This is a purely informative rendering of an RFC that includes verified errata. This rendering may not be used as a reference.

The following 'Verified' errata have been incorporated in this document: EID 6609

Internet Engineering Task Force (IETF)                           A. Ford
Request for Comments: 8684                                         Pexip
Obsoletes: 6824                                                C. Raiciu
Category: Standards Track                    U. Politehnica of Bucharest
ISSN: 2070-1721                                               M. Handley
                                                       U. College London
                                                          O. Bonaventure
                                                U. catholique de Louvain
                                                               C. Paasch
                                                             Apple, Inc.
                                                              March 2020

     TCP Extensions for Multipath Operation with Multiple Addresses


   TCP/IP communication is currently restricted to a single path per
   connection, yet multiple paths often exist between peers.  The
   simultaneous use of these multiple paths for a TCP/IP session would
   improve resource usage within the network and thus improve user
   experience through higher throughput and improved resilience to
   network failure.

   Multipath TCP provides the ability to simultaneously use multiple
   paths between peers.  This document presents a set of extensions to
   traditional TCP to support multipath operation.  The protocol offers
   the same type of service to applications as TCP (i.e., a reliable
   bytestream), and it provides the components necessary to establish
   and use multiple TCP flows across potentially disjoint paths.

   This document specifies v1 of Multipath TCP, obsoleting v0 as
   specified in RFC 6824, through clarifications and modifications
   primarily driven by deployment experience.

Status of This Memo

   This is an Internet Standards Track document.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Further information on
   Internet Standards is available in Section 2 of RFC 7841.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at

Copyright Notice

   Copyright (c) 2020 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
   ( 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
     1.1.  Design Assumptions
     1.2.  Multipath TCP in the Networking Stack
     1.3.  Terminology
     1.4.  MPTCP Concept
     1.5.  Requirements Language
   2.  Operation Overview
     2.1.  Initiating an MPTCP Connection
     2.2.  Associating a New Subflow with an Existing MPTCP Connection
     2.3.  Informing the Other Host about Another Potential Address
     2.4.  Data Transfer Using MPTCP
     2.5.  Requesting a Change in a Path's Priority
     2.6.  Closing an MPTCP Connection
     2.7.  Notable Features
   3.  MPTCP Operations: An Overview
     3.1.  Connection Initiation
     3.2.  Starting a New Subflow
     3.3.  MPTCP Operation and Data Transfer
       3.3.1.  Data Sequence Mapping
       3.3.2.  Data Acknowledgments
       3.3.3.  Closing a Connection
       3.3.4.  Receiver Considerations
       3.3.5.  Sender Considerations
       3.3.6.  Reliability and Retransmissions
       3.3.7.  Congestion Control Considerations
       3.3.8.  Subflow Policy
     3.4.  Address Knowledge Exchange (Path Management)
       3.4.1.  Address Advertisement
       3.4.2.  Remove Address
     3.5.  Fast Close
     3.6.  Subflow Reset
     3.7.  Fallback
     3.8.  Error Handling
     3.9.  Heuristics
       3.9.1.  Port Usage
       3.9.2.  Delayed Subflow Start and Subflow Symmetry
       3.9.3.  Failure Handling
   4.  Semantic Issues
   5.  Security Considerations
   6.  Interactions with Middleboxes
   7.  IANA Considerations
     7.1.  TCP Option Kind Numbers
     7.2.  MPTCP Option Subtypes
     7.3.  MPTCP Handshake Algorithms
     7.4.  MP_TCPRST Reason Codes
   8.  References
     8.1.  Normative References
     8.2.  Informative References
   Appendix A.  Notes on Use of TCP Options
   Appendix B.  TCP Fast Open and MPTCP
     B.1.  TFO Cookie Request with MPTCP
     B.2.  Data Sequence Mapping under TFO
     B.3.  Connection Establishment Examples
   Appendix C.  Control Blocks
     C.1.  MPTCP Control Block
       C.1.1.  Authentication and Metadata
       C.1.2.  Sending Side
       C.1.3.  Receiving Side
     C.2.  TCP Control Blocks
       C.2.1.  Sending Side
       C.2.2.  Receiving Side
   Appendix D.  Finite State Machine
   Appendix E.  Changes from RFC 6824
   Authors' Addresses

1.  Introduction

   Multipath TCP (MPTCP) is a set of extensions to regular TCP [RFC0793]
   to provide a Multipath TCP service [RFC6182], which enables a
   transport connection to operate across multiple paths simultaneously.
   This document presents the protocol changes required to add multipath
   capability to TCP -- specifically, those for signaling and setting up
   multiple paths ("subflows"), managing these subflows, reassembly of
   data, and termination of sessions.  This is not the only information
   required to create a Multipath TCP implementation, however.  This
   document is complemented by three others:

   *  [RFC6182] (MPTCP architecture), which explains the motivations
      behind Multipath TCP, contains a discussion of high-level design
      decisions on which this design is based, and provides an
      explanation of a functional separation through which an extensible
      MPTCP implementation can be developed.

   *  [RFC6356] (congestion control), which presents a safe congestion
      control algorithm for coupling the behavior of the multiple paths
      in order to "do no harm" to other network users.

   *  [RFC6897] (application considerations), which discusses what
      impact MPTCP will have on applications, what applications will
      want to do with MPTCP, and as a consequence of these factors, what
      API extensions an MPTCP implementation should present.

   This document obsoletes the v0 specification of Multipath TCP
   [RFC6824].  This document specifies MPTCP v1, which is not backward
   compatible with MPTCP v0.  This document additionally defines version
   negotiation procedures for implementations that support both

1.1.  Design Assumptions

   In order to limit the potentially huge design space, the MPTCP
   Working Group imposed two key constraints on the Multipath TCP design
   presented in this document:

   *  It must be backward compatible with current, regular TCP, to
      increase its chances of deployment.

   *  It can be assumed that one or both hosts are multihomed and

   To simplify the design, we assume that the presence of multiple
   addresses at a host is sufficient to indicate the existence of
   multiple paths.  These paths need not be entirely disjoint: they may
   share one or many routers between them.  Even in such a situation,
   making use of multiple paths is beneficial, improving resource
   utilization and resilience to a subset of node failures.  The
   congestion control algorithm defined in [RFC6356] ensures that the
   use of multiple paths does not act detrimentally.  Furthermore, there
   may be some scenarios where different TCP ports on a single host can
   provide disjoint paths (such as through certain Equal-Cost Multipath
   (ECMP) implementations [RFC2992]), and so the MPTCP design also
   supports the use of ports in path identifiers.

   There are three aspects to the backward compatibility listed above
   (discussed in more detail in [RFC6182]):

   External Constraints:  The protocol must function through the vast
      majority of existing middleboxes such as NATs, firewalls, and
      proxies, and as such must resemble existing TCP as far as possible
      on the wire.  Furthermore, the protocol must not assume that the
      segments it sends on the wire arrive unmodified at the
      destination: they may be split or coalesced; TCP options may be
      removed or duplicated.

