number of
   0x6bc107.

   A receiver MUST discard a newly unprotected packet unless it is
   certain that it has not processed another packet with the same packet
   number from the same packet number space.  Duplicate suppression MUST
   happen after removing packet protection for the reasons described in
   Section 9.3 of [QUIC-TLS].  An efficient algorithm for duplicate
   suppression can be found in Section 3.4.3 of [RFC2406].

   A Version Negotiation packet (Section 4.3) does not include a packet
   number.  The Retry packet (Section 4.4) has special rules for
   populating the packet number field.

5.  Frames and Frame Types

   The payload of all packets, after removing packet protection,
   consists of a sequence of frames, as shown in Figure 6.  Version
   Negotiation and Stateless Reset do not contain frames.






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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 6: Contents of Protected Payload

   Protected payloads MUST contain at least one frame, and MAY contain
   multiple frames and multiple frame types.

   Frames MUST fit within a single QUIC packet and MUST NOT span a QUIC
   packet boundary.  Each frame begins with a Frame Type, indicating its
   type, followed by additional type-dependent fields:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                       Frame Type (i)                        ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                   Type-Dependent Fields (*)                 ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                      Figure 7: Generic Frame Layout

   The frame types defined in this specification are listed in Table 3.
   The Frame Type in STREAM frames is used to carry other frame-specific
   flags.  For all other frames, the Frame Type field simply identifies
   the frame.  These frames are explained in more detail as they are
   referenced later in the document.















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            +-------------+-------------------+--------------+
            | Type Value  | Frame Type Name   | Definition   |
            +-------------+-------------------+--------------+
            | 0x00        | PADDING           | Section 7.2  |
            |             |                   |              |
            | 0x01        | RST_STREAM        | Section 7.3  |
            |             |                   |              |
            | 0x02        | CONNECTION_CLOSE  | Section 7.4  |
            |             |                   |              |
            | 0x03        | APPLICATION_CLOSE | Section 7.5  |
            |             |                   |              |
            | 0x04        | MAX_DATA          | Section 7.6  |
            |             |                   |              |
            | 0x05        | MAX_STREAM_DATA   | Section 7.7  |
            |             |                   |              |
            | 0x06        | MAX_STREAM_ID     | Section 7.8  |
            |             |                   |              |
            | 0x07        | PING              | Section 7.9  |
            |             |                   |              |
            | 0x08        | BLOCKED           | Section 7.10 |
            |             |                   |              |
            | 0x09        | STREAM_BLOCKED    | Section 7.11 |
            |             |                   |              |
            | 0x0a        | STREAM_ID_BLOCKED | Section 7.12 |
            |             |                   |              |
            | 0x0b        | NEW_CONNECTION_ID | Section 7.13 |
            |             |                   |              |
            | 0x0c        | STOP_SENDING      | Section 7.14 |
            |             |                   |              |
            | 0x0d        | ACK               | Section 7.15 |
            |             |                   |              |
            | 0x0e        | PATH_CHALLENGE    | Section 7.17 |
            |             |                   |              |
            | 0x0f        | PATH_RESPONSE     | Section 7.18 |
            |             |                   |              |
            | 0x10 - 0x17 | STREAM            | Section 7.20 |
            |             |                   |              |
            | 0x18        | CRYPTO            | Section 7.21 |
            |             |                   |              |
            | 0x19        | NEW_TOKEN         | Section 7.19 |
            |             |                   |              |
            | 0x1a        | ACK_ECN           | Section 7.16 |
            +-------------+-------------------+--------------+

                           Table 3: Frame Types






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   All QUIC frames are idempotent.  That is, a valid frame does not
   cause undesirable side effects or errors when received more than
   once.

   The Frame Type field uses a variable length integer encoding (see
   Section 7.1) with one exception.  To ensure simple and efficient
   implementations of frame parsing, a frame type MUST use the shortest
   possible encoding.  Though a two-, four- or eight-octet encoding of
   the frame types defined in this document is possible, the Frame Type
   field for these frames are encoded on a single octet.  For instance,
   though 0x4007 is a legitimate two-octet encoding for a variable-
   length integer with a value of 7, PING frames are always encoded as a
   single octet with the value 0x07.  An endpoint MUST treat the receipt
   of a frame type that uses a longer encoding than necessary as a
   connection error of type PROTOCOL_VIOLATION.

5.1.  Extension Frames

   QUIC frames do not use a self-describing encoding.  An endpoint
   therefore needs to understand the syntax of all frames before it can
   successfully process a packet.  This allows for efficient encoding of
   frames, but it means that an endpoint cannot send a frame of a type
   that is unknown to its peer.

   An extension to QUIC that wishes to use a new type of frame MUST
   first ensure that a peer is able to understand the frame.  An
   endpoint can use a transport parameter to signal its willingness to
   receive one or more extension frame types with the one transport
   parameter.

   An IANA registry is used to manage the assignment of frame types, see
   Section 13.2.

6.  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 6.4.  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 6.11.  Finally a connection may be terminated by either
   endpoint, as described in Section 6.13.








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6.1.  Connection ID

   Each connection possesses a set of identifiers, any of which could be
   used to distinguish it from other connections.  A connection ID can
   be either 0 octets in length, or between 4 and 18 octets (inclusive).
   Connection IDs are selected independently in each direction.

   The primary function of a connection ID is to ensure that changes in
   addressing at lower protocol layers (UDP, IP, and below) don't cause
   packets for a QUIC connection to be delivered to the wrong endpoint.
   Each endpoint selects connection IDs using an implementation-specific
   (and perhaps deployment-specific) method which will allow packets
   with that connection ID to be routed back to the endpoint and
   identified by the endpoint upon receipt.

   A zero-length connection ID MAY be used when the connection ID is not
   needed for routing and the address/port tuple of packets is
   sufficient to associate them to a connection.  An endpoint whose peer
   has selected a zero-length connection ID MUST continue to use a zero-
   length connection ID for the lifetime of the connection and MUST NOT
   send packets from any other local address.

   When an endpoint has requested a non-zero-length connection ID, it
   will issue a series of connection IDs over the lifetime of a
   connection.  The series of connection IDs issued by an endpoint is
   ordered, with the final connection ID selected during the handshake
   coming first.  Additional connection IDs are provided using the
   NEW_CONNECTION_ID frame (Section 7.13), each with a specified
   sequence number.  The series of connection IDs issued SHOULD be
   contiguous, but might not appear to be upon receipt due to reordering
   or loss.

   Each connection ID MUST be used on only one local address.  When
   packets are sent for the first time on a new local address, a new
   connection ID MUST be used with a higher sequence number than any
   connection ID previously used on any local address.  At any time, an
   endpoint MAY change to a new connection ID on a local address already
   in use.

   An endpoint MUST NOT send packets with a connection ID which has a
   lower sequence number than the highest sequence number of any
   connection ID ever sent or received on that local address.  This
   ensures that when an endpoint migrates to a new path or changes
   connection ID on an existing path, the packets will use a new
   connection ID in both directions.

   Implementations SHOULD ensure that peers have a connection ID with a
   matching sequence number available when changing to a new connection



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   ID.  An implementation could do this by always supplying a
   corresponding connection ID to a peer for each connection ID received
   from that peer.

   While endpoints select connection IDs as appropriate for their
   implementation, the connection ID MUST NOT include the unprotected
   sequence number.  Endpoints need to be able to recover the sequence
   number associated with each connection ID they generate without
   relying on information available to unaffiliated parties.  A
   connection ID that encodes an unencrypted sequence number could be
   used to correlate connection IDs across network paths.

6.2.  Matching Packets to Connections

   Incoming packets are classified on receipt.  Packets can either be
   associated with an existing connection, or - for servers -
   potentially create a new connection.

   Hosts try to associate a packet with an existing connection.  If the
   packet has a Destination Connection ID corresponding to an existing
   connection, QUIC processes that packet accordingly.  Note that more
   than one connection ID can be associated with a connection; see
   Section 6.1.

   If the Destination Connection ID is zero length and the packet
   matches the address/port tuple of a connection where the host did not
   require connection IDs, QUIC processes the packet as part of that
   connection.  Endpoints MUST drop packets with zero-length Destination
   Connection ID fields if they do not correspond to a single
   connection.

6.2.1.  Client Packet Handling

   Valid packets sent to clients always include a Destination Connection
   ID that matches the value the client selects.  Clients that choose to
   receive zero-length connection IDs can use the address/port tuple to
   identify a connection.  Packets that don't match an existing
   connection MAY be discarded.

   Due to packet reordering or loss, clients might receive packets for a
   connection that are encrypted with a key it has not yet computed.
   Clients MAY drop these packets, or MAY buffer them in anticipation of
   later packets that allow it to compute the key.

   If a client receives a packet that has an unsupported version, it
   MUST discard that packet.





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6.2.2.  Server Packet Handling

   If a server receives a packet that has an unsupported version and
   sufficient length to be an Initial packet for some version supported
   by the server, it SHOULD send a Version Negotiation packet as
   described in Section 6.3.1.  Servers MAY rate control these packets
   to avoid storms of Version Negotiation packets.

   The first packet for an unsupported version can use different
   semantics and encodings for any version-specific field.  In
   particular, different packet protection keys might be used for
   different versions.  Servers that do not support a particular version
   are unlikely to be able to decrypt the content of the packet.
   Servers SHOULD NOT attempt to decode or decrypt a packet from an
   unknown version, but instead send a Version Negotiation packet,
   provided that the packet is sufficiently long.

   Servers MUST drop other packets that contain unsupported versions.

   Packets with a supported version, or no version field, are matched to
   a connection as described in Section 6.2.  If not matched, the server
   continues below.

   If the packet is an Initial packet fully conforming with the
   specification, the server proceeds with the handshake (Section 6.4).
   This commits the server to the version that the client selected.

   If a server isn't currently accepting any new connections, it SHOULD
   send a Handshake packet containing a CONNECTION_CLOSE frame with
   error code SERVER_BUSY.

   If the packet is a 0-RTT packet, the server MAY buffer a limited
   number of these packets in anticipation of a late-arriving Initial
   Packet.  Clients are forbidden from sending Handshake packets prior
   to receiving a server response, so servers SHOULD ignore any such
   packets.

   Servers MUST drop incoming packets under all other circumstances.
   They SHOULD send a Stateless Reset (Section 6.13.4) if a connection
   ID is present in the header.

6.3.  Version Negotiation

   Version negotiation ensures that client and server agree to a QUIC
   version that is mutually supported.  A server sends a Version
   Negotiation packet in response to each packet that might initiate a
   new connection, see Section 6.2 for details.




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   The size of the first packet sent by a client will determine whether
   a server sends a Version Negotiation packet.  Clients that support
   multiple QUIC versions SHOULD pad their Initial packets to reflect
   the largest minimum Initial packet size of all their versions.  This
   ensures that the server responds if there are any mutually supported
   versions.

6.3.1.  Sending Version Negotiation Packets

   If the version selected by the client is not acceptable to the
   server, the server responds with a Version Negotiation packet (see
   Section 4.3).  This includes a list of versions that the server will
   accept.

   This system allows a server to process packets with unsupported
   versions without retaining state.  Though either the Initial packet
   or the Version Negotiation packet that is sent in response could be
   lost, the client will send new packets until it successfully receives
   a response or it abandons the connection attempt.

6.3.2.  Handling Version Negotiation Packets

   When the client receives a Version Negotiation packet, it first
   checks that the Destination and Source Connection ID fields match the
   Source and Destination Connection ID fields in a packet that the
   client sent.  If this check fails, the packet MUST be discarded.

   Once the Version Negotiation packet is determined to be valid, the
   client then selects an acceptable protocol version from the list
   provided by the server.  The client then attempts to create a
   connection using that version.  Though the contents of the Initial
   packet the client sends might not change in response to version
   negotiation, a client MUST increase the packet number it uses on
   every packet it sends.  Packets MUST continue to use long headers and
   MUST include the new negotiated protocol version.

   The client MUST use the long header format and include its selected
   version on all packets until it has 1-RTT keys and it has received a
   packet from the server which is not a Version Negotiation packet.

   A client MUST NOT change the version it uses unless it is in response
   to a Version Negotiation packet from the server.  Once a client
   receives a packet from the server which is not a Version Negotiation
   packet, it MUST discard other Version Negotiation packets on the same
   connection.  Similarly, a client MUST ignore a Version Negotiation
   packet if it has already received and acted on a Version Negotiation
   packet.




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   A client MUST ignore a Version Negotiation packet that lists the
   client's chosen version.

   A client MAY attempt 0-RTT after receiving a Version Negotiation
   packet.  A client that sends additional 0-RTT packets MUST NOT reset
   the packet number to 0 as a result, see Section 4.6.4.

   Version negotiation packets have no cryptographic protection.  The
   result of the negotiation MUST be revalidated as part of the
   cryptographic handshake (see Section 6.6.4).

6.3.3.  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 3) while generating a
   Version Negotiation packet.

   The design of version negotiation permits a server to avoid
   maintaining state for packets that it rejects in this fashion.  The
   validation of version negotiation (see Section 6.6.4) only validates
   the result of version negotiation, which is the same no matter which
   reserved version was sent.  A server MAY therefore send different
   reserved version numbers in the Version Negotiation Packet and in its
   transport parameters.

   A client MAY send a packet using a reserved version number.  This can
   be used to solicit a list of supported versions from a server.

6.4.  Cryptographic and Transport Handshake

   QUIC relies on a combined cryptographic and transport handshake to
   minimize connection establishment latency.  QUIC uses the CRYPTO
   frame Section 7.21 to transmit the cryptographic handshake.  Version
   0x00000001 of QUIC uses TLS 1.3 as described in [QUIC-TLS]; a
   different QUIC version number could indicate that a different
   cryptographic handshake protocol is in use.

   QUIC provides reliable, ordered delivery of the cryptographic
   handshake data.  QUIC packet protection ensures confidentiality and
   integrity protection that meets the requirements of the cryptographic
   handshake protocol:

   o  authenticated key exchange, where

      *  a server is always authenticated,

      *  a client is optionally authenticated,



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      *  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 6.6)

   o  authenticated confirmation of version negotiation (see
      Section 6.6.4)

   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 6.9)

   The initial CRYPTO frame MUST be sent in a single packet.  Any second
   attempt that is triggered by address validation MUST also be sent
   within a single packet.  This avoids having to reassemble a message
   from multiple packets.

   The first client packet of the cryptographic handshake protocol MUST
   fit within a 1232 octet QUIC packet payload.  This includes overheads
   that reduce the space available to the cryptographic handshake
   protocol.

   The CRYPTO frame can be sent in different packet number spaces.
   CRYPTO frames in each packet number space carry a separate sequence
   of handshake data starting from an offset of 0.

6.5.  Example Handshake Flows

   Details of how TLS is integrated with QUIC are provided in
   [QUIC-TLS], but some examples are provided here.

   Figure 8 provides an overview of the 1-RTT handshake.  Each line
   shows a QUIC packet with the packet type and packet number shown
   first, followed by the contents.  So, for instance the first packet
   is of type Initial, with packet number 0, and contains a CRYPTO frame
   carrying the ClientHello.

   Note that multiple QUIC packets - even of different encryption levels
   - may be coalesced into a single UDP datagram (see Section 4.9), and
   so this handshake may consist of as few as 4 UDP datagrams, or any



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   number more.  For instance, the server's first flight contains
   packets from the Initial encryption level (obfuscation), the
   Handshake level, and "0.5-RTT data" from the server at the 1-RTT
   encryption level.

   Client                                                  Server

   Initial[0]: CRYPTO[CH] ->

                                    Initial[0]: CRYPTO[SH] ACK[0]
                          Handshake[0]: CRYPTO[EE, CERT, CV, FIN]
                                    <- 1-RTT[0]: STREAM[1, "..."]

   Initial[1]: ACK[0]
   Handshake[0]: CRYPTO[FIN], ACK[0]
   1-RTT[0]: STREAM[0, "..."], ACK[0] ->

                              1-RTT[1]: STREAM[55, "..."], ACK[0]
                                          <- Handshake[1]: ACK[0]

                     Figure 8: Example 1-RTT Handshake

   Figure 9 shows an example of a connection with a 0-RTT handshake and
   a single packet of 0-RTT data.  Note that as described in
   Section 4.11, the server ACKs the 0-RTT data at the 1-RTT encryption
   level, and the client's sequence numbers at the 1-RTT encryption
   level continue to increment from its 0-RTT packets.

   Client                                                  Server

   Initial[0]: CRYPTO[CH]
   0-RTT[0]: STREAM[0, "..."] ->

                                    Initial[0]: CRYPTO[SH] ACK[0]
                           Handshake[0] CRYPTO[EE, CERT, CV, FIN]
                             <- 1-RTT[0]: STREAM[1, "..."] ACK[0]

   Initial[1]: ACK[0]
   0-RTT[1]: CRYPTO[EOED]
   Handshake[0]: CRYPTO[FIN], ACK[0]
   1-RTT[2]: STREAM[0, "..."] ACK[0] ->

                            1-RTT[1]: STREAM[55, "..."], ACK[1,2]
                                          <- Handshake[1]: ACK[0]

                     Figure 9: Example 1-RTT Handshake





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6.6.  Transport Parameters

   During connection establishment, both endpoints make authenticated
   declarations of their transport parameters.  These declarations are
   made unilaterally by each endpoint.  Endpoints are required to comply
   with the restrictions implied by these parameters; the description of
   each parameter includes rules for its handling.

   The format of the transport parameters is the TransportParameters
   struct from Figure 10.  This is described using the presentation
   language from Section 3 of [TLS13].








































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      uint32 QuicVersion;

      enum {
         initial_max_stream_data_bidi_local(0),
         initial_max_data(1),
         initial_max_bidi_streams(2),
         idle_timeout(3),
         preferred_address(4),
         max_packet_size(5),
         stateless_reset_token(6),
         ack_delay_exponent(7),
         initial_max_uni_streams(8),
         disable_migration(9),
         initial_max_stream_data_bidi_remote(10),
         initial_max_stream_data_uni(11),
         (65535)
      } TransportParameterId;

      struct {
         TransportParameterId parameter;
         opaque value<0..2^16-1>;
      } TransportParameter;

      struct {
         select (Handshake.msg_type) {
            case client_hello:
               QuicVersion initial_version;

            case encrypted_extensions:
               QuicVersion negotiated_version;
               QuicVersion supported_versions<4..2^8-4>;
         };
         TransportParameter parameters<22..2^16-1>;
      } TransportParameters;

      struct {
        enum { IPv4(4), IPv6(6), (15) } ipVersion;
        opaque ipAddress<4..2^8-1>;
        uint16 port;
        opaque connectionId<0..18>;
        opaque statelessResetToken[16];
      } PreferredAddress;

               Figure 10: 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.



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   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
   Section 6.6.4) before the connection establishment is considered
   properly complete.

   Definitions for each of the defined transport parameters are included
   in Section 6.6.1.  Any given parameter MUST appear at most once in a
   given transport parameters extension.  An endpoint MUST treat receipt
   of duplicate transport parameters as a connection error of type
   TRANSPORT_PARAMETER_ERROR.

6.6.1.  Transport Parameter Definitions

   An endpoint MUST include the following parameters in its encoded
   TransportParameters:

   idle_timeout (0x0003):  The idle timeout is a value in seconds that
      is encoded as an unsigned 16-bit integer.  The maximum value is
      600 seconds (10 minutes).