   Application Constraints:  The protocol must be usable with no change
      to existing applications that use the common TCP API (although it
      is reasonable that not all features would be available to such
      legacy applications).  Furthermore, the protocol must provide the
      same service model as regular TCP to the application.

   Fallback:  The protocol should be able to fall back to standard TCP
      with no interference from the user, to be able to communicate with
      legacy hosts.

   The complementary application considerations document [RFC6897]
   discusses the necessary features of an API to provide backward
   compatibility, as well as API extensions to convey the behavior of
   MPTCP at a level of control and information equivalent to that
   available with regular, single-path TCP.

   Further discussion of the design constraints and associated design
   decisions is given in the MPTCP architecture document [RFC6182] and
   in [howhard].

1.2.  Multipath TCP in the Networking Stack

   MPTCP operates at the transport layer and aims to be transparent to
   both higher and lower layers.  It is a set of additional features on
   top of standard TCP; Figure 1 illustrates this layering.  MPTCP is
   designed to be usable by legacy applications with no changes;
   detailed discussion of its interactions with applications is given in

                                   |           Application         |
      +---------------+            +-------------------------------+
      |  Application  |            |             MPTCP             |
      +---------------+            + - - - - - - - + - - - - - - - +
      |      TCP      |            | Subflow (TCP) | Subflow (TCP) |
      +---------------+            +-------------------------------+
      |      IP       |            |       IP      |      IP       |
      +---------------+            +-------------------------------+

       Figure 1: Comparison of Standard TCP and MPTCP Protocol Stacks

1.3.  Terminology

   This document makes use of a number of terms that are either MPTCP
   specific or have defined meaning in the context of MPTCP, as follows:

   Path:  A sequence of links between a sender and a receiver, defined
      in this context by a 4-tuple of source and destination
      address/port pairs.

   Subflow:  A flow of TCP segments operating over an individual path,
      which forms part of a larger MPTCP connection.  A subflow is
      started and terminated similarly to a regular TCP connection.

   (MPTCP) Connection:  A set of one or more subflows, over which an
      application can communicate between two hosts.  There is a
      one-to-one mapping between a connection and an application socket.

   Data-level:  The payload data is nominally transferred over a
      connection, which in turn is transported over subflows.  Thus, the
      term "data-level" is synonymous with "connection-level", in
      contrast to "subflow-level", which refers to properties of an
      individual subflow.

   Token:  A locally unique identifier given to a multipath connection
      by a host.  May also be referred to as a "Connection ID".

   Host:  An end host operating an MPTCP implementation, and either
      initiating or accepting an MPTCP connection.

   In addition to these terms, note that MPTCP's interpretation of, and
   effect on, regular single-path TCP semantics are discussed in
   Section 4.

1.4.  MPTCP Concept

   This section provides a high-level summary of normal operation of
   MPTCP; this type of scenario is illustrated in Figure 2.  A detailed
   description of how MPTCP operates is given in Section 3.

               Host A                               Host B
      ------------------------             ------------------------
      Address A1    Address A2             Address B1    Address B2
      ----------    ----------             ----------    ----------
          |             |                      |             |
          |     (initial connection setup)     |             |
          |----------------------------------->|             |
          |<-----------------------------------|             |
          |             |                      |             |
          |            (additional subflow setup)            |
          |             |--------------------->|             |
          |             |<---------------------|             |
          |             |                      |             |
          |             |                      |             |

                   Figure 2: Example MPTCP Usage Scenario

   *  To a non-MPTCP-aware application, MPTCP will behave the same as
      normal TCP.  Extended APIs could provide additional control to
      MPTCP-aware applications [RFC6897].  An application begins by
      opening a TCP socket in the normal way.  MPTCP signaling and
      operation are handled by the MPTCP implementation.

   *  An MPTCP connection begins similarly to a regular TCP connection.
      This is illustrated in Figure 2, where an MPTCP connection is
      established between addresses A1 and B1 on Hosts A and B,

   *  If extra paths are available, additional TCP sessions (termed
      MPTCP "subflows") are created on these paths and are combined with
      the existing session, which continues to appear as a single
      connection to the applications at both ends.  The creation of the
      additional TCP session is illustrated between Address A2 on Host A
      and Address B1 on Host B.

   *  MPTCP identifies multiple paths by the presence of multiple
      addresses at hosts.  Combinations of these multiple addresses
      equate to the additional paths.  In the example, other potential
      paths that could be set up are A1<->B2 and A2<->B2.  Although this
      additional session is shown as being initiated from A2, it could
      equally have been initiated from B1 or B2.

   *  The discovery and setup of additional subflows will be achieved
      through a path management method; this document describes a
      mechanism by which a host can initiate new subflows by using its
      own additional addresses or by signaling its available addresses
      to the other host.

   *  MPTCP adds connection-level sequence numbers to allow the
      reassembly of segments arriving on multiple subflows with
      differing network delays.

   *  Subflows are terminated as regular TCP connections, with a
      four-way FIN handshake.  The MPTCP connection is terminated by a
      connection-level FIN.

1.5.  Requirements Language

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "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.

2.  Operation Overview

   This section presents a single description of common MPTCP operation,
   with reference to the protocol operation.  This is a high-level
   overview of the key functions; the full specification follows in
   Section 3.  Extensibility and negotiated features are not discussed
   here.  Considerable reference is made to symbolic names of MPTCP
   options throughout this section -- these are subtypes of the
   IANA-assigned MPTCP option (see Section 7), and their formats are
   defined in the detailed protocol specification provided in Section 3.

   A Multipath TCP connection provides a bidirectional bytestream
   between two hosts communicating like normal TCP and thus does not
   require any change to the applications.  However, Multipath TCP
   enables the hosts to use different paths with different IP addresses
   to exchange packets belonging to the MPTCP connection.  A Multipath
   TCP connection appears like a normal TCP connection to an
   application.  However, to the network layer, each MPTCP subflow looks
   like a regular TCP flow whose segments carry a new TCP option type.
   Multipath TCP manages the creation, removal, and utilization of these
   subflows to send data.  The number of subflows that are managed
   within a Multipath TCP connection is not fixed, and it can fluctuate
   during the lifetime of the Multipath TCP connection.

   All MPTCP operations are signaled with a TCP option -- a single
   numerical type for MPTCP, with "subtypes" for each MPTCP message.
   What follows is a summary of the purpose and rationale of these

2.1.  Initiating an MPTCP Connection

   This is the same signaling as for initiating a normal TCP connection,
   but the SYN, SYN/ACK, and initial ACK (and data) packets also carry
   the MP_CAPABLE option.  This option has a variable length and serves
   multiple purposes.  Firstly, it verifies whether the remote host
   supports Multipath TCP; secondly, this option allows the hosts to
   exchange some information to authenticate the establishment of
   additional subflows.  Further details are given in Section 3.1.

      Host A                                  Host B
      ------                                  ------
      MP_CAPABLE                ->
                                <-            MP_CAPABLE
                                              [B's key, flags]
      ACK + MP_CAPABLE (+ data) ->
      [A's key, B's key, flags, (data-level details)]

   Retransmission of the ACK + MP_CAPABLE can occur if it is not known
   if it has been received.  The following diagrams show all possible
   exchanges for the initial subflow setup to ensure this reliability.

      Host A (with data to send immediately)  Host B
      ------                                  ------
      MP_CAPABLE                ->
                                <-            MP_CAPABLE
                                              [B's key, flags]
      ACK + MP_CAPABLE + data   ->
      [A's key, B's key, flags, data-level details]

      Host A (with data to send later)        Host B
      ------                                  ------
      MP_CAPABLE                ->
                                <-            MP_CAPABLE
                                              [B's key, flags]
      ACK + MP_CAPABLE          ->
      [A's key, B's key, flags]

      ACK + MP_CAPABLE + data   ->
      [A's key, B's key, flags, data-level details]

      Host A                                  Host B (sending first)
      ------                                  ------
      MP_CAPABLE                ->
                                <-            MP_CAPABLE
                                              [B's key, flags]
      ACK + MP_CAPABLE          ->
      [A's key, B's key, flags]

                                <-            ACK + DSS + data
                                              [data-level details]

2.2.  Associating a New Subflow with an Existing MPTCP Connection

   The exchange of keys in the MP_CAPABLE handshake provides material
   that can be used to authenticate the endpoints when new subflows will
   be set up.  Additional subflows begin in the same way as initiating a
   normal TCP connection, but the SYN, SYN/ACK, and ACK packets also
   carry the MP_JOIN option.

   Host A initiates a new subflow between one of its addresses and one
   of Host B's addresses.  The token -- generated from the key -- is
   used to identify which MPTCP connection it is joining, and the
   Hash-based Message Authentication Code (HMAC) is used for
   authentication.  The HMAC uses the keys exchanged in the MP_CAPABLE
   handshake and the random numbers (nonces) exchanged in these MP_JOIN
   options.  MP_JOIN also contains flags and an Address ID that can be
   used to refer to the source address without the sender needing to
   know if it has been changed by a NAT.  Further details are given in
   Section 3.2.

      Host A                                  Host B
      ------                                  ------
      MP_JOIN               ->
      [B's token, A's nonce,
       A's Address ID, flags]
                            <-                MP_JOIN
                                              [B's HMAC, B's nonce,
                                               B's Address ID, flags]
      ACK + MP_JOIN         ->
      [A's HMAC]

                            <-                ACK

2.3.  Informing the Other Host about Another Potential Address

   The set of IP addresses associated to a multihomed host may change
   during the lifetime of an MPTCP connection.  MPTCP supports the
   addition and removal of addresses on a host both implicitly and
   explicitly.  If Host A has established a subflow starting at
   address/port pair IP#-A1 and wants to open a second subflow starting
   at address/port pair IP#-A2, it simply initiates the establishment of
   the subflow as explained above.  The remote host will then be
   implicitly informed about the new address.

   In some circumstances, a host may want to advertise to the remote
   host the availability of an address without establishing a new
   subflow -- for example, when a NAT prevents setup in one direction.
   In the example below, Host A informs Host B about its alternative
   IP address/port pair (IP#-A2).  Host B may later send an MP_JOIN to
   this new address.  The ADD_ADDR option contains an HMAC to
   authenticate the address as having been sent from the originator of
   the connection.  The receiver of this option echoes it back to the
   client to indicate successful receipt.  Further details are given in
   Section 3.4.1.

      Host A                                 Host B
      ------                                 ------
      ADD_ADDR                  ->
       IP#-A2's Address ID,
       HMAC of IP#-A2]

                                <-          ADD_ADDR
                                             IP#-A2's Address ID,
                                             HMAC of IP#-A2]

   There is a corresponding signal for address removal, making use of
   the Address ID that is signaled in the ADD_ADDR handshake.  Further
   details are given in Section 3.4.2.

      Host A                                 Host B
      ------                                 ------
      REMOVE_ADDR               ->
      [IP#-A2's Address ID]

2.4.  Data Transfer Using MPTCP

   To ensure reliable, in-order delivery of data over subflows that may
   appear and disappear at any time, MPTCP uses a 64-bit Data Sequence
   Number (DSN) to number all data sent over the MPTCP connection.  Each
   subflow has its own 32-bit sequence number space, utilizing the
   regular TCP sequence number header, and an MPTCP option maps the
   subflow sequence space to the data sequence space.  In this way, data
   can be retransmitted on different subflows (mapped to the same DSN)
   in the event of failure.

   The Data Sequence Signal (DSS) carries the Data Sequence Mapping.
   The Data Sequence Mapping consists of the subflow sequence number,
   data sequence number, and length for which this mapping is valid.
   This option can also carry a connection-level acknowledgment (the
   "Data ACK") for the received DSN.

   With MPTCP, all subflows share the same receive buffer and advertise
   the same receive window.  There are two levels of acknowledgment in
   MPTCP.  Regular TCP acknowledgments are used on each subflow to
   acknowledge the reception of the segments sent over the subflow
   independently of their DSN.  In addition, there are connection-level
   acknowledgments for the data sequence space.  These acknowledgments
   track the advancement of the bytestream and slide the receive window.

   Further details are given in Section 3.3.

      Host A                                 Host B
      ------                                 ------
      DSS                       ->
      [Data Sequence Mapping]
      [Data ACK]

2.5.  Requesting a Change in a Path's Priority

   Hosts can indicate at initial subflow setup whether they wish the
   subflow to be used as a regular or backup path -- a backup path only
   being used if there are no regular paths available.  During a
   connection, Host A can request a change in the priority of a subflow
   through the MP_PRIO signal to Host B.  Further details are given in
   Section 3.3.8.

      Host A                                 Host B
      ------                                 ------
      MP_PRIO                   ->

2.6.  Closing an MPTCP Connection

   When a host wants to close an existing subflow but not the whole
   connection, it can initiate a regular TCP FIN/ACK exchange.

   When Host A wants to inform Host B that it has no more data to send,
   it signals this "Data FIN" as part of the DSS (see above).  It has
   the same semantics and behavior as a regular TCP FIN, but at the
   connection level.  Once all the data on the MPTCP connection has been
   successfully received, this message is acknowledged at the connection
   level with a Data ACK.  Further details are given in Section 3.3.3.

      Host A                                 Host B
      ------                                 ------
      DSS                       ->
      [Data FIN]
                                <-           DSS
                                             [Data ACK]

   There is an additional method of connection closure, referred to as
   "Fast Close", which is analogous to closing a single-path TCP
   connection with a RST signal.  The MP_FASTCLOSE signal is used to
   indicate to the peer that the connection will be abruptly closed and
   no data will be accepted anymore.  This can be used on an ACK (which
   ensures reliability of the signal) or a RST (which does not).  Both
   examples are shown in the following diagrams.  Further details are
   given in Section 3.5.