   An endpoint MAY use the following transport parameters:

   initial_max_data (0x0001):  The initial maximum data parameter
      contains the initial value for the maximum amount of data that can
      be sent on the connection.  This parameter is encoded as an
      unsigned 32-bit integer in units of octets.  This is equivalent to
      sending a MAX_DATA (Section 7.6) for the connection immediately
      after completing the handshake.  If the transport parameter is
      absent, the connection starts with a flow control limit of 0.

   initial_max_bidi_streams (0x0002):  The initial maximum bidirectional
      streams parameter contains the initial maximum number of
      bidirectional streams the peer may initiate, encoded as an
      unsigned 16-bit integer.  If this parameter is absent or zero,
      bidirectional streams cannot be created until a MAX_STREAM_ID
      frame is sent.  Setting this parameter is equivalent to sending a
      MAX_STREAM_ID (Section 7.8) immediately after completing the
      handshake containing the corresponding Stream ID.  For example, a
      value of 0x05 would be equivalent to receiving a MAX_STREAM_ID
      containing 16 when received by a client or 17 when received by a
      server.

   initial_max_uni_streams (0x0008):  The initial maximum unidirectional
      streams parameter contains the initial maximum number of
      unidirectional streams the peer may initiate, encoded as an



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      unsigned 16-bit integer.  If this parameter is absent or zero,
      unidirectional streams cannot be created until a MAX_STREAM_ID
      frame is sent.  Setting this parameter is equivalent to sending a
      MAX_STREAM_ID (Section 7.8) immediately after completing the
      handshake containing the corresponding Stream ID.  For example, a
      value of 0x05 would be equivalent to receiving a MAX_STREAM_ID
      containing 18 when received by a client or 19 when received by a
      server.

   max_packet_size (0x0005):  The maximum packet size parameter places a
      limit on the size of packets that the endpoint is willing to
      receive, encoded as an unsigned 16-bit integer.  This indicates
      that packets larger than this limit will be dropped.  The default
      for this parameter is the maximum permitted UDP payload of 65527.
      Values below 1200 are invalid.  This limit only applies to
      protected packets (Section 4.8).

   ack_delay_exponent (0x0007):  An 8-bit unsigned integer value
      indicating an exponent used to decode the ACK Delay field in the
      ACK frame, see Section 7.15.  If this value is absent, a default
      value of 3 is assumed (indicating a multiplier of 8).  The default
      value is also used for ACK frames that are sent in Initial and
      Handshake packets.  Values above 20 are invalid.

   disable_migration (0x0009):  The endpoint does not support connection
      migration (Section 6.11).  Peers MUST NOT send any packets,
      including probing packets (Section 6.11.1), from a local address
      other than that used to perform the handshake.  This parameter is
      a zero-length value.

   Either peer MAY advertise an initial value for the flow control on
   each type of stream on which they might receive data.  Each of the
   following transport parameters is encoded as an unsigned 32-bit
   integer in units of octets:

   initial_max_stream_data_bidi_local (0x0000):  The initial stream
      maximum data for bidirectional, locally-initiated streams
      parameter contains the initial flow control limit for newly
      created bidirectional streams opened by the endpoint that sets the
      transport parameter.  In client transport parameters, this applies
      to streams with an identifier ending in 0x0; in server transport
      parameters, this applies to streams ending in 0x1.

   initial_max_stream_data_bidi_remote (0x000a):  The initial stream
      maximum data for bidirectional, peer-initiated streams parameter
      contains the initial flow control limit for newly created
      bidirectional streams opened by the endpoint that receives the
      transport parameter.  In client transport parameters, this applies



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      to streams with an identifier ending in 0x1; in server transport
      parameters, this applies to streams ending in 0x0.

   initial_max_stream_data_uni (0x000b):  The initial stream maximum
      data for unidirectional streams parameter contains the initial
      flow control limit for newly created unidirectional streams opened
      by the endpoint that receives the transport parameter.  In client
      transport parameters, this applies to streams with an identifier
      ending in 0x3; in server transport parameters, this applies to
      streams ending in 0x2.

   If present, transport parameters that set initial stream flow control
   limits are equivalent to sending a MAX_STREAM_DATA frame
   (Section 7.7) on every stream of the corresponding type immediately
   after opening.  If the transport parameter is absent, streams of that
   type start with a flow control limit of 0.

   A server MAY include the following transport parameters:

   stateless_reset_token (0x0006):  The Stateless Reset Token is used in
      verifying a stateless reset, see Section 6.13.4.  This parameter
      is a sequence of 16 octets.

   preferred_address (0x0004):  The server's Preferred Address is used
      to effect a change in server address at the end of the handshake,
      as described in Section 6.12.

   A client MUST NOT include a stateless reset token or a preferred
   address.  A server MUST treat receipt of either transport parameter
   as a connection error of type TRANSPORT_PARAMETER_ERROR.

6.6.2.  Values of Transport Parameters for 0-RTT

   A client that attempts to send 0-RTT data MUST remember the transport
   parameters used by the server.  The transport parameters that the
   server advertises during connection establishment apply to all
   connections that are resumed using the keying material established
   during that handshake.  Remembered transport parameters apply to the
   new connection until the handshake completes and new transport
   parameters from the server can be provided.

   A server can remember the transport parameters that it advertised, or
   store an integrity-protected copy of the values in the ticket and
   recover the information when accepting 0-RTT data.  A server uses the
   transport parameters in determining whether to accept 0-RTT data.

   A server MAY accept 0-RTT and subsequently provide different values
   for transport parameters for use in the new connection.  If 0-RTT



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   data is accepted by the server, the server MUST NOT reduce any limits
   or alter any values that might be violated by the client with its
   0-RTT data.  In particular, a server that accepts 0-RTT data MUST NOT
   set values for initial_max_data, initial_max_stream_data_bidi_local,
   initial_max_stream_data_bidi_remote, and initial_max_stream_data_uni
   that are smaller than the remembered value of those parameters.
   Similarly, a server MUST NOT reduce the value of
   initial_max_bidi_streams or initial_max_uni_streams.

   Omitting or setting a zero value for certain transport parameters can
   result in 0-RTT data being enabled, but not usable.  The applicable
   subset of transport parameters that permit sending of application
   data SHOULD be set to non-zero values for 0-RTT.  This includes
   initial_max_data and either initial_max_bidi_streams and
   initial_max_stream_data_bidi_remote, or initial_max_uni_streams and
   initial_max_stream_data_uni.

   The value of the server's previous preferred_address MUST NOT be used
   when establishing a new connection; rather, the client should wait to
   observe the server's new preferred_address value in the handshake.

   A server MUST reject 0-RTT data or even abort a handshake if the
   implied values for transport parameters cannot be supported.

6.6.3.  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 13.1.

6.6.4.  Version Negotiation Validation

   Though the cryptographic handshake has integrity protection, two
   forms of QUIC version downgrade are possible.  In the first, an
   attacker replaces the QUIC version in the Initial packet.  In the
   second, a fake Version Negotiation packet is sent by an attacker.  To
   protect against these attacks, the transport parameters include three
   fields that encode version information.  These parameters are used to
   retroactively authenticate the choice of version (see Section 6.3).

   The cryptographic handshake provides integrity protection for the
   negotiated version as part of the transport parameters (see
   Section 6.6).  As a result, attacks on version negotiation by an
   attacker can be detected.



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   The client includes the initial_version field in its transport
   parameters.  The initial_version is the version that the client
   initially attempted to use.  If the server did not send a Version
   Negotiation packet Section 4.3, this will be identical to the
   negotiated_version field in the server transport parameters.

   A server that processes all packets in a stateful fashion can
   remember how version negotiation was performed and validate the
   initial_version value.

   A server that does not maintain state for every packet it receives
   (i.e., a stateless server) uses a different process.  If the
   initial_version matches the version of QUIC that is in use, a
   stateless server can accept the value.

   If the initial_version is different from the version of QUIC that is
   in use, a stateless server MUST check that it would have sent a
   Version Negotiation packet if it had received a packet with the
   indicated initial_version.  If a server would have accepted the
   version included in the initial_version and the value differs from
   the QUIC version that is in use, the server MUST terminate the
   connection with a VERSION_NEGOTIATION_ERROR error.

   The server includes both the version of QUIC that is in use and a
   list of the QUIC versions that the server supports.

   The negotiated_version field is the version that is in use.  This
   MUST be set by the server to the value that is on the Initial packet
   that it accepts (not an Initial packet that triggers a Retry or
   Version Negotiation packet).  A client that receives a
   negotiated_version that does not match the version of QUIC that is in
   use MUST terminate the connection with a VERSION_NEGOTIATION_ERROR
   error code.

   The server includes a list of versions that it would send in any
   version negotiation packet (Section 4.3) in the supported_versions
   field.  The server populates this field even if it did not send a
   version negotiation packet.

   The client validates that the negotiated_version is included in the
   supported_versions list and - if version negotiation was performed -
   that it would have selected the negotiated version.  A client MUST
   terminate the connection with a VERSION_NEGOTIATION_ERROR error code
   if the current QUIC version is not listed in the supported_versions
   list.  A client MUST terminate with a VERSION_NEGOTIATION_ERROR error
   code if version negotiation occurred but it would have selected a
   different version based on the value of the supported_versions list.




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   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.

6.7.  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 6.9), or to defer connection
   establishment costs.

   A server that generates a response to an Initial packet without
   retaining connection state MUST use the Retry packet (Section 4.4).
   This packet causes a client to restart the connection attempt and
   includes the token in the new Initial packet (Section 4.6) to prove
   source address ownership.

6.8.  Using Explicit Congestion Notification

   QUIC endpoints use Explicit Congestion Notification (ECN) [RFC3168]
   to detect and respond to network congestion.  ECN allows a network
   node to indicate congestion in the network by setting a codepoint in
   the IP header of a packet instead of dropping it.  Endpoints react to
   congestion by reducing their sending rate in response, as described
   in [QUIC-RECOVERY].

   To use ECN, QUIC endpoints first determine whether a path and peer
   support ECN marking.  Verifying the path occurs at the beginning of a
   connection and when the connection migrates to a new path (see
   Section 6.11).

   Each endpoint independently verifies and enables ECN for the path
   from it to the peer.

   To verify that both a path and the peer support ECN, an endpoint MUST
   set one of the ECN Capable Transport (ECT) codepoints - ECT(0) or
   ECT(1) - in the IP header [RFC8311] of all outgoing packets.

   If an ECT codepoint set in the IP header is not corrupted by a
   network device, then a received packet contains either the codepoint
   sent by the peer or the Congestion Experienced (CE) codepoint set by
   a network device that is experiencing congestion.



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   On receiving a packet with an ECT or CE codepoint, an endpoint that
   supports ECN increases the corresponding ECT(0), ECT(1), or CE count,
   and includes these counters in subsequent (see Section 8.1) ACK_ECN
   frames (see Section 7.16).

   A packet detected by a receiver as a duplicate does not affect the
   receiver's local ECN codepoint counts; see (Section 12.7) for
   relevant security concerns.

   If an endpoint receives a packet without an ECT or CE codepoint, it
   responds per Section 8.1 with an ACK frame.

   If an endpoint does not support ECN or does not have access to
   received ECN codepoints, it acknowledges received packets per
   Section 8.1 with an ACK frame.

   If a packet sent with an ECT codepoint is newly acknowledged by the
   peer in an ACK frame, the endpoint stops setting ECT codepoints in
   subsequent packets, with the expectation that either the network or
   the peer no longer supports ECN.

   To protect the connection from arbitrary corruption of ECN codepoints
   by the network, an endpoint verifies the following when an ACK_ECN
   frame is received:

   o  The increase in ECT(0) and ECT(1) counters MUST be at least the
      number of packets newly acknowledged that were sent with the
      corresponding codepoint.

   o  The total increase in ECT(0), ECT(1), and CE counters reported in
      the ACK_ECN frame MUST be at least the total number of packets
      newly acknowledged in this ACK_ECN frame.

   An endpoint could miss acknowledgements for a packet when ACK frames
   are lost.  It is therefore possible for the total increase in ECT(0),
   ECT(1), and CE counters to be greater than the number of packets
   acknowledged in an ACK frame.  When this happens, the local reference
   counts MUST be increased to match the counters in the ACK frame.

   Upon successful verification, an endpoint continues to set ECT
   codepoints in subsequent packets with the expectation that the path
   is ECN-capable.

   If verification fails, then the endpoint ceases setting ECT
   codepoints in subsequent packets with the expectation that either the
   network or the peer does not support ECN.





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   If an endpoint sets ECT codepoints on outgoing packets and encounters
   a retransmission timeout due to the absence of acknowledgments from
   the peer (see [QUIC-RECOVERY]), or if an endpoint has reason to
   believe that a network element might be corrupting ECN codepoints,
   the endpoint MAY cease setting ECT codepoints in subsequent packets.
   Doing so allows the connection to traverse network elements that drop
   or corrupt ECN codepoints in the IP header.

6.9.  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.

   This technique is used primarily to avoid QUIC from being used for
   traffic amplification attack.  In such an attack, a packet is sent to
   a server with spoofed source address information that identifies a
   victim.  If a server generates more or larger packets in response to
   that packet, the attacker can use the server to send more data toward
   the victim than it would be able to send on its own.

   Several methods are used in QUIC to mitigate this attack.  Firstly,
   the initial handshake packet is sent in a UDP datagram that contains
   at least 1200 octets of UDP payload.  This allows a server to send a
   similar amount of data without risking causing an amplification
   attack toward an unproven remote address.

   A server eventually confirms that a client has received its messages
   when the first Handshake-level message is received.  This might be
   insufficient, either because the server wishes to avoid the
   computational cost of completing the handshake, or it might be that
   the size of the packets that are sent during the handshake is too
   large.  This is especially important for 0-RTT, where the server
   might wish to provide application data traffic - such as a response
   to a request - in response to the data carried in the early data from
   the client.

   To send additional data prior to completing the cryptographic
   handshake, the server then needs to validate that the client owns the
   address that it claims.

   Source address validation is therefore performed by the core
   transport protocol during the establishment of a connection.

   A different type of source address validation is performed after a
   connection migration, see Section 6.10.



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6.9.1.  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's authenticity can be checked (see
   Section 6.9.3) and the client is able to return that token, it proves
   to the server that it received the token.

   Upon receiving the client's Initial packet, the server can request
   address validation by sending a Retry packet containing a token.
   This token is repeated in the client's next Initial packet.  Because
   the token is consumed by the server that generates it, there is no
   need for a single well-defined format.  A token could include
   information about the claimed client address (IP and port), a
   timestamp, and any other supplementary information the server will
   need to validate the token in the future.

   The Retry packet is sent to the client and a legitimate client will
   respond with an Initial packet containing the token from the Retry
   packet when it continues the handshake.  In response to receiving the
   token, a server can either abort the connection or permit it to
   proceed.

   A connection MAY be accepted without address validation - or with
   only limited validation - but a server SHOULD limit the data it sends
   toward an unvalidated address.  Successful completion of the
   cryptographic handshake implicitly provides proof that the client has
   received packets from the server.

   The client should allow for additional Retry packets being sent in
   response to Initial packets sent containing a token.  There are
   several situations in which the server might not be able to use the
   previously generated token to validate the client's address and must
   send a new Retry.  A reasonable limit to the number of tries the
   client allows for, before giving up, is 3.  That is, the client MUST
   echo the address validation token from a new Retry packet up to 3
   times.  After that, it MAY give up on the connection attempt.

6.9.2.  Address Validation for Future Connections

   A server MAY provide clients with an address validation token during
   one connection that can be used on a subsequent connection.  Address
   validation is especially important with 0-RTT because a server
   potentially sends a significant amount of data to a client in
   response to 0-RTT data.

   The server uses the NEW_TOKEN frame Section 7.19 to provide the
   client with an address validation token that can be used to validate



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   future connections.  The client may then use this token to validate
   future connections by including it in the Initial packet's header.
   The client MUST NOT use the token provided in a Retry for future
   connections.

   Unlike the token that is created for a Retry packet, there might be
   some time between when the token is created and when the token is
   subsequently used.  Thus, a resumption token SHOULD include an
   expiration time.  The server MAY include either an explicit
   expiration time or an issued timestamp and dynamically calculate the
   expiration time.  It is also unlikely that the client port number is
   the same on two different connections; validating the port is
   therefore unlikely to be successful.

6.9.3.  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.

6.10.  Path Validation

   Path validation is used by an endpoint to verify reachability of a
   peer over a specific path.  That is, it tests reachability between a
   specific local address and a specific peer address, where an address
   is the two-tuple of IP address and port.  Path validation tests that
   packets can be both sent to and received from a peer.

   Path validation is used during connection migration (see Section 6.11
   and Section 6.12) by the migrating endpoint to verify reachability of
   a peer from a new local address.  Path validation is also used by the
   peer to verify that the migrating endpoint is able to receive packets
   sent to the its new address.  That is, that the packets received from
   the migrating endpoint do not carry a spoofed source address.

   Path validation can be used at any time by either endpoint.  For
   instance, an endpoint might check that a peer is still in possession
   of its address after a period of quiescence.





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   Path validation is not designed as a NAT traversal mechanism.  Though
   the mechanism described here might be effective for the creation of
   NAT bindings that support NAT traversal, the expectation is that one
   or other peer is able to receive packets without first having sent a
   packet on that path.  Effective NAT traversal needs additional
   synchronization mechanisms that are not provided here.

   An endpoint MAY bundle PATH_CHALLENGE and PATH_RESPONSE frames that
   are used for path validation with other frames.  For instance, an
   endpoint may pad a packet carrying a PATH_CHALLENGE for PMTU
   discovery, or an endpoint may bundle a PATH_RESPONSE with its own
   PATH_CHALLENGE.

6.10.1.  Initiation

   To initiate path validation, an endpoint sends a PATH_CHALLENGE frame
   containing a random payload on the path to be validated.

   An endpoint MAY send additional PATH_CHALLENGE frames to handle
   packet loss.  An endpoint SHOULD NOT send a PATH_CHALLENGE more
   frequently than it would an Initial packet, ensuring that connection
   migration is no more load on a new path than establishing a new
   connection.

   The endpoint MUST use fresh random data in every PATH_CHALLENGE frame
   so that it can associate the peer's response with the causative
   PATH_CHALLENGE.

6.10.2.  Response

   On receiving a PATH_CHALLENGE frame, an endpoint MUST respond
   immediately by echoing the data contained in the PATH_CHALLENGE frame
   in a PATH_RESPONSE frame, with the following stipulation.  Since a
   PATH_CHALLENGE might be sent from a spoofed address, an endpoint MAY
   limit the rate at which it sends PATH_RESPONSE frames and MAY
   silently discard PATH_CHALLENGE frames that would cause it to respond
   at a higher rate.

   To ensure that packets can be both sent to and received from the
   peer, the PATH_RESPONSE MUST be sent on the same path as the
   triggering PATH_CHALLENGE: from the same local address on which the
   PATH_CHALLENGE was received, to the same remote address from which
   the PATH_CHALLENGE was received.








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6.10.3.  Completion

   A new address is considered valid when a PATH_RESPONSE frame is
   received containing data that was sent in a previous PATH_CHALLENGE.
   Receipt of an acknowledgment for a packet containing a PATH_CHALLENGE
   frame is not adequate validation, since the acknowledgment can be
   spoofed by a malicious peer.

   For path validation to be successful, a PATH_RESPONSE frame MUST be
   received from the same remote address to which the corresponding
   PATH_CHALLENGE was sent.  If a PATH_RESPONSE frame is received from a
   different remote address than the one to which the PATH_CHALLENGE was
   sent, path validation is considered to have failed, even if the data
   matches that sent in the PATH_CHALLENGE.

   Additionally, the PATH_RESPONSE frame MUST be received on the same
   local address from which the corresponding PATH_CHALLENGE was sent.
   If a PATH_RESPONSE frame is received on a different local address
   than the one from which the PATH_CHALLENGE was sent, path validation
   is considered to have failed, even if the data matches that sent in
   the PATH_CHALLENGE.  Thus, the endpoint considers the path to be
   valid when a PATH_RESPONSE frame is received on the same path with
   the same payload as the PATH_CHALLENGE frame.

6.10.4.  Abandonment

   An endpoint SHOULD abandon path validation after sending some number
   of PATH_CHALLENGE frames or after some time has passed.  When setting
   this timer, implementations are cautioned that the new path could
   have a longer round-trip time than the original.

   Note that the endpoint might receive packets containing other frames
   on the new path, but a PATH_RESPONSE frame with appropriate data is
   required for path validation to succeed.

   If path validation fails, the path is deemed unusable.  This does not
   necessarily imply a failure of the connection - endpoints can
   continue sending packets over other paths as appropriate.  If no
   paths are available, an endpoint can wait for a new path to become
   available or close the connection.

   A path validation might be abandoned for other reasons besides
   failure.  Primarily, this happens if a connection migration to a new
   path is initiated while a path validation on the old path is in
   progress.