      Host A                                 Host B
      ------                                 ------
      ACK + MP_FASTCLOSE          ->
      [B's key]

      [RST on all other subflows] ->

                                  <-         [RST on all subflows]

      Host A                                 Host B
      ------                                 ------
      RST + MP_FASTCLOSE          ->
      [B's key] [on all subflows]

                                  <-         [RST on all subflows]

2.7.  Notable Features

   It is worth highlighting that MPTCP's signaling has been designed
   with several key requirements in mind:

   *  To cope with NATs on the path, addresses are referred to by
      Address IDs, in case the IP packet's source address gets changed
      by a NAT.  Setting up a new TCP flow is not possible if the
      receiver of the SYN is behind a NAT; to allow subflows to be
      created when either end is behind a NAT, MPTCP uses the ADD_ADDR

   *  MPTCP falls back to ordinary TCP if MPTCP operation is not
      possible -- for example, if one host is not MPTCP capable or if a
      middlebox alters the payload.  This is discussed in Section 3.7.

   *  To address the threats identified in [RFC6181], the following
      steps are taken: keys are sent in the clear in the MP_CAPABLE
      messages; MP_JOIN messages are secured with HMAC-SHA256 ([RFC2104]
      using the algorithm in [RFC6234]) using those keys; and standard
      TCP validity checks are made on the other messages (ensuring that
      sequence numbers are in-window [RFC5961]).  Residual threats to
      MPTCP v0 were identified in [RFC7430], and those affecting the
      protocol (i.e., modifications to ADD_ADDR) have been incorporated
      in this document.  Further discussion of security can be found in
      Section 5.

3.  MPTCP Operations: An Overview

   This section describes the operation of MPTCP.  The subsections below
   discuss each key part of the protocol operation.

   All MPTCP operations are signaled using optional TCP header fields.
   A single TCP option number ("Kind") has been assigned by IANA for
   MPTCP (see Section 7), and then individual messages will be
   determined by a "subtype", the values of which are also stored in an
   IANA registry (and are also listed in Section 7).  As with all TCP
   options, the Length field is specified in bytes and includes the
   2 bytes of Kind and Length.

   Throughout this document, when reference is made to an MPTCP option
   by symbolic name, such as "MP_CAPABLE", this refers to a TCP option
   with the single MPTCP option type, and with the subtype value of the
   symbolic name as defined in Section 7.  This subtype is a 4-bit field
   -- the first 4 bits of the option payload, as shown in Figure 3.  The
   MPTCP messages are defined in the following sections.

                          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
     |     Kind      |    Length     |Subtype|                       |
     +---------------+---------------+-------+                       |
     |                     Subtype-specific data                     |
     |                       (variable length)                       |

                       Figure 3: MPTCP Option Format

   Those MPTCP options associated with subflow initiation are used on
   packets with the SYN flag set.  Additionally, there is one MPTCP
   option for signaling metadata to ensure that segmented data can be
   recombined for delivery to the application.

   The remaining options, however, are signals that do not need to be on
   a specific packet, such as those for signaling additional addresses.
   While an implementation may desire to send MPTCP options as soon as
   possible, it may not be possible to combine all desired options (both
   those for MPTCP and for regular TCP, such as SACK (selective
   acknowledgment) [RFC2018]) on a single packet.  Therefore, an
   implementation may choose to send duplicate ACKs containing the
   additional signaling information.  This changes the semantics of a
   duplicate ACK; these are usually only sent as a signal of a lost
   segment [RFC5681] in regular TCP.  Therefore, an MPTCP implementation
   receiving a duplicate ACK that contains an MPTCP option MUST NOT
   treat it as a signal of congestion.  Additionally, an MPTCP
   implementation SHOULD NOT send more than two duplicate ACKs in a row
   for the purposes of sending MPTCP options alone, in order to ensure
   that no middleboxes misinterpret this as a sign of congestion.

   Furthermore, standard TCP validity checks (such as ensuring that the
   sequence number and acknowledgment number are within the window) MUST
   be undertaken before processing any MPTCP signals, as described in
   [RFC5961], and initial subflow sequence numbers SHOULD be generated
   according to the recommendations in [RFC6528].

3.1.  Connection Initiation

   Connection initiation begins with a SYN, SYN/ACK, ACK exchange on a
   single path.  Each packet contains the Multipath Capable (MP_CAPABLE)
   MPTCP option (Figure 4).  This option declares its sender capable of
   performing Multipath TCP and wishes to do so on this particular

                          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
     |     Kind      |    Length     |Subtype|Version|A|B|C|D|E|F|G|H|
     |                   Option Sender's Key (64 bits)               |
     |                      (if option Length > 4)                   |
     |                                                               |
     |                  Option Receiver's Key (64 bits)              |
     |                      (if option Length > 12)                  |
     |                                                               |
     |  Data-Level Length (16 bits)  |  Checksum (16 bits, optional) |

              Figure 4: Multipath Capable (MP_CAPABLE) Option

   The MP_CAPABLE exchange in this specification (v1) is different than
   that specified in v0.  If a host supports multiple versions of MPTCP,
   the sender of the MP_CAPABLE option SHOULD signal the highest version
   number it supports.  In return, in its MP_CAPABLE option, the
   receiver will signal the version number it wishes to use, which MUST
   be equal to or lower than the version number indicated in the initial
   MP_CAPABLE.  There is a caveat, though, with respect to this version
   negotiation with old listeners that only support v0.  A listener that
   supports v0 expects that the MP_CAPABLE option in the SYN segment
   will include the initiator's key.  If, however, the initiator already
   upgraded to v1, it won't include the key in the SYN segment.  Thus,
   the listener will ignore the MP_CAPABLE of this SYN segment and reply
   with a SYN/ACK that does not include an MP_CAPABLE.  The initiator
   MAY choose to immediately fall back to TCP or MAY choose to attempt a
   connection using MPTCP v0 (if the initiator supports v0), in order to
   discover whether the listener supports the earlier version of MPTCP.
   In general, an MPTCP v0 connection will likely be preferred over a
   TCP connection; however, in a particular deployment scenario, it may
   be known that the listener is unlikely to support MPTCP v0 and so the
   initiator may prefer not to attempt a v0 connection.  An initiator
   MAY cache information for a peer about what version of MPTCP it
   supports, if any, and use this information for future connection

   The MP_CAPABLE option is of variable length, with different fields
   included, depending on which packet the option is used on.  The full
   MP_CAPABLE option is shown in Figure 4.

   The MP_CAPABLE option is carried on the SYN, SYN/ACK, and ACK packets
   that start the first subflow of an MPTCP connection, as well as the
   first packet that carries data, if the initiator wishes to send
   first.  The data carried by each option is as follows, where
   A = initiator and B = listener.

   *  SYN (A->B): only the first 4 octets (Length = 4).

   *  SYN/ACK (B->A): B's key for this connection (Length = 12).

   *  ACK (no data) (A->B): A's key followed by B's key (Length = 20).

   *  ACK (with first data) (A->B): A's key followed by B's key followed
      by Data-Level Length, and optional Checksum (Length = 22 or 24).

   The contents of the option are determined by the SYN and ACK flags of
   the packet, along with the option's Length field.  In Figure 4,
   "Sender" and "Receiver" refer to the sender or receiver of the TCP
   packet (which can be either host).

   The initial SYN, containing just the MP_CAPABLE header, is used to
   define the version of MPTCP being requested and also to exchange
   flags to negotiate connection features, as described later.

   This option is used to declare the 64-bit keys that the end hosts
   have generated for this MPTCP connection.  These keys are used to
   authenticate the addition of future subflows to this connection.
   This is the only time the key will be sent in the clear on the wire
   (unless "Fast Close" (Section 3.5) is used); all future subflows will
   identify the connection using a 32-bit "token".  This token is a
   cryptographic hash of this key.  The algorithm for this process is
   dependent on the authentication algorithm selected; the method of
   selection is defined later in this section.

   Upon reception of the initial SYN segment, a stateful server
   generates a random key and replies with a SYN/ACK.  The key's method
   of generation is implementation specific.  The key MUST be hard to
   guess, and it MUST be unique for the sending host across all its
   current MPTCP connections.  Recommendations for generating random
   numbers for use in keys are given in [RFC4086].  Connections will be
   indexed at each host by the token (a one-way hash of the key).
   Therefore, an implementation will require a mapping from each token
   to the corresponding connection, and in turn to the keys for the

   There is a risk that two different keys will hash to the same token.
   The risk of hash collisions is usually small, unless the host is
   handling many tens of thousands of connections.  Therefore, an
   implementation SHOULD check its list of connection tokens to ensure
   that there is no collision before sending its key, and if there is,
   then it should generate a new key.  This would, however, be costly
   for a server with thousands of connections.  The subflow handshake
   mechanism (Section 3.2) will ensure that new subflows only join the
   correct connection, however, through the cryptographic handshake, as
   well as checking the connection tokens in both directions, and
   ensuring that sequence numbers are in-window.  So, in the worst case,
   if there was a token collision, the new subflow would not succeed,
   but the MPTCP connection would continue to provide a regular TCP

   Since key generation is implementation specific, there is no
   requirement that they simply be random numbers.  An implementation is
   free to exchange cryptographic material out of band and generate
   these keys from this material, in order to provide additional
   mechanisms by which to verify the identity of the communicating
   entities.  For example, an implementation could choose to link its
   MPTCP keys to those used in higher-layer TLS or SSH connections.

   If the server behaves in a stateless manner, it has to generate its
   own key in a verifiable fashion.  This verifiable way of generating
   the key can be done by using a hash of the 4-tuple, sequence number,
   and a local secret (similar to what is done for the TCP sequence
   number [RFC4987]).  It will thus be able to verify whether it is
   indeed the originator of the key echoed back in the subsequent
   MP_CAPABLE option.  As for a stateful server, the tokens SHOULD be
   checked for uniqueness; however, if uniqueness is not met and there
   is no way to generate an alternative verifiable key, then the
   connection MUST fall back to using regular TCP by not sending an

   The ACK carries both A's key and B's key.  This is the first time
   that A's key is seen on the wire, although it is expected that A will
   have generated a key locally before the initial SYN.  The echoing of
   B's key allows B to operate statelessly, as described above.
   Therefore, A's key must be delivered reliably to B, and in order to
   do this, the transmission of this packet must be made reliable.

   If B has data to send first, then the reliable delivery of the
   ACK + MP_CAPABLE is ensured by the receipt of this data with an MPTCP
   Data Sequence Signal (DSS) option (Section 3.3) containing a DATA_ACK
   for the MP_CAPABLE (which is the first octet of the data sequence
   space).  If, however, A wishes to send data first, it has two options
   to ensure the reliable delivery of the ACK + MP_CAPABLE.  If it
   immediately has data to send, then the first ACK (with data) would
   also contain an MP_CAPABLE option with additional data parameters
   (the Data-Level Length and optional Checksum as shown in Figure 4).
   If A does not immediately have data to send, it MUST include the
   MP_CAPABLE on the first ACK, but without the additional data
   parameters.  When A does have data to send, it must repeat the
   sending of the MP_CAPABLE option from the first ACK, with additional
   data parameters.  This MP_CAPABLE option is used in place of the DSS
   and simply specifies (1) the Data-Level Length of the payload and
   (2) the checksum (if the use of checksums is negotiated).  This is
   the minimal data required to establish an MPTCP connection -- it
   allows validation of the payload, and given that it is the first
   data, the Initial Data Sequence Number (IDSN) is also known (as it is
   generated from the key, as described below).  Conveying the keys on
   the first data packet allows the TCP reliability mechanisms to ensure
   that the packet is successfully delivered.  The receiver will
   acknowledge this data at the connection level with a Data ACK, as if
   a DSS option has been received.

   There could be situations where both A and B attempt to transmit
   initial data at the same time.  For example, if A did not initially
   have data to send but then needed to transmit data before it had
   received anything from B, it would use an MP_CAPABLE option with data
   parameters (since it would not know if the MP_CAPABLE on the ACK was
   received).  In such a situation, B may also have transmitted data
   with a DSS option, but it had not yet been received at A.  Therefore,
   B has received data with an MP_CAPABLE mapping after it has sent data
   with a DSS option.  To ensure that these situations can be handled,
   it follows that the data parameters in an MP_CAPABLE are semantically
   equivalent to those in a DSS option and can be used interchangeably.
   Similar situations could occur when the MP_CAPABLE with data is lost
   and retransmitted.  Furthermore, in the case of TCP segmentation
   offloading, the MP_CAPABLE with data parameters may be duplicated
   across multiple packets, and implementations must also be able to
   cope with duplicate MP_CAPABLE mappings as well as duplicate DSS

   Additionally, the MP_CAPABLE exchange allows the safe passage of
   MPTCP options on SYN packets to be determined.  If any of these
   options are dropped, MPTCP will gracefully fall back to regular
   single-path TCP, as documented in Section 3.7.  If at any point in
   the handshake either party thinks the MPTCP negotiation is
   compromised -- for example, by a middlebox corrupting the TCP options
   or by unexpected ACK numbers being present -- the host MUST stop
   using MPTCP and no longer include MPTCP options in future TCP
   packets.  The other host will then also fall back to regular TCP
   using the fallback mechanism.  Note that new subflows MUST NOT be
   established (using the process documented in Section 3.2) until a DSS
   option has been successfully received across the path (as documented
   in Section 3.3).