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6.11.  Connection Migration

   QUIC allows connections to survive changes to endpoint addresses
   (that is, IP address and/or port), such as those caused by a endpoint
   migrating to a new network.  This section describes the process by
   which an endpoint migrates to a new address.

   An endpoint MUST NOT initiate connection migration before the
   handshake is finished and the endpoint has 1-RTT keys.  The design of
   QUIC relies on endpoints retaining a stable address for the duration
   of the handshake.

   An endpoint also MUST NOT initiate connection migration if the peer
   sent the "disable_migration" transport parameter during the
   handshake.  An endpoint which has sent this transport parameter, but
   detects that a peer has nonetheless migrated to a different network
   MAY treat this as a connection error of type INVALID_MIGRATION.

   Not all changes of peer address are intentional migrations.  The peer
   could experience NAT rebinding: a change of address due to a
   middlebox, usually a NAT, allocating a new outgoing port or even a
   new outgoing IP address for a flow.  Endpoints SHOULD perform path
   validation (Section 6.10) if a NAT rebinding does not cause the
   connection to fail.

   This document limits migration of connections to new client
   addresses, except as described in Section 6.12.  Clients are
   responsible for initiating all migrations.  Servers do not send non-
   probing packets (see Section 6.11.1) toward a client address until it
   sees a non-probing packet from that address.  If a client receives
   packets from an unknown server address, the client MAY discard these
   packets.

6.11.1.  Probing a New Path

   An endpoint MAY probe for peer reachability from a new local address
   using path validation Section 6.10 prior to migrating the connection
   to the new local address.  Failure of path validation simply means
   that the new path is not usable for this connection.  Failure to
   validate a path does not cause the connection to end unless there are
   no valid alternative paths available.

   An endpoint uses a new connection ID for probes sent from a new local
   address, see Section 6.11.5 for further discussion.

   Receiving a PATH_CHALLENGE frame from a peer indicates that the peer
   is probing for reachability on a path.  An endpoint sends a
   PATH_RESPONSE in response as per Section 6.10.



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   PATH_CHALLENGE, PATH_RESPONSE, and PADDING frames are "probing
   frames", and all other frames are "non-probing frames".  A packet
   containing only probing frames is a "probing packet", and a packet
   containing any other frame is a "non-probing packet".

6.11.2.  Initiating Connection Migration

   A endpoint can migrate a connection to a new local address by sending
   packets containing frames other than probing frames from that
   address.

   Each endpoint validates its peer's address during connection
   establishment.  Therefore, a migrating endpoint can send to its peer
   knowing that the peer is willing to receive at the peer's current
   address.  Thus an endpoint can migrate to a new local address without
   first validating the peer's address.

   When migrating, the new path might not support the endpoint's current
   sending rate.  Therefore, the endpoint resets its congestion
   controller, as described in Section 6.11.4.

   The new path might not have the same ECN capability.  Therefore, the
   endpoint verifies ECN capability as described in Section 6.8.

   Receiving acknowledgments for data sent on the new path serves as
   proof of the peer's reachability from the new address.  Note that
   since acknowledgments may be received on any path, return
   reachability on the new path is not established.  To establish return
   reachability on the new path, an endpoint MAY concurrently initiate
   path validation Section 6.10 on the new path.

6.11.3.  Responding to Connection Migration

   Receiving a packet from a new peer address containing a non-probing
   frame indicates that the peer has migrated to that address.

   In response to such a packet, an endpoint MUST start sending
   subsequent packets to the new peer address and MUST initiate path
   validation (Section 6.10) to verify the peer's ownership of the
   unvalidated address.

   An endpoint MAY send data to an unvalidated peer address, but it MUST
   protect against potential attacks as described in Section 6.11.3.1
   and Section 6.11.3.2.  An endpoint MAY skip validation of a peer
   address if that address has been seen recently.

   An endpoint only changes the address that it sends packets to in
   response to the highest-numbered non-probing packet.  This ensures



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   that an endpoint does not send packets to an old peer address in the
   case that it receives reordered packets.

   After changing the address to which it sends non-probing packets, an
   endpoint could abandon any path validation for other addresses.

   Receiving a packet from a new peer address might be the result of a
   NAT rebinding at the peer.

   After verifying a new client address, the server SHOULD send new
   address validation tokens (Section 6.9) to the client.

6.11.3.1.  Handling Address Spoofing by a Peer

   It is possible that a peer is spoofing its source address to cause an
   endpoint to send excessive amounts of data to an unwilling host.  If
   the endpoint sends significantly more data than the spoofing peer,
   connection migration might be used to amplify the volume of data that
   an attacker can generate toward a victim.

   As described in Section 6.11.3, an endpoint is required to validate a
   peer's new address to confirm the peer's possession of the new
   address.  Until a peer's address is deemed valid, an endpoint MUST
   limit the rate at which it sends data to this address.  The endpoint
   MUST NOT send more than a minimum congestion window's worth of data
   per estimated round-trip time (kMinimumWindow, as defined in
   [QUIC-RECOVERY]).  In the absence of this limit, an endpoint risks
   being used for a denial of service attack against an unsuspecting
   victim.  Note that since the endpoint will not have any round-trip
   time measurements to this address, the estimate SHOULD be the default
   initial value (see [QUIC-RECOVERY]).

   If an endpoint skips validation of a peer address as described in
   Section 6.11.3, it does not need to limit its sending rate.

6.11.3.2.  Handling Address Spoofing by an On-path Attacker

   An on-path attacker could cause a spurious connection migration by
   copying and forwarding a packet with a spoofed address such that it
   arrives before the original packet.  The packet with the spoofed
   address will be seen to come from a migrating connection, and the
   original packet will be seen as a duplicate and dropped.  After a
   spurious migration, validation of the source address will fail
   because the entity at the source address does not have the necessary
   cryptographic keys to read or respond to the PATH_CHALLENGE frame
   that is sent to it even if it wanted to.





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   To protect the connection from failing due to such a spurious
   migration, an endpoint MUST revert to using the last validated peer
   address when validation of a new peer address fails.

   If an endpoint has no state about the last validated peer address, it
   MUST close the connection silently by discarding all connection
   state.  This results in new packets on the connection being handled
   generically.  For instance, an endpoint MAY send a stateless reset in
   response to any further incoming packets.

   Note that receipt of packets with higher packet numbers from the
   legitimate peer address will trigger another connection migration.
   This will cause the validation of the address of the spurious
   migration to be abandoned.

6.11.4.  Loss Detection and Congestion Control

   The capacity available on the new path might not be the same as the
   old path.  Packets sent on the old path SHOULD NOT contribute to
   congestion control or RTT estimation for the new path.

   On confirming a peer's ownership of its new address, an endpoint
   SHOULD immediately reset the congestion controller and round-trip
   time estimator for the new path.

   An endpoint MUST NOT return to the send rate used for the previous
   path unless it is reasonably sure that the previous send rate is
   valid for the new path.  For instance, a change in the client's port
   number is likely indicative of a rebinding in a middlebox and not a
   complete change in path.  This determination likely depends on
   heuristics, which could be imperfect; if the new path capacity is
   significantly reduced, ultimately this relies on the congestion
   controller responding to congestion signals and reducing send rates
   appropriately.

   There may be apparent reordering at the receiver when an endpoint
   sends data and probes from/to multiple addresses during the migration
   period, since the two resulting paths may have different round-trip
   times.  A receiver of packets on multiple paths will still send ACK
   frames covering all received packets.

   While multiple paths might be used during connection migration, a
   single congestion control context and a single loss recovery context
   (as described in [QUIC-RECOVERY]) may be adequate.  A sender can make
   exceptions for probe packets so that their loss detection is
   independent and does not unduly cause the congestion controller to
   reduce its sending rate.  An endpoint might set a separate timer when
   a PATH_CHALLENGE is sent, which is cancelled when the corresponding



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   PATH_RESPONSE is received.  If the timer fires before the
   PATH_RESPONSE is received, the endpoint might send a new
   PATH_CHALLENGE, and restart the timer for a longer period of time.

6.11.5.  Privacy Implications of Connection Migration

   Using a stable connection ID on multiple network paths allows a
   passive observer to correlate activity between those paths.  An
   endpoint that moves between networks might not wish to have their
   activity correlated by any entity other than their peer, so different
   connection IDs are used when sending from different local addresses,
   as discussed in Section 6.1.

   This eliminates the use of the connection ID for linking activity
   from the same connection on different networks.  Protection of packet
   numbers ensures that packet numbers cannot be used to correlate
   activity.  This does not prevent other properties of packets, such as
   timing and size, from being used to correlate activity.

   Clients MAY move to a new connection ID at any time based on
   implementation-specific concerns.  For example, after a period of
   network inactivity NAT rebinding might occur when the client begins
   sending data again.

   A client might wish to reduce linkability by employing a new
   connection ID and source UDP port when sending traffic after a period
   of inactivity.  Changing the UDP port from which it sends packets at
   the same time might cause the packet to appear as a connection
   migration.  This ensures that the mechanisms that support migration
   are exercised even for clients that don't experience NAT rebindings
   or genuine migrations.  Changing port number can cause a peer to
   reset its congestion state (see Section 6.11.4), so the port SHOULD
   only be changed infrequently.

6.12.  Server's Preferred Address

   QUIC allows servers to accept connections on one IP address and
   attempt to transfer these connections to a more preferred address
   shortly after the handshake.  This is particularly useful when
   clients initially connect to an address shared by multiple servers
   but would prefer to use a unicast address to ensure connection
   stability.  This section describes the protocol for migrating a
   connection to a preferred server address.

   Migrating a connection to a new server address mid-connection is left
   for future work.  If a client receives packets from a new server
   address not indicated by the preferred_address transport parameter,
   the client SHOULD discard these packets.



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6.12.1.  Communicating A Preferred Address

   A server conveys a preferred address by including the
   preferred_address transport parameter in the TLS handshake.

   Once the handshake is finished, the client SHOULD initiate path
   validation (see Section 6.10) of the server's preferred address using
   the connection ID provided in the preferred_address transport
   parameter.

   If path validation succeeds, the client SHOULD immediately begin
   sending all future packets to the new server address using the new
   connection ID and discontinue use of the old server address.  If path
   validation fails, the client MUST continue sending all future packets
   to the server's original IP address.

6.12.2.  Responding to Connection Migration

   A server might receive a packet addressed to its preferred IP address
   at any time after the handshake is completed.  If this packet
   contains a PATH_CHALLENGE frame, the server sends a PATH_RESPONSE
   frame as per Section 6.10, but the server MUST continue sending all
   other packets from its original IP address.

   The server SHOULD also initiate path validation of the client using
   its preferred address and the address from which it received the
   client probe.  This helps to guard against spurious migration
   initiated by an attacker.

   Once the server has completed its path validation and has received a
   non-probing packet with a new largest packet number on its preferred
   address, the server begins sending to the client exclusively from its
   preferred IP address.  It SHOULD drop packets for this connection
   received on the old IP address, but MAY continue to process delayed
   packets.

6.12.3.  Interaction of Client Migration and Preferred Address

   A client might need to perform a connection migration before it has
   migrated to the server's preferred address.  In this case, the client
   SHOULD perform path validation to both the original and preferred
   server address from the client's new address concurrently.

   If path validation of the server's preferred address succeeds, the
   client MUST abandon validation of the original address and migrate to
   using the server's preferred address.  If path validation of the
   server's preferred address fails, but validation of the server's




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   original address succeeds, the client MAY migrate to using the
   original address from the client's new address.

   If the connection to the server's preferred address is not from the
   same client address, the server MUST protect against potential
   attacks as described in Section 6.11.3.1 and Section 6.11.3.2.  In
   addition to intentional simultaneous migration, this might also occur
   because the client's access network used a different NAT binding for
   the server's preferred address.

   Servers SHOULD initiate path validation to the client's new address
   upon receiving a probe packet from a different address.  Servers MUST
   NOT send more than a minimum congestion window's worth of non-probing
   packets to the new address before path validation is complete.

6.13.  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:

   o  idle timeout (Section 6.13.2)

   o  immediate close (Section 6.13.3)

   o  stateless reset (Section 6.13.4)

6.13.1.  Closing and Draining Connection States

   The closing and draining connection states exist to ensure that
   connections close cleanly and that delayed or reordered packets are
   properly discarded.  These states SHOULD persist for three times the
   current Retransmission Timeout (RTO) interval as defined in
   [QUIC-RECOVERY].

   An endpoint enters a closing period after initiating an immediate
   close (Section 6.13.3).  While closing, an endpoint MUST NOT send
   packets unless they contain a CONNECTION_CLOSE or APPLICATION_CLOSE
   frame (see Section 6.13.3 for details).

   In the closing state, only a packet containing a closing frame can be
   sent.  An endpoint retains only enough information to generate a
   packet containing a closing frame and to identify packets as
   belonging to the connection.  The connection ID and QUIC version is
   sufficient information to identify packets for a closing connection;
   an endpoint can discard all other connection state.  An endpoint MAY
   retain packet protection keys for incoming packets to allow it to
   read and process a closing frame.



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   The draining state is entered once an endpoint receives a signal that
   its peer is closing or draining.  While otherwise identical to the
   closing state, an endpoint in the draining state MUST NOT send any
   packets.  Retaining packet protection keys is unnecessary once a
   connection is in the draining state.

   An endpoint MAY transition from the closing period to the draining
   period if it can confirm that its peer is also closing or draining.
   Receiving a closing frame is sufficient confirmation, as is receiving
   a stateless reset.  The draining period SHOULD end when the closing
   period would have ended.  In other words, the endpoint can use the
   same end time, but cease retransmission of the closing packet.

   Disposing of connection state prior to the end of the closing or
   draining period could cause delayed or reordered packets to be
   handled poorly.  Endpoints that have some alternative means to ensure
   that late-arriving packets on the connection do not create QUIC
   state, such as those that are able to close the UDP socket, MAY use
   an abbreviated draining period which can allow for faster resource
   recovery.  Servers that retain an open socket for accepting new
   connections SHOULD NOT exit the closing or draining period early.

   Once the closing or draining period has ended, an endpoint SHOULD
   discard all connection state.  This results in new packets on the
   connection being handled generically.  For instance, an endpoint MAY
   send a stateless reset in response to any further incoming packets.

   The draining and closing periods do not apply when a stateless reset
   (Section 6.13.4) is sent.

   An endpoint is not expected to handle key updates when it is closing
   or draining.  A key update might prevent the endpoint from moving
   from the closing state to draining, but it otherwise has no impact.

   An endpoint could receive packets from a new source address,
   indicating a client connection migration (Section 6.11), while in the
   closing period.  An endpoint in the closing state MUST strictly limit
   the number of packets it sends to this new address until the address
   is validated (see Section 6.10).  A server in the closing state MAY
   instead choose to discard packets received from a new source address.

6.13.2.  Idle Timeout

   A connection that remains idle for longer than the advertised idle
   timeout (see Section 6.6.1) is closed.  A connection enters the
   draining state when the idle timeout expires.





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   Each endpoint advertises their own idle timeout to their peer.  The
   idle timeout starts from the last packet received.  In order to
   ensure that initiating new activity postpones an idle timeout, an
   endpoint restarts this timer when sending a packet.  An endpoint does
   not postpone the idle timeout if another packet has been sent
   containing frames other than ACK or PADDING, and that other packet
   has not been acknowledged or declared lost.  Packets that contain
   only ACK or PADDING frames are not acknowledged until an endpoint has
   other frames to send, so they could prevent the timeout from being
   refreshed.

   The value for an idle timeout can be asymmetric.  The value
   advertised by an endpoint is only used to determine whether the
   connection is live at that endpoint.  An endpoint that sends packets
   near the end of the idle timeout period of a peer risks having those
   packets discarded if its peer enters the draining state before the
   packets arrive.  If a peer could timeout within an RTO (see
   Section 4.3.3 of [QUIC-RECOVERY]), it is advisable to test for
   liveness before sending any data that cannot be retried safely.

6.13.3.  Immediate Close

   An endpoint sends a closing frame (CONNECTION_CLOSE or
   APPLICATION_CLOSE) to terminate the connection immediately.  Any
   closing frame causes all streams to immediately become closed; open
   streams can be assumed to be implicitly reset.

   After sending a closing frame, endpoints immediately enter the
   closing state.  During the closing period, an endpoint that sends a
   closing frame SHOULD respond to any packet that it receives with
   another packet containing a closing frame.  To minimize the state
   that an endpoint maintains for a closing connection, endpoints MAY
   send the exact same packet.  However, endpoints SHOULD limit the
   number of packets they generate containing a closing frame.  For
   instance, an endpoint could progressively increase the number of
   packets that it receives before sending additional packets or
   increase the time between packets.

   Note:  Allowing retransmission of a packet contradicts other advice
      in this document that recommends the creation of new packet
      numbers for every packet.  Sending new packet numbers is primarily
      of advantage to loss recovery and congestion control, which are
      not expected to be relevant for a closed connection.
      Retransmitting the final packet requires less state.

   After receiving a closing frame, endpoints enter the draining state.
   An endpoint that receives a closing frame MAY send a single packet
   containing a closing frame before entering the draining state, using



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   a CONNECTION_CLOSE frame and a NO_ERROR code if appropriate.  An
   endpoint MUST NOT send further packets, which could result in a
   constant exchange of closing frames until the closing period on
   either peer ended.

   An immediate close can be used after an application protocol has
   arranged to close a connection.  This might be after the application
   protocols negotiates a graceful shutdown.  The application protocol
   exchanges whatever messages that are needed to cause both endpoints
   to agree to close the connection, after which the application
   requests that the connection be closed.  The application protocol can
   use an APPLICATION_CLOSE message with an appropriate error code to
   signal closure.

6.13.4.  Stateless Reset

   A stateless reset is provided as an option of last resort for an
   endpoint that does not have access to the state of a connection.  A
   crash or outage might result in peers continuing to send data to an
   endpoint that is unable to properly continue the connection.  An
   endpoint that wishes to communicate a fatal connection error MUST use
   a closing frame if it has sufficient state to do so.

   To support this process, a token is sent by endpoints.  The token is
   carried in the NEW_CONNECTION_ID frame sent by either peer, and
   servers can specify the stateless_reset_token transport parameter
   during the handshake (clients cannot because their transport
   parameters don't have confidentiality protection).  This value is
   protected by encryption, so only client and server know this value.

   An endpoint that receives packets that it cannot process sends a
   packet in the following layout:



















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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|K|1|1|0|0|0|0|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                      Random Octets (160..)                  ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +                                                               +
   |                                                               |
   +                   Stateless Reset Token (128)                 +
   |                                                               |
   +                                                               +
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                     Figure 11: Stateless Reset Packet

   This design ensures that a stateless reset packet is - to the extent
   possible - indistinguishable from a regular packet with a short
   header.

   The message consists of a header octet, followed by an arbitrary
   number of random octets, followed by a Stateless Reset Token.

   A stateless reset will be interpreted by a recipient as a packet with
   a short header.  For the packet to appear as valid, the Random Octets
   field needs to include at least 20 octets of random or unpredictable
   values.  This is intended to allow for a destination connection ID of
   the maximum length permitted, a packet number, and minimal payload.
   The Stateless Reset Token corresponds to the minimum expansion of the
   packet protection AEAD.  More random octets might be necessary if the
   endpoint could have negotiated a packet protection scheme with a
   larger minimum AEAD expansion.

   An endpoint SHOULD NOT send a stateless reset that is significantly
   larger than the packet it receives.  Endpoints MUST discard packets
   that are too small to be valid QUIC packets.  With the set of AEAD
   functions defined in [QUIC-TLS], packets less than 19 octets long are
   never valid.

   An endpoint MAY send a stateless reset in response to a packet with a
   long header.  This would not be effective if the stateless reset
   token was not yet available to a peer.  In this QUIC version, packets
   with a long header are only used during connection establishment.
   Because the stateless reset token is not available until connection
   establishment is complete or near completion, ignoring an unknown
   packet with a long header might be more effective.