   Like all MPTCP options, the MP_CAPABLE option starts with the Kind
   and Length to specify the TCP option's kind and length.  This
   information is followed by the MP_CAPABLE option.  The first 4 bits
   of the first octet in the MP_CAPABLE option (Figure 4) define the
   MPTCP Option Subtype (see Section 7; for MP_CAPABLE, this value is
   0x0), and the remaining 4 bits of this octet specify the MPTCP
   version in use (for this specification, this value is 1).

   The second octet is reserved for flags, allocated as follows:

   A:            The leftmost bit, labeled "A", SHOULD be set to 1 to
                 indicate "Checksum required", unless the system
                 administrator has decided that checksums are not
                 required (for example, if the environment is controlled
                 and no middleboxes exist that might adjust the

   B:            The second bit, labeled "B", is an extensibility flag.
                 It MUST be set to 0 for current implementations.  This
                 flag will be used for an extensibility mechanism in a
                 future specification, and the impact of this flag will
                 be defined at a later date.  It is expected, but not
                 mandated, that this flag would be used as part of an
                 alternative security mechanism that does not require a
                 full version upgrade of the protocol but does require
                 redefining some elements of the handshake.  If
                 receiving a message with the "B" flag set to 1 and this
                 is not understood, then the MP_CAPABLE in this SYN MUST
                 be silently ignored, which triggers a fallback to
                 regular TCP; the sender is expected to retry with a
                 format compatible with this legacy specification.  Note
                 that the length of the MP_CAPABLE option, and the
                 meanings of bits "D" through "H", may be altered by
                 setting B=1.

   C:            The third bit, labeled "C", is set to 1 to indicate
                 that the sender of this option will not accept
                 additional MPTCP subflows to the source address and
                 port, and therefore the receiver MUST NOT try to open
                 any additional subflows toward this address and port.
                 This improves efficiency in situations where the sender
                 knows a restriction is in place -- for example, if the
                 sender is behind a strict NAT or operating behind a
                 legacy Layer 4 load balancer.

   D through H:  The remaining bits, labeled "D" through "H", are used
                 for crypto algorithm negotiation.  In this
                 specification, only the rightmost bit, labeled "H", is
                 assigned.  Bit "H" indicates the use of HMAC-SHA256 (as
                 defined in Section 3.2).  An implementation that only
                 supports this method MUST set bit "H" to 1 and bits "D"
                 through "G" to 0.

   A crypto algorithm MUST be specified.  If flag bits "D" through "H"
   are all 0, the MP_CAPABLE option MUST be treated as invalid and
   ignored (that is, it must be treated as a regular TCP handshake).

   The selection of the authentication algorithm also impacts the
   algorithm used to generate the token and the IDSN.  In this
   specification, with only the SHA-256 algorithm (bit "H") specified
   and selected, the token MUST be a truncated (most significant
   32 bits) SHA-256 hash [RFC6234] of the key.  A different, 64-bit
   truncation (the least significant 64 bits) of the SHA-256 hash of the
   key MUST be used as the IDSN.  Note that the key MUST be hashed in
   network byte order.  Also note that the "least significant" bits MUST
   be the rightmost bits of the SHA-256 digest, as per [RFC6234].
   Future specifications of the use of the crypto bits may choose to
   specify different algorithms for token and IDSN generation.

   Both the crypto and checksum bits negotiate capabilities in similar
   ways.  For the "Checksum required" bit (labeled "A"), if either host
   requires the use of checksums, checksums MUST be used.  In other
   words, the only way for checksums not to be used is if both hosts in
   their SYNs set A=0.  This decision is confirmed by the setting of the
   "A" bit in the third packet (the ACK) of the handshake.  For example,
   if the initiator sets A=0 in the SYN but the responder sets A=1 in
   the SYN/ACK, checksums MUST be used in both directions, and the
   initiator will set A=1 in the ACK.  The decision regarding whether to
   use checksums will be stored by an implementation in a per-connection
   binary state variable.  If A=1 is received by a host that does not
   want to use checksums, it MUST fall back to regular TCP by ignoring
   the MP_CAPABLE option as if it was invalid.

   For crypto negotiation, the responder has the choice.  The initiator
   creates a proposal setting a bit for each algorithm it supports to 1
   (in this version of the specification, there is only one proposal, so
   bit "H" will always be set to 1).  The responder responds with only
   1 bit set -- this is the chosen algorithm.  The rationale for this
   behavior is that the responder will typically be a server with
   potentially many thousands of connections, so it may wish to choose
   an algorithm with minimal computational complexity, depending on the
   load.  If a responder does not support (or does not want to support)
   any of the initiator's proposals, it MUST respond without an
   MP_CAPABLE option, thus forcing a fallback to regular TCP.

   The MP_CAPABLE option is only used in the first subflow of a
   connection, in order to identify the connection; all subsequent
   subflows will use the MP_JOIN option (see Section 3.2) to join the
   existing connection.

   If a SYN contains an MP_CAPABLE option but the SYN/ACK does not, it
   is assumed that the sender of the SYN/ACK is not multipath capable;
   thus, the MPTCP session MUST operate as a regular, single-path TCP
   session.  If a SYN does not contain an MP_CAPABLE option, the SYN/ACK
   MUST NOT contain one in response.  If the third packet (the ACK) does
   not contain the MP_CAPABLE option, then the session MUST fall back to
   operating as a regular, single-path TCP session.  This is done to
   maintain compatibility with middleboxes on the path that drop some or
   all TCP options.  Note that an implementation MAY choose to attempt
   sending MPTCP options more than one time before making this decision
   to operate as regular TCP (see Section 3.9).

   If the SYN packets are unacknowledged, it is up to local policy to
   decide how to respond.  It is expected that a sender will eventually
   fall back to single-path TCP (i.e., without the MP_CAPABLE option) in
   order to work around middleboxes that may drop packets with unknown
   options; however, the number of multipath-capable attempts that are
   made first will be up to local policy.  It is possible that MPTCP and
   non-MPTCP SYNs could get reordered in the network.  Therefore, the
   final state is inferred from the presence or absence of the
   MP_CAPABLE option in the third packet of the TCP handshake.  If this
   option is not present, the connection SHOULD fall back to regular
   TCP, as documented in Section 3.7.

   The IDSN on an MPTCP connection is generated from the key.  The
   algorithm for IDSN generation is also determined from the negotiated
   authentication algorithm.  In this specification, with only the
   SHA-256 algorithm specified and selected, the IDSN of a host MUST be
   the least significant 64 bits of the SHA-256 hash of its key, i.e.,
   IDSN-A = Hash(Key-A) and IDSN-B = Hash(Key-B).  This deterministic
   generation of the IDSN allows a receiver to ensure that there are no
   gaps in sequence space at the start of the connection.  The SYN with
   MP_CAPABLE occupies the first octet of data sequence space, although
   this does not need to be acknowledged at the connection level until
   the first data is sent (see Section 3.3).

3.2.  Starting a New Subflow

   Once an MPTCP connection has begun with the MP_CAPABLE exchange,
   further subflows can be added to the connection.  Hosts have
   knowledge of their own address(es) and can become aware of the other
   host's addresses through signaling exchanges as described in
   Section 3.4.  Using this knowledge, a host can initiate a new subflow
   over a currently unused pair of addresses.  It is permissible for
   either host in a connection to initiate the creation of a new
   subflow, but it is expected that this will normally be the original
   connection initiator (see Section 3.9 for heuristics).

   A new subflow is started as a normal TCP SYN/ACK exchange.  The Join
   Connection (MP_JOIN) MPTCP option is used to identify the connection
   to be joined by the new subflow.  It uses keying material that was
   exchanged in the initial MP_CAPABLE handshake (Section 3.1), and that
   handshake also negotiates the crypto algorithm in use for the MP_JOIN

   This section specifies the behavior of MP_JOIN using the HMAC-SHA256
   algorithm.  An MP_JOIN option is present in the SYN, SYN/ACK, and ACK
   of the three-way handshake, although in each case with a different

   In the first MP_JOIN on the SYN packet, illustrated in Figure 5, the
   initiator sends a token, random number, and Address ID.

                          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
     |     Kind      |  Length = 12  |Subtype|(rsv)|B|   Address ID  |
     |                   Receiver's Token (32 bits)                  |
     |                Sender's Random Number (32 bits)               |

        Figure 5: Join Connection (MP_JOIN) Option (for Initial SYN)

   The token is used to identify the MPTCP connection and is a
   cryptographic hash of the receiver's key, as exchanged in the initial
   MP_CAPABLE handshake (Section 3.1).  In this specification, the
   tokens presented in this option are generated by the SHA-256
   algorithm [RFC6234], truncated to the most significant 32 bits.  The
   token included in the MP_JOIN option is the token that the receiver
   of the packet uses to identify this connection; i.e., Host A will
   send Token-B (which is generated from Key-B).  Note that the hash
   generation algorithm can be overridden by the choice of cryptographic
   handshake algorithm, as defined in Section 3.1.

   The MP_JOIN SYN sends not only the token (which is static for a
   connection) but also random numbers (nonces) that are used to prevent
   replay attacks on the authentication method.  Recommendations for the
   generation of random numbers for this purpose are given in [RFC4086].

   The MP_JOIN option includes an "Address ID".  This is an identifier
   generated by the sender of the option, used to identify the source
   address of this packet, even if the IP header has been changed in
   transit by a middlebox.  The numeric value of this field is generated
   by the sender and must map uniquely to a source IP address for the
   sending host.  The Address ID allows address removal (Section 3.4.2)
   without needing to know what the source address at the receiver is,
   thus allowing address removal through NATs.  The Address ID also
   allows correlation between new subflow setup attempts and address
   signaling (Section 3.4.1), to prevent setting up duplicate subflows
   on the same path, if an MP_JOIN and ADD_ADDR are sent at the same

   The Address IDs of the subflow used in the initial SYN exchange of
   the first subflow in the connection are implicit and have the value
   zero.  A host MUST store the mappings between Address IDs and
   addresses both for itself and the remote host.  An implementation
   will also need to know which local and remote Address IDs are
   associated with which established subflows, for when addresses are
   removed from a local or remote host.

   The MP_JOIN option on packets with the SYN flag set also includes
   4 bits of flags, 3 of which are currently reserved and MUST be set to
   0 by the sender.  The final bit, labeled "B", indicates whether the
   sender of this option (1) wishes this subflow to be used as a backup
   path (B=1) in the event of failure of other paths or (2) wants the
   subflow to be used as part of the connection immediately.  By setting
   B=1, the sender of the option is requesting that the other host only
   send data on this subflow if there are no available subflows where
   B=0.  Subflow policy is discussed in more detail in Section 3.3.8.

   When receiving a SYN with an MP_JOIN option that contains a valid
   token for an existing MPTCP connection, the recipient SHOULD respond
   with a SYN/ACK also containing an MP_JOIN option containing a random
   number and a truncated (leftmost 64 bits) HMAC.  This version of the
   option is shown in Figure 6.  If the token is unknown or the host
   wants to refuse subflow establishment (for example, due to a limit on
   the number of subflows it will permit), the receiver will send back a
   reset (RST) signal, analogous to an unknown port in TCP, containing
   an MP_TCPRST option (Section 3.6) with an "MPTCP specific error"
   reason code.  Although calculating an HMAC requires cryptographic
   operations, it is believed that the 32-bit token in the MP_JOIN SYN
   gives sufficient protection against blind state exhaustion attacks;
   therefore, there is no need to provide mechanisms to allow a
   responder to operate statelessly at the MP_JOIN stage.

                          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
     |     Kind      |  Length = 16  |Subtype|(rsv)|B|   Address ID  |
     |                                                               |
     |                Sender's Truncated HMAC (64 bits)              |
     |                                                               |
     |                Sender's Random Number (32 bits)               |

    Figure 6: Join Connection (MP_JOIN) Option (for Responding SYN/ACK)

   An HMAC is sent by both hosts -- by the initiator (Host A) in the
   third packet (the ACK) and by the responder (Host B) in the second
   packet (the SYN/ACK).  Doing the HMAC exchange at this stage allows
   both hosts to have first exchanged random data (in the first two SYN
   packets) that is used as the "message".  This specification defines
   that HMAC as defined in [RFC2104] is used, along with the SHA-256
   hash algorithm [RFC6234], and that the output is truncated to the
   leftmost 160 bits (20 octets).  Due to option space limitations, the
   HMAC included in the SYN/ACK is truncated to the leftmost 64 bits,
   but this is acceptable, since random numbers are used; thus, an
   attacker only has one chance to correctly guess the HMAC that matches
   the random number previously sent by the peer (if the HMAC is
   incorrect, the TCP connection is closed, so a new MP_JOIN negotiation
   with a new random number is required).

   The initiator's authentication information is sent in its first ACK
   (the third packet of the handshake), as shown in Figure 7.  This data
   needs to be sent reliably, since it is the only time this HMAC is
   sent; therefore, receipt of this packet MUST trigger a regular TCP
   ACK in response, and the packet MUST be retransmitted if this ACK is
   not received.  In other words, sending the ACK/MP_JOIN packet places
   the subflow in the PRE_ESTABLISHED state, and it moves to the
   ESTABLISHED state only on receipt of an ACK from the receiver.  It is
   not permissible to send data while in the PRE_ESTABLISHED state.  The
   reserved bits in this option MUST be set to 0 by the sender.