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   An endpoint cannot determine the Source Connection ID from a packet
   with a short header, therefore it cannot set the Destination
   Connection ID in the stateless reset packet.  The destination
   connection ID will therefore differ from the value used in previous
   packets.  A random Destination Connection ID makes the connection ID
   appear to be the result of moving to a new connection ID that was
   provided using a NEW_CONNECTION_ID frame (Section 7.13).

   Using a randomized connection ID results in two problems:

   o  The packet might not reach the peer.  If the Destination
      Connection ID is critical for routing toward the peer, then this
      packet could be incorrectly routed.  This might also trigger
      another Stateless Reset in response, see Section 6.13.4.3.  A
      Stateless Reset that is not correctly routed is ineffective in
      causing errors to be quickly detected and recovered.  In this
      case, endpoints will need to rely on other methods - such as
      timers - to detect that the connection has failed.

   o  The randomly generated connection ID can be used by entities other
      than the peer to identify this as a potential stateless reset.  An
      endpoint that occasionally uses different connection IDs might
      introduce some uncertainty about this.

   Finally, the last 16 octets of the packet are set to the value of the
   Stateless Reset Token.

   A stateless reset is not appropriate for signaling error conditions.
   An endpoint that wishes to communicate a fatal connection error MUST
   use a CONNECTION_CLOSE or APPLICATION_CLOSE frame if it has
   sufficient state to do so.

   This stateless reset design is specific to QUIC version 1.  An
   endpoint that supports multiple versions of QUIC needs to generate a
   stateless reset that will be accepted by peers that support any
   version that the endpoint might support (or might have supported
   prior to losing state).  Designers of new versions of QUIC need to be
   aware of this and either reuse this design, or use a portion of the
   packet other than the last 16 octets for carrying data.

6.13.4.1.  Detecting a Stateless Reset

   An endpoint detects a potential stateless reset when a packet with a
   short header either cannot be decrypted or is marked as a duplicate
   packet.  The endpoint then compares the last 16 octets of the packet
   with the Stateless Reset Token provided by its peer, either in a
   NEW_CONNECTION_ID frame or the server's transport parameters.  If
   these values are identical, the endpoint MUST enter the draining



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   period and not send any further packets on this connection.  If the
   comparison fails, the packet can be discarded.

6.13.4.2.  Calculating a Stateless Reset Token

   The stateless reset token MUST be difficult to guess.  In order to
   create a Stateless Reset Token, an endpoint could randomly generate
   [RFC4086] a secret for every connection that it creates.  However,
   this presents a coordination problem when there are multiple
   instances in a cluster or a storage problem for a endpoint that might
   lose state.  Stateless reset specifically exists to handle the case
   where state is lost, so this approach is suboptimal.

   A single static key can be used across all connections to the same
   endpoint by generating the proof using a second iteration of a
   preimage-resistant function that takes a static key and the
   connection ID chosen by the endpoint (see Section 6.1) as input.  An
   endpoint could use HMAC [RFC2104] (for example, HMAC(static_key,
   connection_id)) or HKDF [RFC5869] (for example, using the static key
   as input keying material, with the connection ID as salt).  The
   output of this function is truncated to 16 octets to produce the
   Stateless Reset Token for that connection.

   An endpoint that loses state can use the same method to generate a
   valid Stateless Reset Token.  The connection ID comes from the packet
   that the endpoint receives.

   This design relies on the peer always sending a connection ID in its
   packets so that the endpoint can use the connection ID from a packet
   to reset the connection.  An endpoint that uses this design MUST
   either use the same connection ID length for all connections or
   encode the length of the connection ID such that it can be recovered
   without state.  In addition, it MUST NOT provide a zero-length
   connection ID.

   Revealing the Stateless Reset Token allows any entity to terminate
   the connection, so a value can only be used once.  This method for
   choosing the Stateless Reset Token means that the combination of
   connection ID and static key cannot occur for another connection.  A
   denial of service attack is possible if the same connection ID is
   used by instances that share a static key, or if an attacker can
   cause a packet to be routed to an instance that has no state but the
   same static key (see Section 12.8).  A connection ID from a
   connection that is reset by revealing the Stateless Reset Token
   cannot be reused for new connections at nodes that share a static
   key.





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   Note that Stateless Reset packets do not have any cryptographic
   protection.

6.13.4.3.  Looping

   The design of a Stateless Reset is such that it is indistinguishable
   from a valid packet.  This means that a Stateless Reset might trigger
   the sending of a Stateless Reset in response, which could lead to
   infinite exchanges.

   An endpoint MUST ensure that every Stateless Reset that it sends is
   smaller than the packet triggered it, unless it maintains state
   sufficient to prevent looping.  In the event of a loop, this results
   in packets eventually being too small to trigger a response.

   An endpoint can remember the number of Stateless Reset packets that
   it has sent and stop generating new Stateless Reset packets once a
   limit is reached.  Using separate limits for different remote
   addresses will ensure that Stateless Reset packets can be used to
   close connections when other peers or connections have exhausted
   limits.

   Reducing the size of a Stateless Reset below the recommended minimum
   size of 37 octets could mean that the packet could reveal to an
   observer that it is a Stateless Reset.  Conversely, refusing to send
   a Stateless Reset in response to a small packet might result in
   Stateless Reset not being useful in detecting cases of broken
   connections where only very small packets are sent; such failures
   might only be detected by other means, such as timers.

7.  Frame Types and Formats

   As described in Section 5, packets contain one or more frames.  This
   section describes the format and semantics of the core QUIC frame
   types.

7.1.  Variable-Length Integer Encoding

   QUIC frames commonly use a variable-length encoding for non-negative
   integer values.  This encoding ensures that smaller integer values
   need fewer octets to encode.

   The QUIC variable-length integer encoding reserves the two most
   significant bits of the first octet to encode the base 2 logarithm of
   the integer encoding length in octets.  The integer value is encoded
   on the remaining bits, in network byte order.





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   This means that integers are encoded on 1, 2, 4, or 8 octets and can
   encode 6, 14, 30, or 62 bit values respectively.  Table 4 summarizes
   the encoding properties.

          +------+--------+-------------+-----------------------+
          | 2Bit | Length | Usable Bits | Range                 |
          +------+--------+-------------+-----------------------+
          | 00   | 1      | 6           | 0-63                  |
          |      |        |             |                       |
          | 01   | 2      | 14          | 0-16383               |
          |      |        |             |                       |
          | 10   | 4      | 30          | 0-1073741823          |
          |      |        |             |                       |
          | 11   | 8      | 62          | 0-4611686018427387903 |
          +------+--------+-------------+-----------------------+

                   Table 4: Summary of Integer Encodings

   For example, the eight octet sequence c2 19 7c 5e ff 14 e8 8c (in
   hexadecimal) decodes to the decimal value 151288809941952652; the
   four octet sequence 9d 7f 3e 7d decodes to 494878333; the two octet
   sequence 7b bd decodes to 15293; and the single octet 25 decodes to
   37 (as does the two octet sequence 40 25).

   Error codes (Section 11.3) are described using integers, but do not
   use this encoding.

7.2.  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.

7.3.  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 and
   retransmission of STREAM frames on the identified stream.  A receiver
   of RST_STREAM can discard any data that it already received on that
   stream.





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   An endpoint that receives a RST_STREAM frame for a send-only stream
   MUST terminate the connection with error PROTOCOL_VIOLATION.

   The RST_STREAM frame is as follows:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        Stream ID (i)                        ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |  Application Error Code (16)  |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                       Final Offset (i)                     ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   The fields are:

   Stream ID:  A variable-length integer encoding of the Stream ID of
      the stream being terminated.

   Application Protocol Error Code:  A 16-bit application protocol error
      code (see Section 11.4) which indicates why the stream is being
      closed.

   Final Offset:  A variable-length integer indicating the absolute byte
      offset of the end of data written on this stream by the RST_STREAM
      sender.

7.4.  CONNECTION_CLOSE frame

   An endpoint sends a CONNECTION_CLOSE frame (type=0x02) to notify its
   peer that the connection is being closed.  CONNECTION_CLOSE is used
   to signal errors at the QUIC layer, or the absence of errors (with
   the NO_ERROR code).

   If there are open streams that haven't been explicitly closed, they
   are implicitly closed when the connection is closed.

   The CONNECTION_CLOSE frame is as follows:












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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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           Error Code (16)     |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         Frame Type (i)                      ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                    Reason Phrase Length (i)                 ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        Reason Phrase (*)                    ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   The fields of a CONNECTION_CLOSE frame are as follows:

   Error Code:  A 16-bit error code which indicates the reason for
      closing this connection.  CONNECTION_CLOSE uses codes from the
      space defined in Section 11.3.

   Frame Type:  A variable-length integer encoding the type of frame
      that triggered the error.  A value of 0 (equivalent to the mention
      of the PADDING frame) is used when the frame type is unknown.

   Reason Phrase Length:  A variable-length integer specifying the
      length of the reason phrase in bytes.  Note that a
      CONNECTION_CLOSE frame cannot be split between packets, so in
      practice any limits on packet size will also limit the space
      available for a reason phrase.

   Reason Phrase:  A human-readable explanation for why the connection
      was closed.  This can be zero length if the sender chooses to not
      give details beyond the Error Code.  This SHOULD be a UTF-8
      encoded string [RFC3629].

7.5.  APPLICATION_CLOSE frame

   An APPLICATION_CLOSE frame (type=0x03) is used to signal an error
   with the protocol that uses QUIC.

   The APPLICATION_CLOSE frame is as follows:












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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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           Error Code (16)     |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                    Reason Phrase Length (i)                 ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        Reason Phrase (*)                    ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   The fields of a APPLICATION_CLOSE frame are as follows:

   Error Code:  A 16-bit error code which indicates the reason for
      closing this connection.  APPLICATION_CLOSE uses codes from the
      application protocol error code space, see Section 11.4.

   Reason Phrase Length:  This field is identical in format and
      semantics to the Reason Phrase Length field from CONNECTION_CLOSE.

   Reason Phrase:  This field is identical in format and semantics to
      the Reason Phrase field from CONNECTION_CLOSE.

   APPLICATION_CLOSE has similar format and semantics to the
   CONNECTION_CLOSE frame (Section 7.4).  Aside from the semantics of
   the Error Code field and the omission of the Frame Type field, both
   frames are used to close the connection.

7.6.  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 (i)                     ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   The fields in the MAX_DATA frame are as follows:

   Maximum Data:  A variable-length integer indicating the maximum
      amount of data that can be sent on the entire connection, in units
      of octets.





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   All data sent in STREAM frames counts toward this limit.  The sum of
   the largest received offsets on all streams - including streams in
   terminal states - MUST NOT exceed the value advertised by a receiver.
   An endpoint MUST terminate a connection with a
   QUIC_FLOW_CONTROL_RECEIVED_TOO_MUCH_DATA error if it receives more
   data than the maximum data value that it has sent, unless this is a
   result of a change in the initial limits (see Section 6.6.2).

7.7.  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.

   An endpoint that receives a MAX_STREAM_DATA frame for a receive-only
   stream MUST terminate the connection with error PROTOCOL_VIOLATION.

   An endpoint that receives a MAX_STREAM_DATA frame for a send-only
   stream it has not opened MUST terminate the connection with error
   PROTOCOL_VIOLATION.

   Note that an endpoint may legally receive a MAX_STREAM_DATA frame on
   a bidirectional stream it has not opened.

   The frame is as follows:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        Stream ID (i)                        ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                    Maximum Stream Data (i)                  ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   The fields in the MAX_STREAM_DATA frame are as follows:

   Stream ID:  The stream ID of the stream that is affected encoded as a
      variable-length integer.

   Maximum Stream Data:  A variable-length integer indicating the
      maximum amount of data that can be sent on the identified stream,
      in units of octets.

   When counting data toward this limit, an endpoint accounts for the
   largest received offset of data that is sent or received on the
   stream.  Loss or reordering can mean that the largest received offset
   on a stream can be greater than the total size of data received on




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   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 6.6.2).

7.8.  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 (i)                    ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   The fields in the MAX_STREAM_ID frame are as follows:

   Maximum Stream ID:  ID of the maximum unidirectional or bidirectional
      peer-initiated stream ID for the connection encoded as a variable-
      length integer.  The limit applies to unidirectional steams if the
      second least signification bit of the stream ID is 1, and applies
      to bidirectional streams if it is 0.

   Loss or reordering can mean that a MAX_STREAM_ID frame can be
   received which states a lower stream limit than the client has
   previously received.  MAX_STREAM_ID frames which do not increase the
   maximum stream ID MUST be ignored.

   A peer MUST NOT initiate a stream with a higher stream ID than the
   greatest maximum stream ID it has received.  An endpoint MUST
   terminate a connection with a STREAM_ID_ERROR error if a peer
   initiates a stream with a higher stream ID than it has sent, unless
   this is a result of a change in the initial limits (see
   Section 6.6.2).

7.9.  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.



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   The receiver of a PING frame simply needs to acknowledge the packet
   containing this frame.

   The PING frame can be used to keep a connection alive when an
   application or application protocol wishes to prevent the connection
   from timing out.  An application protocol SHOULD provide guidance
   about the conditions under which generating a PING is recommended.
   This guidance SHOULD indicate whether it is the client or the server
   that is expected to send the PING.  Having both endpoints send PING
   frames without coordination can produce an excessive number of
   packets and poor performance.

   A connection will time out if no packets are sent or received for a
   period longer than the time specified in the idle_timeout transport
   parameter (see Section 6.13).  However, state in middleboxes might
   time out earlier than that.  Though REQ-5 in [RFC4787] recommends a 2
   minute timeout interval, experience shows that sending packets every
   15 to 30 seconds is necessary to prevent the majority of middleboxes
   from losing state for UDP flows.

7.10.  BLOCKED Frame

   A sender SHOULD send a BLOCKED frame (type=0x08) when it wishes to
   send data, but is unable to due to connection-level flow control (see
   Section 10.2.1).  BLOCKED frames can be used as input to tuning of
   flow control algorithms (see Section 10.1.2).

   The BLOCKED frame is as follows:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         Offset (i)                         ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   The BLOCKED frame contains a single field.

   Offset:  A variable-length integer indicating the connection-level
      offset at which the blocking occurred.

7.11.  STREAM_BLOCKED Frame

   A sender SHOULD send a STREAM_BLOCKED frame (type=0x09) when it
   wishes to send data, but is unable to due to stream-level flow
   control.  This frame is analogous to BLOCKED (Section 7.10).

   An endpoint that receives a STREAM_BLOCKED frame for a send-only
   stream MUST terminate the connection with error PROTOCOL_VIOLATION.



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   The STREAM_BLOCKED frame is as follows:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        Stream ID (i)                        ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         Offset (i)                          ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   The STREAM_BLOCKED frame contains two fields:

   Stream ID:  A variable-length integer indicating the stream which is
      flow control blocked.

   Offset:  A variable-length integer indicating the offset of the
      stream at which the blocking occurred.

7.12.  STREAM_ID_BLOCKED Frame

   A sender MAY send a STREAM_ID_BLOCKED frame (type=0x0a) when it
   wishes to open a stream, but is unable to due to the maximum stream
   ID limit set by its peer (see Section 7.8).  This does not open the
   stream, but informs the peer that a new stream was needed, but the
   stream limit prevented the creation of the stream.

   The STREAM_ID_BLOCKED frame is as follows:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        Stream ID (i)                        ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   The STREAM_ID_BLOCKED frame contains a single field.

   Stream ID:  A variable-length integer indicating the highest stream
      ID that the sender was permitted to open.

7.13.  NEW_CONNECTION_ID Frame

   An endpoint sends a NEW_CONNECTION_ID frame (type=0x0b) to provide
   its peer with alternative connection IDs that can be used to break
   linkability when migrating connections (see Section 6.11.5).

   The NEW_CONNECTION_ID 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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                          Sequence (i)                       ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   Length (8)  |          Connection ID (32..144)            ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +                                                               +
   |                                                               |
   +                   Stateless Reset Token (128)                 +
   |                                                               |
   +                                                               +
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   The fields are:

   Sequence:  A variable-length integer.  This value starts at 0 and
      increases by 1 for each connection ID that is provided by the
      server.  The connection ID that is assigned during the handshake
      is assumed to have a sequence of -1.  That is, the value selected
      during the handshake comes immediately before the first value that
      a server can send.

   Length:  An 8-bit unsigned integer containing the length of the
      connection ID.  Values less than 4 and greater than 18 are invalid
      and MUST be treated as a connection error of type
      PROTOCOL_VIOLATION.

   Connection ID:  A connection ID of the specified length.

   Stateless Reset Token:  A 128-bit value that will be used to for a
      stateless reset when the associated connection ID is used (see
      Section 6.13.4).

   An endpoint MUST NOT send this frame if it currently requires that
   its peer send packets with a zero-length Destination Connection ID.
   Changing the length of a connection ID to or from zero-length makes
   it difficult to identify when the value of the connection ID changed.
   An endpoint that is sending packets with a zero-length Destination
   Connection ID MUST treat receipt of a NEW_CONNECTION_ID frame as a
   connection error of type PROTOCOL_VIOLATION.








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7.14.  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.

   Receipt of a STOP_SENDING frame is only valid for a send stream that
   exists and is not in the "Ready" state (see Section 9.2.1).
   Receiving a STOP_SENDING frame for a send stream that is "Ready" or
   non-existent MUST be treated as a connection error of type
   PROTOCOL_VIOLATION.  An endpoint that receives a STOP_SENDING frame
   for a receive-only stream MUST terminate the connection with error
   PROTOCOL_VIOLATION.

   The STOP_SENDING frame is as follows:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        Stream ID (i)                        ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |  Application Error Code (16)  |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   The fields are:

   Stream ID:  A variable-length integer carrying the Stream ID of the
      stream being ignored.

   Application Error Code:  A 16-bit, application-specified reason the
      sender is ignoring the stream (see Section 11.4).

7.15.  ACK Frame

   Receivers send ACK frames (type=0x0d) to inform senders which packets
   they have received and processed.  The ACK frame contains any number
   of ACK blocks.  ACK blocks are ranges of acknowledged packets.

   QUIC acknowledgements are irrevocable.  Once acknowledged, a packet
   remains acknowledged, even if it does not appear in a future ACK
   frame.  This is unlike TCP SACKs ([RFC2018]).

   It is expected that a sender will reuse the same packet number across
   different packet number spaces.  ACK frames only acknowledge the
   packet numbers that were transmitted by the sender in the same packet
   number space of the packet that the ACK was received in.




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   A client MUST NOT acknowledge Retry packets.  Retry packets include
   the packet number from the Initial packet it responds to.  Version
   Negotiation packets cannot be acknowledged because they do not
   contain a packet number.  Rather than relying on ACK frames, these
   packets are implicitly acknowledged by the next Initial packet sent
   by the client.

   An ACK frame is shown below.

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                     Largest Acknowledged (i)                ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                          ACK Delay (i)                      ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                       ACK Block Count (i)                   ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                          ACK Blocks (*)                     ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                        Figure 12: ACK Frame Format

   The fields in the ACK frame are as follows:

   Largest Acknowledged:  A variable-length integer representing the
      largest packet number the peer is acknowledging; this is usually
      the largest packet number that the peer has received prior to
      generating the ACK frame.  Unlike the packet number in the QUIC
      long or short header, the value in an ACK frame is not truncated.

   ACK Delay:  A variable-length integer including the time in
      microseconds that the largest acknowledged packet, as indicated in
      the Largest Acknowledged field, was received by this peer to when
      this ACK was sent.  The value of the ACK Delay field is scaled by
      multiplying the encoded value by the 2 to the power of the value
      of the "ack_delay_exponent" transport parameter set by the sender
      of the ACK frame.  The "ack_delay_exponent" defaults to 3, or a
      multiplier of 8 (see Section 6.6.1).  Scaling in this fashion
      allows for a larger range of values with a shorter encoding at the
      cost of lower resolution.

   ACK Block Count:  A variable-length integer specifying the number of
      Additional ACK Block (and Gap) fields after the First ACK Block.

   ACK Blocks:  Contains one or more blocks of packet numbers which have
      been successfully received, see Section 7.15.1.




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7.15.1.  ACK Block Section

   The ACK Block Section consists of alternating Gap and ACK Block
   fields in descending packet number order.  A First Ack Block field is
   followed by a variable number of alternating Gap and Additional ACK
   Blocks.  The number of Gap and Additional ACK Block fields is
   determined by the ACK Block Count field.

   Gap and ACK Block fields use a relative integer encoding for
   efficiency.  Though each encoded value is positive, the values are
   subtracted, so that each ACK Block describes progressively lower-
   numbered packets.  As long as contiguous ranges of packets are small,
   the variable-length integer encoding ensures that each range can be
   expressed in a small number of octets.

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                      First ACK Block (i)                    ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                             Gap (i)                         ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                    Additional ACK Block (i)                 ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                             Gap (i)                         ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                    Additional ACK Block (i)                 ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
                                  ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                             Gap (i)                         ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                    Additional ACK Block (i)                 ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                       Figure 13: ACK Block Section

   Each ACK Block acknowledges a contiguous range of packets by
   indicating the number of acknowledged packets that precede the
   largest packet number in that block.  A value of zero indicates that
   only the largest packet number is acknowledged.  Larger ACK Block
   values indicate a larger range, with corresponding lower values for
   the smallest packet number in the range.  Thus, given a largest
   packet number for the ACK, the smallest value is determined by the
   formula:

      smallest = largest - ack_block




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   The range of packets that are acknowledged by the ACK block include
   the range from the smallest packet number to the largest, inclusive.

   The largest value for the First ACK Block is determined by the
   Largest Acknowledged field; the largest for Additional ACK Blocks is
   determined by cumulatively subtracting the size of all preceding ACK
   Blocks and Gaps.

   Each Gap indicates a range of packets that are not being
   acknowledged.  The number of packets in the gap is one higher than
   the encoded value of the Gap Field.

   The value of the Gap field establishes the largest packet number
   value for the ACK block that follows the gap using the following
   formula:

     largest = previous_smallest - gap - 2

   If the calculated value for largest or smallest packet number for any
   ACK Block is negative, an endpoint MUST generate a connection error
   of type FRAME_ENCODING_ERROR indicating an error in an ACK frame.

   The fields in the ACK Block Section are:

   First ACK Block:  A variable-length integer indicating the number of
      contiguous packets preceding the Largest Acknowledged that are
      being acknowledged.

   Gap (repeated):  A variable-length integer indicating the number of
      contiguous unacknowledged packets preceding the packet number one
      lower than the smallest in the preceding ACK Block.

   ACK Block (repeated):  A variable-length integer indicating the
      number of contiguous acknowledged packets preceding the largest
      packet number, as determined by the preceding Gap.

7.15.2.  Sending ACK Frames

   Implementations MUST NOT generate packets that only contain ACK
   frames in response to packets which only contain ACK frames.
   However, they MUST acknowledge packets containing only ACK frames
   when sending ACK frames in response to other packets.
   Implementations MUST NOT send more than one packet containing only
   ACK frames per received packet that contains frames other than ACK
   frames.  Packets containing non-ACK frames MUST be acknowledged
   immediately or when a delayed ack timer expires.





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   To limit ACK blocks to those that have not yet been received by the
   sender, the receiver SHOULD track which ACK frames have been
   acknowledged by its peer.  Once an ACK frame has been acknowledged,
   the packets it acknowledges SHOULD NOT be acknowledged again.

   Because ACK frames are not sent in response to ACK-only packets, a
   receiver that is only sending ACK frames will only receive
   acknowledgements for its packets if the sender includes them in
   packets with non-ACK frames.  A sender SHOULD bundle ACK frames with
   other frames when possible.

   Endpoints can only acknowledge packets sent in a particular packet
   number space by sending ACK frames in packets from the same packet
   number space.

   To limit receiver state or the size of ACK frames, a receiver MAY
   limit the number of ACK blocks it sends.  A receiver can do this even
   without receiving acknowledgment of its ACK frames, with the
   knowledge this could cause the sender to unnecessarily retransmit
   some data.  Standard QUIC [QUIC-RECOVERY] algorithms declare packets
   lost after sufficiently newer packets are acknowledged.  Therefore,
   the receiver SHOULD repeatedly acknowledge newly received packets in
   preference to packets received in the past.

7.15.3.  ACK Frames and Packet Protection

   ACK frames MUST only be carried in a packet that has the same packet
   number space as the packet being ACKed (see Section 4.8).  For
   instance, packets that are protected with 1-RTT keys MUST be
   acknowledged in packets that are also protected with 1-RTT keys.

   Packets that a client sends with 0-RTT packet protection MUST be
   acknowledged by the server in packets protected by 1-RTT keys.  This
   can mean that the client is unable to use these acknowledgments if
   the server cryptographic handshake messages are delayed or lost.
   Note that the same limitation applies to other data sent by the
   server protected by the 1-RTT keys.

   Endpoints SHOULD send acknowledgments for packets containing CRYPTO
   frames with a reduced delay; see Section 4.3.1 of [QUIC-RECOVERY].

7.16.  ACK_ECN Frame

   The ACK_ECN frame (type=0x1a) is used by an endpoint that supports
   ECN to acknowledge packets received with ECN codepoints of ECT(0),
   ECT(1), or CE in the packet's IP header.





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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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                     Largest Acknowledged (i)                ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                          ACK Delay (i)                      ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        ECT(0) Count (i)                     ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        ECT(1) Count (i)                     ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        ECN-CE Count (i)                     ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                       ACK Block Count (i)                   ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                          ACK Blocks (*)                     ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                      Figure 14: ACK_ECN Frame Format

   An ACK_ECN frame contains all the elements of the ACK frame
   (Section 7.15) with the addition of three counts following the ACK
   Delay field.

   ECT(0) Count:  A variable-length integer representing the total
      number packets received with the ECT(0) codepoint.

   ECT(1) Count:  A variable-length integer representing the total
      number packets received with the ECT(1) codepoint.

   CE Count:  A variable-length integer representing the total number
      packets received with the CE codepoint.

7.17.  PATH_CHALLENGE Frame

   Endpoints can use PATH_CHALLENGE frames (type=0x0e) to check
   reachability to the peer and for path validation during connection
   establishment and connection migration.

   PATH_CHALLENGE frames contain an 8-byte payload.

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +                            Data (8)                           +
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+



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   Data:  This 8-byte field contains arbitrary data.

   A PATH_CHALLENGE frame containing 8 octets that are hard to guess is
   sufficient to ensure that it is easier to receive the packet than it
   is to guess the value correctly.

   The recipient of this frame MUST generate a PATH_RESPONSE frame
   (Section 7.18) containing the same Data.

7.18.  PATH_RESPONSE Frame

   The PATH_RESPONSE frame (type=0x0f) is sent in response to a
   PATH_CHALLENGE frame.  Its format is identical to the PATH_CHALLENGE
   frame (Section 7.17).

   If the content of a PATH_RESPONSE frame does not match the content of
   a PATH_CHALLENGE frame previously sent by the endpoint, the endpoint
   MAY generate a connection error of type PROTOCOL_VIOLATION.

7.19.  NEW_TOKEN frame

   A server sends a NEW_TOKEN frame (type=0x19) to provide the client a
   token to send in the header of an Initial packet for a future
   connection.

   The NEW_TOKEN frame is as follows:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                     Token Length (i)  ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                            Token (*)                        ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   The fields of a NEW_TOKEN frame are as follows:

   Token Length:  A variable-length integer specifying the length of the
      token in bytes.

   Token:  An opaque blob that the client may use with a future Initial
      packet.

7.20.  STREAM Frames

   STREAM frames implicitly create a stream and carry stream data.  The
   STREAM frame takes the form 0b00010XXX (or the set of values from




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   0x10 to 0x17).  The value of the three low-order bits of the frame
   type determine the fields that are present in the frame.

   o  The OFF bit (0x04) in the frame type is set to indicate that there
      is an Offset field present.  When set to 1, the Offset field is
      present; when set to 0, the Offset field is absent and the Stream
      Data starts at an offset of 0 (that is, the frame contains the
      first octets of the stream, or the end of a stream that includes
      no data).

   o  The LEN bit (0x02) in the frame type is set to indicate that there
      is a Length field present.  If this bit is set to 0, the Length
      field is absent and the Stream Data field extends to the end of
      the packet.  If this bit is set to 1, the Length field is present.

   o  The FIN bit (0x01) of the frame type is set only on frames that
      contain the final offset of the stream.  Setting this bit
      indicates that the frame marks the end of the stream.

   An endpoint that receives a STREAM frame for a send-only stream MUST
   terminate the connection with error PROTOCOL_VIOLATION.

   A STREAM frame is shown below.

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         Stream ID (i)                       ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         [Offset (i)]                        ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         [Length (i)]                        ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        Stream Data (*)                      ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                      Figure 15: STREAM Frame Format

   The STREAM frame contains the following fields:

   Stream ID:  A variable-length integer indicating the stream ID of the
      stream (see Section 9.1).

   Offset:  A variable-length integer specifying the byte offset in the
      stream for the data in this STREAM frame.  This field is present
      when the OFF bit is set to 1.  When the Offset field is absent,
      the offset is 0.




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   Length:  A variable-length integer specifying the length of the
      Stream Data field in this STREAM frame.  This field is present
      when the LEN bit is set to 1.  When the LEN bit is set to 0, the
      Stream Data field consumes all the remaining octets in the packet.

   Stream Data:  The bytes from the designated stream to be delivered.

   When a Stream Data field has a length of 0, the offset in the STREAM
   frame is the offset of the next byte that would be sent.

   The first byte in the stream has an offset of 0.  The largest offset
   delivered on a stream - the sum of the re-constructed offset and data
   length - MUST be less than 2^62.

   Stream multiplexing is achieved by interleaving STREAM frames from
   multiple streams into one or more QUIC packets.  A single QUIC packet
   can include multiple STREAM frames from one or more streams.

   Implementation note: One of the benefits of QUIC is avoidance of
   head-of-line blocking across multiple streams.  When a packet loss
   occurs, only streams with data in that packet are blocked waiting for
   a retransmission to be received, while other streams can continue
   making progress.  Note that when data from multiple streams is
   bundled into a single QUIC packet, loss of that packet blocks all
   those streams from making progress.  An implementation is therefore
   advised to bundle as few streams as necessary in outgoing packets
   without losing transmission efficiency to underfilled packets.

7.21.  CRYPTO Frame

   The CRYPTO frame (type=0x18) is used to transmit cryptographic
   handshake messages.  It can be sent in all packet types.  The CRYPTO
   frame offers the cryptographic protocol an in-order stream of bytes.
   CRYPTO frames are functionally identical to STREAM frames, except
   that they do not bear a stream identifier; they are not flow
   controlled; and they do not carry markers for optional offset,
   optional length, and the end of the stream.

   A CRYPTO frame is shown below.












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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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                          Offset (i)                         ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                          Length (i)                         ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        Crypto Data (*)                      ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                      Figure 16: CRYPTO Frame Format

   The CRYPTO frame contains the following fields:

   Offset:  A variable-length integer specifying the byte offset in the
      stream for the data in this CRYPTO frame.

   Length:  A variable-length integer specifying the length of the
      Crypto Data field in this CRYPTO frame.

   Crypto Data:  The cryptographic message data.

   There is a separate flow of cryptographic handshake data in each
   encryption level, each of which starts at an offset of 0.  This
   implies that each encryption level is treated as a separate CRYPTO
   stream of data.

   Unlike STREAM frames, which include a Stream ID indicating to which
   stream the data belongs, the CRYPTO frame carries data for a single
   stream per encryption level.  The stream does not have an explicit
   end, so CRYPTO frames do not have a FIN bit.

8.  Packetization and Reliability

   A sender bundles one or more frames in a QUIC packet (see Section 5).

   A sender SHOULD minimize per-packet bandwidth and computational costs
   by bundling as many frames as possible within a QUIC packet.  A
   sender MAY wait for a short period of time to bundle multiple frames
   before sending a packet that is not maximally packed, to avoid
   sending out large numbers of small packets.  An implementation may
   use knowledge about application sending behavior or heuristics to
   determine whether and for how long to wait.  This waiting period is
   an implementation decision, and an implementation should be careful
   to delay conservatively, since any delay is likely to increase
   application-visible latency.





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8.1.  Packet Processing and Acknowledgment

   A packet MUST NOT be acknowledged until packet protection has been
   successfully removed and all frames contained in the packet have been
   processed.  Any stream state transitions triggered by the frame MUST
   have occurred.  For STREAM frames, this means the data has been
   enqueued in preparation to be received by the application protocol,
   but it does not require that data is delivered and consumed.

   Once the packet has been fully processed, a receiver acknowledges
   receipt by sending one or more ACK frames containing the packet
   number of the received packet.  To avoid creating an indefinite
   feedback loop, an endpoint MUST NOT send an ACK frame in response to
   a packet containing only ACK or PADDING frames, even if there are
   packet gaps which precede the received packet.  The endpoint MUST
   acknowledge packets containing only ACK or PADDING frames in the next
   ACK frame that it sends.

   While PADDING frames do not elicit an ACK frame from a receiver, they
   are considered to be in flight for congestion control purposes
   [QUIC-RECOVERY].  Sending only PADDING frames might cause the sender
   to become limited by the congestion controller (as described in
   [QUIC-RECOVERY]) with no acknowledgments forthcoming from the
   receiver.  Therefore, a sender should ensure that other frames are
   sent in addition to PADDING frames to elicit acknowledgments from the
   receiver.

   Strategies and implications of the frequency of generating
   acknowledgments are discussed in more detail in [QUIC-RECOVERY].

8.2.  Retransmission of Information

   QUIC packets that are determined to be lost are not retransmitted
   whole.  The same applies to the frames that are contained within lost
   packets.  Instead, the information that might be carried in frames is
   sent again in new frames as needed.

   New frames and packets are used to carry information that is
   determined to have been lost.  In general, information is sent again
   when a packet containing that information is determined to be lost
   and sending ceases when a packet containing that information is
   acknowledged.

   o  Data sent in CRYPTO frames are retransmitted according to the
      rules in [QUIC-RECOVERY], until either all data has been
      acknowledged or the crypto state machine implicitly knows that the
      peer received the data.




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   o  Application data sent in STREAM frames is retransmitted in new
      STREAM frames unless the endpoint has sent a RST_STREAM for that
      stream.  Once an endpoint sends a RST_STREAM frame, no further
      STREAM frames are needed.

   o  The most recent set of acknowledgments are sent in ACK frames.  An
      ACK frame SHOULD contain all unacknowledged acknowledgments, as
      described in Section 7.15.2.

   o  Cancellation of stream transmission, as carried in a RST_STREAM
      frame, is sent until acknowledged or until all stream data is
      acknowledged by the peer (that is, either the "Reset Recvd" or
      "Data Recvd" state is reached on the send stream).  The content of
      a RST_STREAM frame MUST NOT change when it is sent again.

   o  Similarly, a request to cancel stream transmission, as encoded in
      a STOP_SENDING frame, is sent until the receive stream enters
      either a "Data Recvd" or "Reset Recvd" state, see Section 9.3.

   o  Connection close signals, including those that use
      CONNECTION_CLOSE and APPLICATION_CLOSE frames, are not sent again
      when packet loss is detected, but as described in Section 6.13.

   o  The current connection maximum data is sent in MAX_DATA frames.
      An updated value is sent in a MAX_DATA frame if the packet
      containing the most recently sent MAX_DATA frame is declared lost,
      or when the endpoint decides to update the limit.  Care is
      necessary to avoid sending this frame too often as the limit can
      increase frequently and cause an unnecessarily large number of
      MAX_DATA frames to be sent.

   o  The current maximum stream data offset is sent in MAX_STREAM_DATA
      frames.  Like MAX_DATA, an updated value is sent when the packet
      containing the most recent MAX_STREAM_DATA frame for a stream is
      lost or when the limit is updated, with care taken to prevent the
      frame from being sent too often.  An endpoint SHOULD stop sending
      MAX_STREAM_DATA frames when the receive stream enters a "Size
      Known" state.

   o  The maximum stream ID for a stream of a given type is sent in
      MAX_STREAM_ID frames.  Like MAX_DATA, an updated value is sent
      when a packet containing the most recent MAX_STREAM_ID for a
      stream type frame is declared lost or when the limit is updated,
      with care taken to prevent the frame from being sent too often.

   o  Blocked signals are carried in BLOCKED, STREAM_BLOCKED, and
      STREAM_ID_BLOCKED frames.  BLOCKED streams have connection scope,
      STREAM_BLOCKED frames have stream scope, and STREAM_ID_BLOCKED



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      frames are scoped to a specific stream type.  New frames are sent
      if packets containing the most recent frame for a scope is lost,
      but only while the endpoint is blocked on the corresponding limit.
      These frames always include the limit that is causing blocking at
      the time that they are transmitted.

   o  A liveness or path validation check using PATH_CHALLENGE frames is
      sent periodically until a matching PATH_RESPONSE frame is received
      or until there is no remaining need for liveness or path
      validation checking.  PATH_CHALLENGE frames include a different
      payload each time they are sent.

   o  Responses to path validation using PATH_RESPONSE frames are sent
      just once.  A new PATH_CHALLENGE frame will be sent if another
      PATH_RESPONSE frame is needed.

   o  New connection IDs are sent in NEW_CONNECTION_ID frames and
      retransmitted if the packet containing them is lost.

   o  PADDING frames contain no information, so lost PADDING frames do
      not require repair.

   Upon detecting losses, a sender MUST take appropriate congestion
   control action.  The details of loss detection and congestion control
   are described in [QUIC-RECOVERY].

8.3.  Packet Size

   The QUIC packet size includes the QUIC header and integrity check,
   but not the UDP or IP header.

   Clients MUST ensure that the first Initial packet it sends is sent in
   a UDP datagram that is at least 1200 octets.  Padding the Initial
   packet or including a 0-RTT packet in the same datagram are ways to
   meet this requirement.  Sending a UDP datagram of this size ensures
   that the network path supports a reasonable Maximum Transmission Unit
   (MTU), and helps reduce the amplitude of amplification attacks caused
   by server responses toward an unverified client address.

   The datagram containing the first Initial packet from a client MAY
   exceed 1200 octets if the client believes that the Path Maximum
   Transmission Unit (PMTU) supports the size that it chooses.

   A server MAY send a CONNECTION_CLOSE frame with error code
   PROTOCOL_VIOLATION in response to the first Initial packet it
   receives from a client if the UDP datagram is smaller than 1200
   octets.  It MUST NOT send any other frame type in response, or




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   otherwise behave as if any part of the offending packet was processed
   as valid.

8.4.  Path Maximum Transmission Unit

   The Path Maximum Transmission Unit (PMTU) is the maximum size of the
   entire IP header, UDP header, and UDP payload.  The UDP payload
   includes the QUIC packet header, protected payload, and any
   authentication fields.

   All QUIC packets SHOULD be sized to fit within the estimated PMTU to
   avoid IP fragmentation or packet drops.  To optimize bandwidth
   efficiency, endpoints SHOULD use Packetization Layer PMTU Discovery
   ([PLPMTUD]).  Endpoints MAY use PMTU Discovery ([PMTUDv4], [PMTUDv6])
   for detecting the PMTU, setting the PMTU appropriately, and storing
   the result of previous PMTU determinations.

   In the absence of these mechanisms, QUIC endpoints SHOULD NOT send IP
   packets larger than 1280 octets.  Assuming the minimum IP header
   size, this results in a QUIC packet size of 1232 octets for IPv6 and
   1252 octets for IPv4.  Some QUIC implementations MAY wish to be more
   conservative in computing allowed QUIC packet size given unknown
   tunneling overheads or IP header options.

   QUIC endpoints that implement any kind of PMTU discovery SHOULD
   maintain an estimate for each combination of local and remote IP
   addresses.  Each pairing of local and remote addresses could have a
   different maximum MTU in the path.

   QUIC depends on the network path supporting a MTU of at least 1280
   octets.  This is the IPv6 minimum MTU and therefore also supported by
   most modern IPv4 networks.  An endpoint MUST NOT reduce its MTU below
   this number, even if it receives signals that indicate a smaller
   limit might exist.

   If a QUIC endpoint determines that the PMTU between any pair of local
   and remote IP addresses has fallen below 1280 octets, it MUST
   immediately cease sending QUIC packets on the affected path.  This
   could result in termination of the connection if an alternative path
   cannot be found.

8.4.1.  IPv4 PMTU Discovery

   Traditional ICMP-based path MTU discovery in IPv4 [PMTUDv4] is
   potentially vulnerable to off-path attacks that successfully guess
   the IP/port 4-tuple and reduce the MTU to a bandwidth-inefficient
   value.  TCP connections mitigate this risk by using the (at minimum)
   8 bytes of transport header echoed in the ICMP message to validate



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   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.

8.4.2.  Special Considerations for Packetization Layer PMTU Discovery

   The PADDING frame provides a useful option for PMTU probe packets.
   PADDING frames generate acknowledgements, but they need not be
   delivered reliably.  As a result, the loss of PADDING frames in probe
   packets does not require delay-inducing retransmission.  However,
   PADDING frames do consume congestion window, which may delay the
   transmission of subsequent application data.

   When implementing the algorithm in Section 7.2 of [PLPMTUD], the
   initial value of search_low SHOULD be consistent with the IPv6
   minimum packet size.  Paths that do not support this size cannot
   deliver Initial packets, and therefore are not QUIC-compliant.

   Section 7.3 of [PLPMTUD] discusses trade-offs between small and large
   increases in the size of probe packets.  As QUIC probe packets need
   not contain application data, aggressive increases in probe size
   carry fewer consequences.

9.  Streams: QUIC's Data Structuring Abstraction

   Streams in QUIC provide a lightweight, ordered byte-stream
   abstraction.

   There are two basic types of stream in QUIC.  Unidirectional streams
   carry data in one direction only; bidirectional streams allow for
   data to be sent in both directions.  Different stream identifiers are



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   used to distinguish between unidirectional and bidirectional streams,
   as well as to create a separation between streams that are initiated
   by the client and server (see Section 9.1).

   Either type of stream can be created by either endpoint, can
   concurrently send data interleaved with other streams, and can be
   cancelled.

   Stream offsets allow for the octets on a stream to be placed in
   order.  An endpoint MUST be capable of delivering data received on a
   stream in order.  Implementations MAY choose to offer the ability to
   deliver data out of order.  There is no means of ensuring ordering
   between octets on different streams.

   The creation and destruction of streams are expected to have minimal
   bandwidth and computational cost.  A single STREAM frame may create,
   carry data for, and terminate a stream, or a stream may last the
   entire duration of a connection.

   Streams are individually flow controlled, allowing an endpoint to
   limit memory commitment and to apply back pressure.  The creation of
   streams is also flow controlled, with each peer declaring the maximum
   stream ID it is willing to accept at a given time.

   An alternative view of QUIC streams is as an elastic "message"
   abstraction, similar to the way ephemeral streams are used in SST
   [SST], which may be a more appealing description for some
   applications.

9.1.  Stream Identifiers

   Streams are identified by an unsigned 62-bit integer, referred to as
   the Stream ID.  The least significant two bits of the Stream ID are
   used to identify the type of stream (unidirectional or bidirectional)
   and the initiator of the stream.

   The least significant bit (0x1) of the Stream ID identifies the
   initiator of the stream.  Clients initiate even-numbered streams
   (those with the least significant bit set to 0); servers initiate
   odd-numbered streams (with the bit set to 1).  Separation of the
   stream identifiers ensures that client and server are able to open
   streams without the latency imposed by negotiating for an identifier.

   If an endpoint receives a frame for a stream that it expects to
   initiate (i.e., odd-numbered for the client or even-numbered for the
   server), but which it has not yet opened, it MUST close the
   connection with error code STREAM_STATE_ERROR.




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   The second least significant bit (0x2) of the Stream ID
   differentiates between unidirectional streams and bidirectional
   streams.  Unidirectional streams always have this bit set to 1 and
   bidirectional streams have this bit set to 0.

   The two type bits from a Stream ID therefore identify streams as
   summarized in Table 5.

              +----------+----------------------------------+
              | Low Bits | Stream Type                      |
              +----------+----------------------------------+
              | 0x0      | Client-Initiated, Bidirectional  |
              |          |                                  |
              | 0x1      | Server-Initiated, Bidirectional  |
              |          |                                  |
              | 0x2      | Client-Initiated, Unidirectional |
              |          |                                  |
              | 0x3      | Server-Initiated, Unidirectional |
              +----------+----------------------------------+

                         Table 5: Stream ID Types

   The first bi-directional stream opened by the client is stream 0.

   A QUIC endpoint MUST NOT reuse a Stream ID.  Streams of each type are
   created in numeric order.  Streams that are used out of order result
   in opening all lower-numbered streams of the same type in the same
   direction.

   Stream IDs are encoded as a variable-length integer (see
   Section 7.1).

9.2.  Stream States

   This section describes the two types of QUIC stream in terms of the
   states of their send or receive components.  Two state machines are
   described: one for streams on which an endpoint transmits data
   (Section 9.2.1); another for streams from which an endpoint receives
   data (Section 9.2.2).

   Unidirectional streams use the applicable state machine directly.
   Bidirectional streams use both state machines.  For the most part,
   the use of these state machines is the same whether the stream is
   unidirectional or bidirectional.  The conditions for opening a stream
   are slightly more complex for a bidirectional stream because the
   opening of either send or receive sides causes the stream to open in
   both directions.




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   An endpoint can open streams up to its maximum stream limit in any
   order, however endpoints SHOULD open the send side of streams for
   each type in order.

   Note:  These states are largely informative.  This document uses
      stream states to describe rules for when and how different types
      of frames can be sent and the reactions that are expected when
      different types of frames are received.  Though these state
      machines are intended to be useful in implementing QUIC, these
      states aren't intended to constrain implementations.  An
      implementation can define a different state machine as long as its
      behavior is consistent with an implementation that implements
      these states.

9.2.1.  Send Stream States

   Figure 17 shows the states for the part of a stream that sends data
   to a peer.

































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          o
          | Create Stream (Sending)
          | Create Bidirectional Stream (Receiving)
          v
      +-------+
      | Ready | Send RST_STREAM
      |       |-----------------------.
      +-------+                       |
          |                           |
          | Send STREAM /             |
          |      STREAM_BLOCKED       |
          v                           |
      +-------+                       |
      | Send  | Send RST_STREAM       |
      |       |---------------------->|
      +-------+                       |
          |                           |
          | Send STREAM + FIN         |
          v                           v
      +-------+                   +-------+
      | Data  | Send RST_STREAM   | Reset |
      | Sent  +------------------>| Sent  |
      +-------+                   +-------+
          |                           |
          | Recv All ACKs             | Recv ACK
          v                           v
      +-------+                   +-------+
      | Data  |                   | Reset |
      | Recvd |                   | Recvd |
      +-------+                   +-------+

                    Figure 17: States for Send Streams

   The sending part of stream that the endpoint initiates (types 0 and 2
   for clients, 1 and 3 for servers) is opened by the application or
   application protocol.  The "Ready" state represents a newly created
   stream that is able to accept data from the application.  Stream data
   might be buffered in this state in preparation for sending.

   The sending part of a bidirectional stream initiated by a peer (type
   0 for a server, type 1 for a client) enters the "Ready" state if the
   receiving part enters the "Recv" state.

   Sending the first STREAM or STREAM_BLOCKED frame causes a send stream
   to enter the "Send" state.  An implementation might choose to defer
   allocating a Stream ID to a send stream until it sends the first
   frame and enters this state, which can allow for better stream
   prioritization.



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   In the "Send" state, an endpoint transmits - and retransmits as
   necessary - data in STREAM frames.  The endpoint respects the flow
   control limits of its peer, accepting MAX_STREAM_DATA frames.  An
   endpoint in the "Send" state generates STREAM_BLOCKED frames if it
   encounters flow control limits.

   After the application indicates that stream data is complete and a
   STREAM frame containing the FIN bit is sent, the send stream enters
   the "Data Sent" state.  From this state, the endpoint only
   retransmits stream data as necessary.  The endpoint no longer needs
   to track flow control limits or send STREAM_BLOCKED frames for a send
   stream in this state.  The endpoint can ignore any MAX_STREAM_DATA
   frames it receives from its peer in this state; MAX_STREAM_DATA
   frames might be received until the peer receives the final stream
   offset.

   Once all stream data has been successfully acknowledged, the send
   stream enters the "Data Recvd" state, which is a terminal state.

   From any of the "Ready", "Send", or "Data Sent" states, an
   application can signal that it wishes to abandon transmission of
   stream data.  Similarly, the endpoint might receive a STOP_SENDING
   frame from its peer.  In either case, the endpoint sends a RST_STREAM
   frame, which causes the stream to enter the "Reset Sent" state.

   An endpoint MAY send a RST_STREAM as the first frame on a send
   stream; this causes the send stream to open and then immediately
   transition to the "Reset Sent" state.

   Once a packet containing a RST_STREAM has been acknowledged, the send
   stream enters the "Reset Recvd" state, which is a terminal state.

9.2.2.  Receive Stream States

   Figure 18 shows the states for the part of a stream that receives
   data from a peer.  The states for a receive stream mirror only some
   of the states of the send stream at the peer.  A receive stream
   doesn't track states on the send stream that cannot be observed, such
   as the "Ready" state; instead, receive streams track the delivery of
   data to the application or application protocol some of which cannot
   be observed by the sender.










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          o
          | Recv STREAM / STREAM_BLOCKED / RST_STREAM
          | Create Bidirectional Stream (Sending)
          | Recv MAX_STREAM_DATA
          | Create Higher-Numbered Stream
          v
      +-------+
      | Recv  | Recv RST_STREAM
      |       |-----------------------.
      +-------+                       |
          |                           |
          | Recv STREAM + FIN         |
          v                           |
      +-------+                       |
      | Size  | Recv RST_STREAM       |
      | Known +---------------------->|
      +-------+                       |
          |                           |
          | Recv All Data             |
          v                           v
      +-------+                   +-------+
      | Data  | Recv RST_STREAM   | Reset |
      | Recvd +<-- (optional) --->| Recvd |
      +-------+                   +-------+
          |                           |
          | App Read All Data         | App Read RST
          v                           v
      +-------+                   +-------+
      | Data  |                   | Reset |
      | Read  |                   | Read  |
      +-------+                   +-------+

                   Figure 18: States for Receive Streams

   The receiving part of a stream initiated by a peer (types 1 and 3 for
   a client, or 0 and 2 for a server) are created when the first STREAM,
   STREAM_BLOCKED, RST_STREAM, or MAX_STREAM_DATA (bidirectional only,
   see below) is received for that stream.  The initial state for a
   receive stream is "Recv".  Receiving a RST_STREAM frame causes the
   receive stream to immediately transition to the "Reset Recvd".

   The receive stream enters the "Recv" state when the sending part of a
   bidirectional stream initiated by the endpoint (type 0 for a client,
   type 1 for a server) enters the "Ready" state.

   A bidirectional stream also opens when a MAX_STREAM_DATA frame is
   received.  Receiving a MAX_STREAM_DATA frame implies that the remote
   peer has opened the stream and is providing flow control credit.  A



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   MAX_STREAM_DATA frame might arrive before a STREAM or STREAM_BLOCKED
   frame if packets are lost or reordered.

   Before creating a stream, all lower-numbered streams of the same type
   MUST be created.  That means that receipt of a frame that would open
   a stream causes all lower-numbered streams of the same type to be
   opened in numeric order.  This ensures that the creation order for
   streams is consistent on both endpoints.

   In the "Recv" state, the endpoint receives STREAM and STREAM_BLOCKED
   frames.  Incoming data is buffered and can be reassembled into the
   correct order for delivery to the application.  As data is consumed
   by the application and buffer space becomes available, the endpoint
   sends MAX_STREAM_DATA frames to allow the peer to send more data.

   When a STREAM frame with a FIN bit is received, the final offset (see
   Section 10.3) is known.  The receive stream enters the "Size Known"
   state.  In this state, the endpoint no longer needs to send
   MAX_STREAM_DATA frames, it only receives any retransmissions of
   stream data.

   Once all data for the stream has been received, the receive stream
   enters the "Data Recvd" state.  This might happen as a result of
   receiving the same STREAM frame that causes the transition to "Size
   Known".  In this state, the endpoint has all stream data.  Any STREAM
   or STREAM_BLOCKED frames it receives for the stream can be discarded.

   The "Data Recvd" state persists until stream data has been delivered
   to the application or application protocol.  Once stream data has
   been delivered, the stream enters the "Data Read" state, which is a
   terminal state.

   Receiving a RST_STREAM frame in the "Recv" or "Size Known" states
   causes the stream to enter the "Reset Recvd" state.  This might cause
   the delivery of stream data to the application to be interrupted.

   It is possible that all stream data is received when a RST_STREAM is
   received (that is, from the "Data Recvd" state).  Similarly, it is
   possible for remaining stream data to arrive after receiving a
   RST_STREAM frame (the "Reset Recvd" state).  An implementation is
   able to manage this situation as they choose.  Sending RST_STREAM
   means that an endpoint cannot guarantee delivery of stream data;
   however there is no requirement that stream data not be delivered if
   a RST_STREAM is received.  An implementation MAY interrupt delivery
   of stream data, discard any data that was not consumed, and signal
   the existence of the RST_STREAM immediately.  Alternatively, the
   RST_STREAM signal might be suppressed or withheld if stream data is




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   completely received.  In the latter case, the receive stream
   effectively transitions to "Data Recvd" from "Reset Recvd".

   Once the application has been delivered the signal indicating that
   the receive stream was reset, the receive stream transitions to the
   "Reset Read" state, which is a terminal state.

9.2.3.  Permitted Frame Types

   The sender of a stream sends just three frame types that affect the
   state of a stream at either sender or receiver: STREAM
   (Section 7.20), STREAM_BLOCKED (Section 7.11), and RST_STREAM
   (Section 7.3).

   A sender MUST NOT send any of these frames from a terminal state
   ("Data Recvd" or "Reset Recvd").  A sender MUST NOT send STREAM or
   STREAM_BLOCKED after sending a RST_STREAM; that is, in the "Reset
   Sent" state in addition to the terminal states.  A receiver could
   receive any of these frames in any state, but only due to the
   possibility of delayed delivery of packets carrying them.

   The receiver of a stream sends MAX_STREAM_DATA (Section 7.7) and
   STOP_SENDING frames (Section 7.14).

   The receiver only sends MAX_STREAM_DATA in the "Recv" state.  A
   receiver can send STOP_SENDING in any state where it has not received
   a RST_STREAM frame; that is states other than "Reset Recvd" or "Reset
   Read".  However there is little value in sending a STOP_SENDING frame
   after all stream data has been received in the "Data Recvd" state.  A
   sender could receive these frames in any state as a result of delayed
   delivery of packets.

9.2.4.  Bidirectional Stream States

   A bidirectional stream is composed of a send stream and a receive
   stream.  Implementations may represent states of the bidirectional
   stream as composites of send and receive stream states.  The simplest
   model presents the stream as "open" when either send or receive
   stream is in a non-terminal state and "closed" when both send and
   receive streams are in a terminal state.

   Table 6 shows a more complex mapping of bidirectional stream states
   that loosely correspond to the stream states in HTTP/2 [HTTP2].  This
   shows that multiple states on send or receive streams are mapped to
   the same composite state.  Note that this is just one possibility for
   such a mapping; this mapping requires that data is acknowledged
   before the transition to a "closed" or "half-closed" state.




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   +-----------------------+---------------------+---------------------+
   | Send Stream           | Receive Stream      | Composite State     |
   +-----------------------+---------------------+---------------------+
   | No Stream/Ready       | No Stream/Recv *1   | idle                |
   |                       |                     |                     |
   | Ready/Send/Data Sent  | Recv/Size Known     | open                |
   |                       |                     |                     |
   | Ready/Send/Data Sent  | Data Recvd/Data     | half-closed         |
   |                       | Read                | (remote)            |
   |                       |                     |                     |
   | Ready/Send/Data Sent  | Reset Recvd/Reset   | half-closed         |
   |                       | Read                | (remote)            |
   |                       |                     |                     |
   | Data Recvd            | Recv/Size Known     | half-closed (local) |
   |                       |                     |                     |
   | Reset Sent/Reset      | Recv/Size Known     | half-closed (local) |
   | Recvd                 |                     |                     |
   |                       |                     |                     |
   | Data Recvd            | Recv/Size Known     | half-closed (local) |
   |                       |                     |                     |
   | Reset Sent/Reset      | Data Recvd/Data     | closed              |
   | Recvd                 | Read                |                     |
   |                       |                     |                     |
   | Reset Sent/Reset      | Reset Recvd/Reset   | closed              |
   | Recvd                 | Read                |                     |
   |                       |                     |                     |
   | Data Recvd            | Data Recvd/Data     | closed              |
   |                       | Read                |                     |
   |                       |                     |                     |
   | Data Recvd            | Reset Recvd/Reset   | closed              |
   |                       | Read                |                     |
   +-----------------------+---------------------+---------------------+

           Table 6: Possible Mapping of Stream States to HTTP/2

   Note (*1):  A stream is considered "idle" if it has not yet been
      created, or if the receive stream is in the "Recv" state without
      yet having received any frames.

9.3.  Solicited State Transitions

   If an endpoint is no longer interested in the data it is receiving on
   a stream, it MAY send a STOP_SENDING frame identifying that stream to
   prompt closure of the stream in the opposite direction.  This
   typically indicates that the receiving application is no longer
   reading data it receives from the stream, but is not a guarantee that
   incoming data will be ignored.




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   STREAM frames received after sending STOP_SENDING are still counted
   toward the connection and stream flow-control windows, even though
   these frames will be discarded upon receipt.  This avoids potential
   ambiguity about which STREAM frames count toward flow control.

   A STOP_SENDING frame requests that the receiving endpoint send a
   RST_STREAM frame.  An endpoint that receives a STOP_SENDING frame
   MUST send a RST_STREAM frame for that stream, and can use an error
   code of STOPPING.  If the STOP_SENDING frame is received on a send
   stream that is already in the "Data Sent" state, a RST_STREAM frame
   MAY still be sent in order to cancel retransmission of previously-
   sent STREAM frames.

   STOP_SENDING SHOULD only be sent for a receive stream that has not
   been reset.  STOP_SENDING is most useful for streams in the "Recv" or
   "Size Known" states.

   An endpoint is expected to send another STOP_SENDING frame if a
   packet containing a previous STOP_SENDING is lost.  However, once
   either all stream data or a RST_STREAM frame has been received for
   the stream - that is, the stream is in any state other than "Recv" or
   "Size Known" - sending a STOP_SENDING frame is unnecessary.

9.4.  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 6.6.1) and is subsequently
   increased by MAX_STREAM_ID frames (see Section 7.8).

   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 11), unless this is a result of a change in the initial
   limits (see Section 6.6.2).

   A receiver cannot renege on an advertisement; that is, once a
   receiver advertises a stream ID via a MAX_STREAM_ID frame,
   advertising a smaller maximum ID has no effect.  A sender MUST ignore
   any MAX_STREAM_ID frame that does not increase the maximum stream ID.




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9.5.  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.

   When new data is to be sent on a stream, a sender MUST set the
   encapsulating STREAM frame's offset field to the stream offset of the
   first byte of this new data.  The first octet of data on a stream has
   an offset of 0.  An endpoint is expected to send every stream octet.
   The largest offset delivered on a stream MUST be less than 2^62.

   QUIC makes no specific allowances for partial reliability or delivery
   of stream data out of order.  Endpoints MUST be able to deliver
   stream data to an application as an ordered byte-stream.  Delivering
   an ordered byte-stream requires that an endpoint buffer any data that
   is received out of order, up to the advertised flow control limit.

   An endpoint could receive the same octets multiple times; octets that
   have already been received can be discarded.  The value for a given
   octet MUST NOT change if it is sent multiple times; an endpoint MAY
   treat receipt of a changed octet as a connection error of type
   PROTOCOL_VIOLATION.

   An endpoint MUST NOT send data on any stream without ensuring that it
   is within the data limits set by its peer.

   Flow control is described in detail in Section 10, and congestion
   control is described in the companion document [QUIC-RECOVERY].

9.6.  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 [HTTP2], shows that effective prioritization strategies have a
   significant positive impact on performance.

   QUIC does not provide frames for exchanging prioritization
   information.  Instead it relies on receiving priority information
   from the application that uses QUIC.  Protocols that use QUIC are
   able to define any prioritization scheme that suits their application
   semantics.  A protocol might define explicit messages for signaling
   priority, such as those defined in HTTP/2; it could define rules that



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   allow an endpoint to determine priority based on context; or it could
   leave the determination to the application.

   A QUIC implementation SHOULD provide ways in which an application can
   indicate the relative priority of streams.  When deciding which
   streams to dedicate resources to, QUIC SHOULD use the information
   provided by the application.  Failure to account for priority of
   streams can result in suboptimal performance.

   Stream priority is most relevant when deciding which stream data will
   be transmitted.  Often, there will be limits on what can be
   transmitted as a result of connection flow control or the current
   congestion controller state.

   Giving preference to the transmission of its own management frames
   ensures that the protocol functions efficiently.  That is,
   prioritizing frames other than STREAM frames ensures that loss
   recovery, congestion control, and flow control operate effectively.

   CRYPTO frames SHOULD be prioritized over other streams prior to the
   completion of the cryptographic handshake.  This includes the
   retransmission of the second flight of client handshake messages,
   that is, the TLS Finished and any client authentication messages.

   STREAM data in frames determined to be lost SHOULD be retransmitted
   before sending new data, unless application priorities indicate
   otherwise.  Retransmitting lost stream data can fill in gaps, which
   allows the peer to consume already received data and free up flow
   control window.

10.  Flow Control

   It is necessary to limit the amount of data that a sender may have
   outstanding at any time, so as to prevent a fast sender from
   overwhelming a slow receiver, or to prevent a malicious sender from
   consuming significant resources at a receiver.  This section
   describes QUIC's flow-control mechanisms.

   QUIC employs a credit-based flow-control scheme similar to HTTP/2's
   flow control [HTTP2].  A receiver advertises the number of octets it
   is prepared to receive on a given stream and for the entire
   connection.  This leads to two levels of flow control in QUIC: (i)
   Connection flow control, which prevents senders from exceeding a
   receiver's buffer capacity for the connection, and (ii) Stream flow
   control, which prevents a single stream from consuming the entire
   receive buffer for a connection.





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   A data receiver sends MAX_STREAM_DATA or MAX_DATA frames to the
   sender to advertise additional credit.  MAX_STREAM_DATA frames send
   the maximum absolute byte offset of a stream, while MAX_DATA sends
   the maximum of the sum of the absolute byte offsets of all streams.

   A receiver MAY advertise a larger offset at any point by sending
   MAX_DATA or MAX_STREAM_DATA frames.  A receiver cannot renege on an
   advertisement; that is, once a receiver advertises an offset,
   advertising a smaller offset has no effect.  A sender MUST therefore
   ignore any MAX_DATA or MAX_STREAM_DATA frames that do not increase
   flow control limits.

   A receiver MUST close the connection with a FLOW_CONTROL_ERROR error
   (Section 11) if the peer violates the advertised connection or stream
   data limits.

   A sender SHOULD send BLOCKED or STREAM_BLOCKED frames to indicate it
   has data to write but is blocked by flow control limits.  These
   frames are expected to be sent infrequently in common cases, but they
   are considered useful for debugging and monitoring purposes.

   A receiver advertises credit for a stream by sending a
   MAX_STREAM_DATA frame with the Stream ID set appropriately.  A
   receiver could use the current offset of data consumed to determine
   the flow control offset to be advertised.  A receiver MAY send
   MAX_STREAM_DATA frames in multiple packets in order to make sure that
   the sender receives an update before running out of flow control
   credit, even if one of the packets is lost.

   Connection flow control is a limit to the total bytes of stream data
   sent in STREAM frames on all streams.  A receiver advertises credit
   for a connection by sending a MAX_DATA frame.  A receiver maintains a
   cumulative sum of bytes received on all contributing streams, which
   are used to check for flow control violations.  A receiver might use
   a sum of bytes consumed on all contributing streams to determine the
   maximum data limit to be advertised.

10.1.  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




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   deadlock, with the sender waiting for a MAX_DATA or MAX_STREAM_DATA
   frame which will never come.

   On receipt of a RST_STREAM frame, an endpoint will tear down state
   for the matching stream and ignore further data arriving on that
   stream.  This could result in the endpoints getting out of sync,
   since the RST_STREAM frame may have arrived out of order and there
   may be further bytes in flight.  The data sender would have counted
   the data against its connection level flow control budget, but a
   receiver that has not received these bytes would not know to include
   them as well.  The receiver must learn the number of bytes that were
   sent on the stream to make the same adjustment in its connection flow
   controller.

   To avoid this de-synchronization, a RST_STREAM sender MUST include
   the final byte offset sent on the stream in the RST_STREAM frame.  On
   receiving a RST_STREAM frame, a receiver definitively knows how many
   bytes were sent on that stream before the RST_STREAM frame, and the
   receiver MUST use the final offset to account for all bytes sent on
   the stream in its connection level flow controller.

10.1.1.  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.

10.1.2.  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 trade-off between resource commitment and overhead
   when determining how large a limit is advertised.

   A receiver MAY use an autotuning mechanism to tune the frequency and
   amount that it increases data limits based on a round-trip time
   estimate and the rate at which the receiving application consumes
   data, similar to common TCP implementations.



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10.2.  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.

10.2.1.  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
   SHOULD send a BLOCKED or STREAM_BLOCKED frame.  These frames are
   expected to be useful for debugging at the receiver; they do not
   require any other action.  A receiver SHOULD NOT wait for a BLOCKED
   or STREAM_BLOCKED frame before sending MAX_DATA or MAX_STREAM_DATA,
   since doing so will mean that a sender is unable to send for an
   entire round trip.

   For smooth operation of the congestion controller, it is generally
   considered best to not let the sender go into quiescence if
   avoidable.  To avoid blocking a sender, and to reasonably account for
   the possibility of loss, a receiver should send a MAX_DATA or
   MAX_STREAM_DATA frame at least two round trips before it expects the
   sender to get blocked.

   A sender sends a single BLOCKED or STREAM_BLOCKED frame only once
   when it reaches a data limit.  A sender SHOULD NOT send multiple
   BLOCKED or STREAM_BLOCKED frames for the same data limit, unless the
   original frame is determined to be lost.  Another BLOCKED or
   STREAM_BLOCKED frame can be sent after the data limit is increased.

10.3.  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 a RST_STREAM frame.  Otherwise, the
   final offset is the offset of the end of the data carried in a STREAM
   frame marked with a FIN flag, or 0 in the case of incoming
   unidirectional streams.

   An endpoint will know the final offset for a stream when the receive
   stream enters the "Size Known" or "Reset Recvd" state.



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   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 11).  A receiver SHOULD treat receipt of data at or
   beyond the final offset as a FINAL_OFFSET_ERROR error, even after a
   stream is closed.  Generating these errors is not mandatory, but only
   because requiring that an endpoint generate these errors also means
   that the endpoint needs to maintain the final offset state for closed
   streams, which could mean a significant state commitment.

10.4.  Flow Control for Cryptographic Handshake

   Data sent in CRYPTO frames is not flow controlled in the same way as
   STREAM frames.  QUIC relies on the cryptographic protocol
   implementation to avoid excessive buffering of data, see [QUIC-TLS].
   The implementation SHOULD provide an interface to QUIC to tell it
   about its buffering limits so that there is not excessive buffering
   at multiple layers.

11.  Error Handling

   An endpoint that detects an error SHOULD signal the existence of that
   error to its peer.  Both transport-level and application-level errors
   can affect an entire connection (see Section 11.1), while only
   application-level errors can be isolated to a single stream (see
   Section 11.2).

   The most appropriate error code (Section 11.3) 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 6.13.4) is not suitable for any error that
   can be signaled with a CONNECTION_CLOSE, APPLICATION_CLOSE, or
   RST_STREAM frame.  A stateless reset MUST NOT be used by an endpoint
   that has the state necessary to send a frame on the connection.

11.1.  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 or APPLICATION_CLOSE frame (Section 7.4,
   Section 7.5).  An endpoint MAY close the connection in this manner
   even if the error only affects a single stream.



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   Application protocols can signal application-specific protocol errors
   using the APPLICATION_CLOSE frame.  Errors that are specific to the
   transport, including all those described in this document, are
   carried in a CONNECTION_CLOSE frame.  Other than the type of error
   code they carry, these frames are identical in format and semantics.

   A CONNECTION_CLOSE or APPLICATION_CLOSE frame could be sent in a
   packet that is lost.  An endpoint SHOULD be prepared to retransmit a
   packet containing either frame type if it receives more packets on a
   terminated connection.  Limiting the number of retransmissions and
   the time over which this final packet is sent limits the effort
   expended on terminated connections.

   An endpoint that chooses not to retransmit packets containing
   CONNECTION_CLOSE or APPLICATION_CLOSE risks a peer missing the first
   such packet.  The only mechanism available to an endpoint that
   continues to receive data for a terminated connection is to use the
   stateless reset process (Section 6.13.4).

   An endpoint that receives an invalid CONNECTION_CLOSE or
   APPLICATION_CLOSE frame MUST NOT signal the existence of the error to
   its peer.

11.2.  Stream Errors

   If an application-level error affects a single stream, but otherwise
   leaves the connection in a recoverable state, the endpoint can send a
   RST_STREAM frame (Section 7.3) with an appropriate error code to
   terminate just the affected stream.

   Other than STOPPING (Section 9.3), RST_STREAM MUST be instigated by
   the application and MUST carry an application error code.  Resetting
   a stream without knowledge of the application protocol could cause
   the protocol to enter an unrecoverable state.  Application protocols
   might require certain streams to be reliably delivered in order to
   guarantee consistent state between endpoints.

11.3.  Transport Error Codes

   QUIC error codes are 16-bit unsigned integers.

   This section lists the defined QUIC transport error codes that may be
   used in a CONNECTION_CLOSE frame.  These errors apply to the entire
   connection.

   NO_ERROR (0x0):  An endpoint uses this with CONNECTION_CLOSE to
      signal that the connection is being closed abruptly in the absence
      of any error.



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   INTERNAL_ERROR (0x1):  The endpoint encountered an internal error and
      cannot continue with the connection.

   SERVER_BUSY (0x2):  The server is currently busy and does not accept
      any new connections.

   FLOW_CONTROL_ERROR (0x3):  An endpoint received more data than it
      permitted in its advertised data limits (see Section 10).

   STREAM_ID_ERROR (0x4):  An endpoint received a frame for a stream
      identifier that exceeded its advertised maximum stream ID.

   STREAM_STATE_ERROR (0x5):  An endpoint received a frame for a stream
      that was not in a state that permitted that frame (see
      Section 9.2).

   FINAL_OFFSET_ERROR (0x6):  An endpoint received a STREAM frame
      containing data that exceeded the previously established final
      offset.  Or an endpoint received a RST_STREAM frame containing a
      final offset that was lower than the maximum offset of data that
      was already received.  Or an endpoint received a RST_STREAM frame
      containing a different final offset to the one already
      established.

   FRAME_ENCODING_ERROR (0x7):  An endpoint received a frame that was
      badly formatted.  For instance, an empty STREAM frame that omitted
      the FIN flag, or an ACK frame that has more acknowledgment ranges
      than the remainder of the packet could carry.

   TRANSPORT_PARAMETER_ERROR (0x8):  An endpoint received transport
      parameters that were badly formatted, included an invalid value,
      was absent even though it is mandatory, was present though it is
      forbidden, or is otherwise in error.

   VERSION_NEGOTIATION_ERROR (0x9):  An endpoint received transport
      parameters that contained version negotiation parameters that
      disagreed with the version negotiation that it performed.  This
      error code indicates a potential version downgrade attack.

   PROTOCOL_VIOLATION (0xA):  An endpoint detected an error with
      protocol compliance that was not covered by more specific error
      codes.

   INVALID_MIGRATION (0xC):  A peer has migrated to a different network
      when the endpoint had disabled migration.

   CRYPTO_ERROR (0x1XX):  The cryptographic handshake failed.  A range
      of 256 values is reserved for carrying error codes specific to the



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      cryptographic handshake that is used.  Codes for errors occurring
      when TLS is used for the crypto handshake are described in
      Section 4.8 of [QUIC-TLS].

   See Section 13.3 for details of registering new error codes.

11.4.  Application Protocol Error Codes

   Application protocol error codes are 16-bit unsigned integers, but
   the management of application error codes are left to application
   protocols.  Application protocol error codes are used for the
   RST_STREAM (Section 7.3) and APPLICATION_CLOSE (Section 7.5) frames.

   There is no restriction on the use of the 16-bit error code space for
   application protocols.  However, QUIC reserves the error code with a
   value of 0 to mean STOPPING.  The application error code of STOPPING
   (0) is used by the transport to cancel a stream in response to
   receipt of a STOP_SENDING frame.

12.  Security Considerations

12.1.  Handshake Denial of Service

   As an encrypted and authenticated transport QUIC provides a range of
   protections against denial of service.  Once the cryptographic
   handshake is complete, QUIC endpoints discard most packets that are
   not authenticated, greatly limiting the ability of an attacker to
   interfere with existing connections.

   Once a connection is established QUIC endpoints might accept some
   unauthenticated ICMP packets (see Section 8.4.1), but the use of
   these packets is extremely limited.  The only other type of packet
   that an endpoint might accept is a stateless reset (Section 6.13.4)
   which relies on the token being kept secret until it is used.

   During the creation of a connection, QUIC only provides protection
   against attack from off the network path.  All QUIC packets contain
   proof that the recipient saw a preceding packet from its peer.

   The first mechanism used is the source and destination connection
   IDs, which are required to match those set by a peer.  Except for an
   Initial and stateless reset packets, an endpoint only accepts packets
   that include a destination connection that matches a connection ID
   the endpoint previously chose.  This is the only protection offered
   for Version Negotiation packets.

   The destination connection ID in an Initial packet is selected by a
   client to be unpredictable, which serves an additional purpose.  The



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   packets that carry the cryptographic handshake are protected with a
   key that is derived from this connection ID and salt specific to the
   QUIC version.  This allows endpoints to use the same process for
   authenticating packets that they receive as they use after the
   cryptographic handshake completes.  Packets that cannot be
   authenticated are discarded.  Protecting packets in this fashion
   provides a strong assurance that the sender of the packet saw the
   Initial packet and understood it.

   These protections are not intended to be effective against an
   attacker that is able to receive QUIC packets prior to the connection
   being established.  Such an attacker can potentially send packets
   that will be accepted by QUIC endpoints.  This version of QUIC
   attempts to detect this sort of attack, but it expects that endpoints
   will fail to establish a connection rather than recovering.  For the
   most part, the cryptographic handshake protocol [QUIC-TLS] is
   responsible for detecting tampering during the handshake, though
   additional validation is required for version negotiation (see
   Section 6.6.4).

   Endpoints are permitted to use other methods to detect and attempt to
   recover from interference with the handshake.  Invalid packets may be
   identified and discarded using other methods, but no specific method
   is mandated in this document.

12.2.  Spoofed ACK Attack

   An attacker might be able to receive an address validation token
   (Section 6.9) from the server and then release the IP address it used
   to acquire that token.  The attacker may, in the future, spoof this
   same address (which now presumably addresses a different endpoint),
   and initiate a 0-RTT connection with a server on the victim's behalf.
   The attacker can then spoof ACK frames to the server which cause the
   server to send excessive amounts of data toward the new owner of the
   IP address.

   There are two possible mitigations to this attack.  The simplest one
   is that a server can unilaterally create a gap in packet-number
   space.  In the non-attack scenario, the client will send an ACK frame
   with the larger value for largest acknowledged.  In the attack
   scenario, the attacker could acknowledge a packet in the gap.  If the
   server sees an acknowledgment for a packet that was never sent, the
   connection can be aborted.

   The second mitigation is that the server can require that
   acknowledgments for sent packets match the encryption level of the
   sent packet.  This mitigation is useful if the connection has an
   ephemeral forward-secure key that is generated and used for every new



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   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.

12.3.  Optimistic ACK Attack

   An endpoint that acknowledges packets it has not received might cause
   a congestion controller to permit sending at rates beyond what the
   network supports.  An endpoint MAY skip packet numbers when sending
   packets to detect this behavior.  An endpoint can then immediately
   close the connection with a connection error of type
   PROTOCOL_VIOLATION (see Section 6.13.3).

12.4.  Slowloris Attacks

   The attacks commonly known as Slowloris [SLOWLORIS] try to keep many
   connections to the target endpoint open and hold them open as long as
   possible.  These attacks can be executed against a QUIC endpoint by
   generating the minimum amount of activity necessary to avoid being
   closed for inactivity.  This might involve sending small amounts of
   data, gradually opening flow control windows in order to control the
   sender rate, or manufacturing ACK frames that simulate a high loss
   rate.

   QUIC deployments SHOULD provide mitigations for the Slowloris
   attacks, such as increasing the maximum number of clients the server
   will allow, limiting the number of connections a single IP address is
   allowed to make, imposing restrictions on the minimum transfer speed
   a connection is allowed to have, and restricting the length of time
   an endpoint is allowed to stay connected.

12.5.  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.





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   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.

12.6.  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 9.1.  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.

   The number of active streams is limited by the concurrent stream
   limit transport parameter, as explained in Section 9.4.  If chosen
   judiciously, this limit mitigates the effect of the stream commitment
   attack.  However, setting the limit too low could affect performance
   when applications expect to open large number of streams.

12.7.  Explicit Congestion Notification Attacks

   An on-path attacker could manipulate the value of ECN codepoints in
   the IP header to influence the sender's rate.  [RFC3168] discusses
   manipulations and their effects in more detail.

   An on-the-side attacker can duplicate and send packets with modified
   ECN codepoints to affect the sender's rate.  If duplicate packets are
   discarded by a receiver, an off-path attacker will need to race the
   duplicate packet against the original to be successful in this
   attack.  Therefore, QUIC receivers ignore ECN codepoints set in
   duplicate packets (see Section 6.8).

12.8.  Stateless Reset Oracle

   Stateless resets create a possible denial of service attack analogous
   to a TCP reset injection.  This attack is possible if an attacker is
   able to cause a stateless reset token to be generated for a
   connection with a selected connection ID.  An attacker that can cause
   this token to be generated can reset an active connection with the
   same connection ID.



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   If a packet can be routed to different instances that share a static
   key, for example by changing an IP address or port, then an attacker
   can cause the server to send a stateless reset.  To defend against
   this style of denial service, endpoints that share a static key for
   stateless reset (see Section 6.13.4.2) MUST be arranged so that
   packets with a given connection ID always arrive at an instance that
   has connection state, unless that connection is no longer active.

   In the case of a cluster that uses dynamic load balancing, it's
   possible that a change in load balancer configuration could happen
   while an active instance retains connection state; even if an
   instance retains connection state, the change in routing and
   resulting stateless reset will result in the connection being
   terminated.  If there is no chance in the packet being routed to the
   correct instance, it is better to send a stateless reset than wait
   for connections to time out.  However, this is acceptable only if the
   routing cannot be influenced by an attacker.

13.  IANA Considerations

13.1.  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
   [RFC8126].  Values with the first byte 0xff are reserved for Private
   Use [RFC8126].

   Registrations MUST include the following fields:

   Value:  The numeric value of the assignment (registrations will be
      between 0x0000 and 0xfeff).

   Parameter Name:  A short mnemonic for the parameter.

   Specification:  A reference to a publicly available specification for
      the value.

   The nominated expert(s) verify that a specification exists and is
   readily accessible.  Expert(s) are encouraged to be biased towards
   approving registrations unless they are abusive, frivolous, or
   actively harmful (not merely aesthetically displeasing, or
   architecturally dubious).




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   The initial contents of this registry are shown in Table 7.

     +--------+-------------------------------------+---------------+
     | Value  | Parameter Name                      | Specification |
     +--------+-------------------------------------+---------------+
     | 0x0000 | initial_max_stream_data_bidi_local  | Section 6.6.1 |
     |        |                                     |               |
     | 0x0001 | initial_max_data                    | Section 6.6.1 |
     |        |                                     |               |
     | 0x0002 | initial_max_bidi_streams            | Section 6.6.1 |
     |        |                                     |               |
     | 0x0003 | idle_timeout                        | Section 6.6.1 |
     |        |                                     |               |
     | 0x0004 | preferred_address                   | Section 6.6.1 |
     |        |                                     |               |
     | 0x0005 | max_packet_size                     | Section 6.6.1 |
     |        |                                     |               |
     | 0x0006 | stateless_reset_token               | Section 6.6.1 |
     |        |                                     |               |
     | 0x0007 | ack_delay_exponent                  | Section 6.6.1 |
     |        |                                     |               |
     | 0x0008 | initial_max_uni_streams             | Section 6.6.1 |
     |        |                                     |               |
     | 0x0009 | disable_migration                   | Section 6.6.1 |
     |        |                                     |               |
     | 0x000a | initial_max_stream_data_bidi_remote | Section 6.6.1 |
     |        |                                     |               |
     | 0x000b | initial_max_stream_data_uni         | Section 6.6.1 |
     +--------+-------------------------------------+---------------+

            Table 7: Initial QUIC Transport Parameters Entries

13.2.  QUIC Frame Type Registry

   IANA [SHALL add/has added] a registry for "QUIC Frame Types" under a
   "QUIC Protocol" heading.

   The "QUIC Frame Types" registry governs a 62-bit space.  This space
   is split into three spaces that are governed by different policies.
   Values between 0x00 and 0x3f (in hexadecimal) are assigned via the
   Standards Action or IESG Review policies [RFC8126].  Values from 0x40
   to 0x3fff operate on the Specification Required policy [RFC8126].
   All other values are assigned to Private Use [RFC8126].

   Registrations MUST include the following fields:

   Value:  The numeric value of the assignment (registrations will be
      between 0x00 and 0x3fff).  A range of values MAY be assigned.



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   Frame Name:  A short mnemonic for the frame type.

   Specification:  A reference to a publicly available specification for
      the value.

   The nominated expert(s) verify that a specification exists and is
   readily accessible.  Specifications for new registrations need to
   describe the means by which an endpoint might determine that it can
   send the identified type of frame.  An accompanying transport
   parameter registration (see Section 13.1) is expected for most
   registrations.  The specification needs to describe the format and
   assigned semantics of any fields in the frame.

   Expert(s) are encouraged to be biased towards approving registrations
   unless they are abusive, frivolous, or actively harmful (not merely
   aesthetically displeasing, or architecturally dubious).

   The initial contents of this registry are tabulated in Table 3.

13.3.  QUIC Transport Error Codes Registry

   IANA [SHALL add/has added] a registry for "QUIC Transport Error
   Codes" under a "QUIC Protocol" heading.

   The "QUIC Transport Error Codes" registry governs a 16-bit space.
   This space is split into two spaces that are governed by different
   policies.  Values with the first byte in the range 0x00 to 0xfe (in
   hexadecimal) are assigned via the Specification Required policy
   [RFC8126].  Values with the first byte 0xff are reserved for Private
   Use [RFC8126].

   Registrations MUST include the following fields:

   Value:  The numeric value of the assignment (registrations will be
      between 0x0000 and 0xfeff).

   Code:  A short mnemonic for the parameter.

   Description:  A brief description of the error code semantics, which
      MAY be a summary if a specification reference is provided.

   Specification:  A reference to a publicly available specification for
      the value.

   The initial contents of this registry are shown in Table 8.  Values
   from 0xFF00 to 0xFFFF are reserved for Private Use [RFC8126].





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   +------+---------------------------+----------------+---------------+
   | Valu | Error                     | Description    | Specification |
   | e    |                           |                |               |
   +------+---------------------------+----------------+---------------+
   | 0x0  | NO_ERROR                  | No error       | Section 11.3  |
   |      |                           |                |               |
   | 0x1  | INTERNAL_ERROR            | Implementation | Section 11.3  |
   |      |                           | error          |               |
   |      |                           |                |               |
   | 0x2  | SERVER_BUSY               | Server         | Section 11.3  |
   |      |                           | currently busy |               |
   |      |                           |                |               |
   | 0x3  | FLOW_CONTROL_ERROR        | Flow control   | Section 11.3  |
   |      |                           | error          |               |
   |      |                           |                |               |
   | 0x4  | STREAM_ID_ERROR           | Invalid stream | Section 11.3  |
   |      |                           | ID             |               |
   |      |                           |                |               |
   | 0x5  | STREAM_STATE_ERROR        | Frame received | Section 11.3  |
   |      |                           | in invalid     |               |
   |      |                           | stream state   |               |
   |      |                           |                |               |
   | 0x6  | FINAL_OFFSET_ERROR        | Change to      | Section 11.3  |
   |      |                           | final stream   |               |
   |      |                           | offset         |               |
   |      |                           |                |               |
   | 0x7  | FRAME_ENCODING_ERROR      | Frame encoding | Section 11.3  |
   |      |                           | error          |               |
   |      |                           |                |               |
   | 0x8  | TRANSPORT_PARAMETER_ERROR | Error in       | Section 11.3  |
   |      |                           | transport      |               |
   |      |                           | parameters     |               |
   |      |                           |                |               |
   | 0x9  | VERSION_NEGOTIATION_ERROR | Version        | Section 11.3  |
   |      |                           | negotiation    |               |
   |      |                           | failure        |               |
   |      |                           |                |               |
   | 0xA  | PROTOCOL_VIOLATION        | Generic        | Section 11.3  |
   |      |                           | protocol       |               |
   |      |                           | violation      |               |
   |      |                           |                |               |
   | 0xC  | INVALID_MIGRATION         | Violated       | Section 11.3  |
   |      |                           | disabled       |               |
   |      |                           | migration      |               |
   +------+---------------------------+----------------+---------------+

            Table 8: Initial QUIC Transport Error Codes Entries




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14.  References

14.1.  Normative References

   [PLPMTUD]  Mathis, M. and J. Heffner, "Packetization Layer Path MTU
              Discovery", RFC 4821, DOI 10.17487/RFC4821, March 2007,
              <https://www.rfc-editor.org/info/rfc4821>.

   [PMTUDv4]  Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191,
              DOI 10.17487/RFC1191, November 1990,
              <https://www.rfc-editor.org/info/rfc1191>.

   [PMTUDv6]  McCann, J., Deering, S., Mogul, J., and R. Hinden, Ed.,
              "Path MTU Discovery for IP version 6", STD 87, RFC 8201,
              DOI 10.17487/RFC8201, July 2017,
              <https://www.rfc-editor.org/info/rfc8201>.

   [QUIC-RECOVERY]
              Iyengar, J., Ed. and I. Swett, Ed., "QUIC Loss Detection
              and Congestion Control", draft-ietf-quic-recovery-14 (work
              in progress), August 2018.

   [QUIC-TLS]
              Thomson, M., Ed. and S. Turner, Ed., "Using Transport
              Layer Security (TLS) to Secure QUIC", draft-ietf-quic-
              tls-14 (work in progress), August 2018.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,
              <https://www.rfc-editor.org/info/rfc2119>.

   [RFC3168]  Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
              of Explicit Congestion Notification (ECN) to IP",
              RFC 3168, DOI 10.17487/RFC3168, September 2001,
              <https://www.rfc-editor.org/info/rfc3168>.

   [RFC3629]  Yergeau, F., "UTF-8, a transformation format of ISO
              10646", STD 63, RFC 3629, DOI 10.17487/RFC3629, November
              2003, <https://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,
              <https://www.rfc-editor.org/info/rfc4086>.






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   [RFC8126]  Cotton, M., Leiba, B., and T. Narten, "Guidelines for
              Writing an IANA Considerations Section in RFCs", BCP 26,
              RFC 8126, DOI 10.17487/RFC8126, June 2017,
              <https://www.rfc-editor.org/info/rfc8126>.

   [RFC8174]  Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
              2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
              May 2017, <https://www.rfc-editor.org/info/rfc8174>.

   [RFC8311]  Black, D., "Relaxing Restrictions on Explicit Congestion
              Notification (ECN) Experimentation", RFC 8311,
              DOI 10.17487/RFC8311, January 2018,
              <https://www.rfc-editor.org/info/rfc8311>.

   [TLS13]    Rescorla, E., "The Transport Layer Security (TLS) Protocol
              Version 1.3", RFC 8446, DOI 10.17487/RFC8446, August 2018,
              <https://www.rfc-editor.org/info/rfc8446>.

14.2.  Informative References

   [EARLY-DESIGN]
              Roskind, J., "QUIC: Multiplexed Transport Over UDP",
              December 2013, <https://goo.gl/dMVtFi>.

   [HTTP2]    Belshe, M., Peon, R., and M. Thomson, Ed., "Hypertext
              Transfer Protocol Version 2 (HTTP/2)", RFC 7540,
              DOI 10.17487/RFC7540, May 2015,
              <https://www.rfc-editor.org/info/rfc7540>.

   [QUIC-INVARIANTS]
              Thomson, M., "Version-Independent Properties of QUIC",
              draft-ietf-quic-invariants-01 (work in progress), August
              2018.

   [RFC2018]  Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP
              Selective Acknowledgment Options", RFC 2018,
              DOI 10.17487/RFC2018, October 1996,
              <https://www.rfc-editor.org/info/rfc2018>.

   [RFC2104]  Krawczyk, H., Bellare, M., and R. Canetti, "HMAC: Keyed-
              Hashing for Message Authentication", RFC 2104,
              DOI 10.17487/RFC2104, February 1997,
              <https://www.rfc-editor.org/info/rfc2104>.

   [RFC2360]  Scott, G., "Guide for Internet Standards Writers", BCP 22,
              RFC 2360, DOI 10.17487/RFC2360, June 1998,
              <https://www.rfc-editor.org/info/rfc2360>.




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   [RFC2406]  Kent, S. and R. Atkinson, "IP Encapsulating Security
              Payload (ESP)", RFC 2406, DOI 10.17487/RFC2406, November
              1998, <https://www.rfc-editor.org/info/rfc2406>.

   [RFC4787]  Audet, F., Ed. and C. Jennings, "Network Address
              Translation (NAT) Behavioral Requirements for Unicast
              UDP", BCP 127, RFC 4787, DOI 10.17487/RFC4787, January
              2007, <https://www.rfc-editor.org/info/rfc4787>.

   [RFC5869]  Krawczyk, H. and P. Eronen, "HMAC-based Extract-and-Expand
              Key Derivation Function (HKDF)", RFC 5869,
              DOI 10.17487/RFC5869, May 2010,
              <https://www.rfc-editor.org/info/rfc5869>.

   [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, <https://www.rfc-editor.org/info/rfc7301>.

   [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.

Appendix A.  Sample Packet Number Decoding Algorithm

   The following pseudo-code shows how an implementation can decode
   packet numbers after packet number protection has been removed.



















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   DecodePacketNumber(largest_pn, truncated_pn, pn_nbits):
      expected_pn  = largest_pn + 1
      pn_win       = 1 << pn_nbits
      pn_hwin      = pn_win / 2
      pn_mask      = pn_win - 1
      // The incoming packet number should be greater than
      // expected_pn - pn_hwin and less than or equal to
      // expected_pn + pn_hwin
      //
      // This means we can't just strip the trailing bits from
      // expected_pn and add the truncated_pn because that might
      // yield a value outside the window.
      //
      // The following code calculates a candidate value and
      // makes sure it's within the packet number window.
      candidate_pn = (expected_pn & ~pn_mask) | truncated_pn
      if candidate_pn <= expected_pn - pn_hwin:
         return candidate_pn + pn_win
      // Note the extra check for underflow when candidate_pn
      // is near zero.
      if candidate_pn > expected_pn + pn_hwin and
         candidate_pn > pn_win:
         return candidate_pn - pn_win
      return candidate_pn

Appendix B.  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.

B.1.  Since draft-ietf-quic-transport-13

   o  Streams open when higher-numbered streams of the same type open
      (#1342, #1549)

   o  Split initial stream flow control limit into 3 transport
      parameters (#1016, #1542)

   o  All flow control transport parameters are optional (#1610)

   o  Removed UNSOLICITED_PATH_RESPONSE error code (#1265, #1539)

   o  Permit stateless reset in response to any packet (#1348, #1553)

   o  Recommended defense against stateless reset spoofing (#1386,
      #1554)



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   o  Prevent infinite stateless reset exchanges (#1443, #1627)

   o  Forbid processing of the same packet number twice (#1405, #1624)

   o  Added a packet number decoding example (#1493)

   o  More precisely define idle timeout (#1429, #1614, #1652)

   o  Corrected format of Retry packet and prevented looping (#1492,
      #1451, #1448, #1498)

   o  Permit 0-RTT after receiving Version Negotiation or Retry (#1507,
      #1514, #1621)

   o  Permit Retry in response to 0-RTT (#1547, #1552)

   o  Looser verification of ECN counters to account for ACK loss
      (#1555, #1481, #1565)

   o  Remove frame type field from APPLICATION_CLOSE (#1508, #1528)

B.2.  Since draft-ietf-quic-transport-12

   o  Changes to integration of the TLS handshake (#829, #1018, #1094,
      #1165, #1190, #1233, #1242, #1252, #1450, #1458)

      *  The cryptographic handshake uses CRYPTO frames, not stream 0

      *  QUIC packet protection is used in place of TLS record
         protection

      *  Separate QUIC packet number spaces are used for the handshake

      *  Changed Retry to be independent of the cryptographic handshake

      *  Added NEW_TOKEN frame and Token fields to Initial packet

      *  Limit the use of HelloRetryRequest to address TLS needs (like
         key shares)

   o  Enable server to transition connections to a preferred address
      (#560, #1251, #1373)

   o  Added ECN feedback mechanisms and handling; new ACK_ECN frame
      (#804, #805, #1372)

   o  Changed rules and recommendations for use of new connection IDs
      (#1258, #1264, #1276, #1280, #1419, #1452, #1453, #1465)



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   o  Added a transport parameter to disable intentional connection
      migration (#1271, #1447)

   o  Packets from different connection ID can't be coalesced (#1287,
      #1423)

   o  Fixed sampling method for packet number encryption; the length
      field in long headers includes the packet number field in addition
      to the packet payload (#1387, #1389)

   o  Stateless Reset is now symmetric and subject to size constraints
      (#466, #1346)

   o  Added frame type extension mechanism (#58, #1473)

B.3.  Since draft-ietf-quic-transport-11

   o  Enable server to transition connections to a preferred address
      (#560, #1251)

   o  Packet numbers are encrypted (#1174, #1043, #1048, #1034, #850,
      #990, #734, #1317, #1267, #1079)

   o  Packet numbers use a variable-length encoding (#989, #1334)

   o  STREAM frames can now be empty (#1350)

B.4.  Since draft-ietf-quic-transport-10

   o  Swap payload length and packed number fields in long header
      (#1294)

   o  Clarified that CONNECTION_CLOSE is allowed in Handshake packet
      (#1274)

   o  Spin bit reserved (#1283)

   o  Coalescing multiple QUIC packets in a UDP datagram (#1262, #1285)

   o  A more complete connection migration (#1249)

   o  Refine opportunistic ACK defense text (#305, #1030, #1185)

   o  A Stateless Reset Token isn't mandatory (#818, #1191)

   o  Removed implicit stream opening (#896, #1193)





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   o  An empty STREAM frame can be used to open a stream without sending
      data (#901, #1194)

   o  Define stream counts in transport parameters rather than a maximum
      stream ID (#1023, #1065)

   o  STOP_SENDING is now prohibited before streams are used (#1050)

   o  Recommend including ACK in Retry packets and allow PADDING (#1067,
      #882)

   o  Endpoints now become closing after an idle timeout (#1178, #1179)

   o  Remove implication that Version Negotiation is sent when a packet
      of the wrong version is received (#1197)

B.5.  Since draft-ietf-quic-transport-09

   o  Added PATH_CHALLENGE and PATH_RESPONSE frames to replace PING with
      Data and PONG frame.  Changed ACK frame type from 0x0e to 0x0d.
      (#1091, #725, #1086)

   o  A server can now only send 3 packets without validating the client
      address (#38, #1090)

   o  Delivery order of stream data is no longer strongly specified
      (#252, #1070)

   o  Rework of packet handling and version negotiation (#1038)

   o  Stream 0 is now exempt from flow control until the handshake
      completes (#1074, #725, #825, #1082)

   o  Improved retransmission rules for all frame types: information is
      retransmitted, not packets or frames (#463, #765, #1095, #1053)

   o  Added an error code for server busy signals (#1137)

   o  Endpoints now set the connection ID that their peer uses.
      Connection IDs are variable length.  Removed the
      omit_connection_id transport parameter and the corresponding short
      header flag. (#1089, #1052, #1146, #821, #745, #821, #1166, #1151)

B.6.  Since draft-ietf-quic-transport-08

   o  Clarified requirements for BLOCKED usage (#65, #924)





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   o  BLOCKED frame now includes reason for blocking (#452, #924, #927,
      #928)

   o  GAP limitation in ACK Frame (#613)

   o  Improved PMTUD description (#614, #1036)

   o  Clarified stream state machine (#634, #662, #743, #894)

   o  Reserved versions don't need to be generated deterministically
      (#831, #931)

   o  You don't always need the draining period (#871)

   o  Stateless reset clarified as version-specific (#930, #986)

   o  initial_max_stream_id_x transport parameters are optional (#970,
      #971)

   o  Ack Delay assumes a default value during the handshake (#1007,
      #1009)

   o  Removed transport parameters from NewSessionTicket (#1015)

B.7.  Since draft-ietf-quic-transport-07

   o  The long header now has version before packet number (#926, #939)

   o  Rename and consolidate packet types (#846, #822, #847)

   o  Packet types are assigned new codepoints and the Connection ID
      Flag is inverted (#426, #956)

   o  Removed type for Version Negotiation and use Version 0 (#963,
      #968)

   o  Streams are split into unidirectional and bidirectional (#643,
      #656, #720, #872, #175, #885)

      *  Stream limits now have separate uni- and bi-directional
         transport parameters (#909, #958)

      *  Stream limit transport parameters are now optional and default
         to 0 (#970, #971)

   o  The stream state machine has been split into read and write (#634,
      #894)




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   o  Employ variable-length integer encodings throughout (#595)

   o  Improvements to connection close

      *  Added distinct closing and draining states (#899, #871)

      *  Draining period can terminate early (#869, #870)

      *  Clarifications about stateless reset (#889, #890)

   o  Address validation for connection migration (#161, #732, #878)

   o  Clearly defined retransmission rules for BLOCKED (#452, #65, #924)

   o  negotiated_version is sent in server transport parameters (#710,
      #959)

   o  Increased the range over which packet numbers are randomized
      (#864, #850, #964)

B.8.  Since draft-ietf-quic-transport-06

   o  Replaced FNV-1a with AES-GCM for all "Cleartext" packets (#554)

   o  Split error code space between application and transport (#485)

   o  Stateless reset token moved to end (#820)

   o  1-RTT-protected long header types removed (#848)

   o  No acknowledgments during draining period (#852)

   o  Remove "application close" as a separate close type (#854)

   o  Remove timestamps from the ACK frame (#841)

   o  Require transport parameters to only appear once (#792)

B.9.  Since draft-ietf-quic-transport-05

   o  Stateless token is server-only (#726)

   o  Refactor section on connection termination (#733, #748, #328,
      #177)

   o  Limit size of Version Negotiation packet (#585)

   o  Clarify when and what to ack (#736)



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   o  Renamed STREAM_ID_NEEDED to STREAM_ID_BLOCKED

   o  Clarify Keep-alive requirements (#729)

B.10.  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)

B.11.  Since draft-ietf-quic-transport-03

   o  Change STREAM and RST_STREAM layout

   o  Add MAX_STREAM_ID settings








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B.12.  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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B.13.  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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B.14.  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

B.15.  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

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.

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.






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Authors' Addresses

   Jana Iyengar (editor)
   Fastly

   Email: jri.ietf@gmail.com


   Martin Thomson (editor)
   Mozilla

   Email: martin.thomson@gmail.com







































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