                          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
     |     Kind      |  Length = 24  |Subtype|      (reserved)       |
     |                                                               |
     |                                                               |
     |              Sender's Truncated HMAC (160 bits)               |
     |                                                               |
     |                                                               |

                 Figure 7: Join Connection (MP_JOIN) Option
                        (for Initiator's First ACK)

   The key for the HMAC algorithm, in the case of the message
   transmitted by Host A, will be Key-A followed by Key-B; and in the
   case of Host B, Key-B followed by Key-A.  These are the keys that
   were exchanged in the original MP_CAPABLE handshake.  The "message"
   for the HMAC algorithm in each case is the concatenations of random
   numbers for each host (denoted by R): for Host A, R-A followed by
   R-B; and for Host B, R-B followed by R-A.

   These various MPTCP options fit together to enable authenticated
   subflow setup as illustrated in Figure 8.

                   Host A                                  Host B
          ------------------------                       ----------
          Address A1    Address A2                       Address B1
          ----------    ----------                       ----------
              |             |                                |
              |             |  SYN + MP_CAPABLE              |
              |          SYN/ACK + MP_CAPABLE(Key-B)         |
              |             |                                |
              |        ACK + MP_CAPABLE(Key-A, Key-B)        |
              |             |                                |
              |             |   SYN + MP_JOIN(Token-B, R-A)  |
              |             |------------------------------->|
              |             |<-------------------------------|
              |             | SYN/ACK + MP_JOIN(HMAC-B, R-B) |
              |             |                                |
              |             |     ACK + MP_JOIN(HMAC-A)      |
              |             |------------------------------->|
              |             |<-------------------------------|
              |             |             ACK                |

          HMAC-A = HMAC(Key=(Key-A + Key-B), Msg=(R-A + R-B))
          HMAC-B = HMAC(Key=(Key-B + Key-A), Msg=(R-B + R-A))

               Figure 8: Example Use of MPTCP Authentication

   If the token received at Host B is unknown or local policy prohibits
   the acceptance of the new subflow, the recipient MUST respond with a
   TCP RST for the subflow.  If appropriate, an MP_TCPRST option with an
   "Administratively prohibited" reason code (Section 3.6) should be

   If the token is accepted at Host B but the HMAC returned to Host A
   does not match the one expected, Host A MUST close the subflow with a
   TCP RST.  In this and all subsequent cases of sending a RST as
   described in this section, the sender SHOULD send an MP_TCPRST option
   (Section 3.6) on this RST packet with the reason code for an "MPTCP-
   specific error".

   If Host B does not receive the expected HMAC or the MP_JOIN option is
   missing from the ACK, it MUST close the subflow with a TCP RST.

   If the HMACs are verified as correct, then both hosts have verified
   each other as being the same peers as those that existed at the start
   of the connection, and they have agreed of which connection this
   subflow will become a part.

   If the SYN/ACK as received at Host A does not have an MP_JOIN option,
   Host A MUST close the subflow with a TCP RST.

   This covers all cases of the loss of an MP_JOIN.  In more detail, if
   an MP_JOIN is stripped from the SYN on the path from A to B and
   Host B does not have a listener on the relevant port, it will respond
   with a RST in the normal way.  If in response to a SYN with an
   MP_JOIN option a SYN/ACK is received without the MP_JOIN option
   (because it was either stripped on the return path, or stripped on
   the outgoing path leading to Host B responding as if it was a new
   regular TCP session), then the subflow is unusable and Host A MUST
   close it with a RST.

   Note that additional subflows can be created between any pair of
   ports (but see Section 3.9 for heuristics); no explicit application-
   level accept calls or bind calls are required to open additional
   subflows.  To associate a new subflow with an existing connection,
   the token supplied in the subflow's SYN exchange is used for
   demultiplexing.  This then binds the 5-tuple of the TCP subflow to
   the local token of the connection.  One consequence is that it is
   possible to allow any port pairs to be used for a connection.

   Demultiplexing subflow SYNs MUST be done using the token; this is
   unlike traditional TCP, where the destination port is used for
   demultiplexing SYN packets.  Once a subflow is set up, demultiplexing
   packets is done using the 5-tuple, as in traditional TCP.  The
   5-tuples will be mapped to the local connection identifier (token).
   Note that Host A will know its local token for the subflow even
   though it is not sent on the wire -- only the responder's token is

3.3.  MPTCP Operation and Data Transfer

   This section discusses the operation of MPTCP for data transfer.  At
   a high level, an MPTCP implementation will take one input data stream
   from an application and split it into one or more subflows, with
   sufficient control information to allow it to be reassembled and
   delivered reliably and in order to the recipient application.  The
   following subsections define this behavior in detail.

   The Data Sequence Mapping and the Data ACK are signaled in the DSS
   option (Figure 9).  Either or both can be signaled in one DSS,
   depending on the flags set.  The Data Sequence Mapping defines how
   the sequence space on the subflow maps to the connection level, and
   the Data ACK acknowledges receipt of data at the connection level.
   These functions are described in more detail in the following two

                          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
     |     Kind      |    Length     |Subtype| (reserved) |F|m|M|a|A|
     |           Data ACK (4 or 8 octets, depending on flags)       |
     |   Data Sequence Number (4 or 8 octets, depending on flags)   |
     |              Subflow Sequence Number (4 octets)              |
     |  Data-Level Length (2 octets) |      Checksum (2 octets)     |

                Figure 9: Data Sequence Signal (DSS) Option

   The flags, when set, define the contents of this option, as follows:

   *  A = Data ACK present

   *  a = Data ACK is 8 octets (if not set, Data ACK is 4 octets)

   *  M = Data Sequence Number (DSN), Subflow Sequence Number (SSN),
      Data-Level Length, and Checksum (if negotiated) present

   *  m = Data Sequence Number is 8 octets (if not set, DSN is 4 octets)

   The flags "a" and "m" only have meaning if the corresponding "A" or
   "M" flags are set; otherwise, they will be ignored.  The maximum
   length of this option, with all flags set, is 28 octets.

   The "F" flag indicates "Data FIN".  If present, this means that this
   mapping covers the final data from the sender.  This is the
   connection-level equivalent of the FIN flag in single-path TCP.  A
   connection is not closed unless there has been a Data FIN exchange,
   an MP_FASTCLOSE (Section 3.5) message, or an implementation-specific
   connection-level send timeout.  The purpose of the Data FIN and the
   interactions between this flag, the subflow-level FIN flag, and the
   Data Sequence Mapping are described in Section 3.3.3.  The remaining
   reserved bits MUST be set to 0 by an implementation of this

   Note that the checksum is only present in this option if the use of
   MPTCP checksumming has been negotiated at the MP_CAPABLE handshake
   (see Section 3.1).  The presence of the checksum can be inferred from
   the length of the option.  If a checksum is present but its use had
   not been negotiated in the MP_CAPABLE handshake, the receiver MUST
   close the subflow with a RST, as it is not behaving as negotiated.
   If a checksum is not present when its use has been negotiated, the
   receiver MUST close the subflow with a RST, as it is considered
   broken.  In both cases, this RST SHOULD be accompanied by an
   MP_TCPRST option (Section 3.6) with the reason code for an "MPTCP-
   specific error".

3.3.1.  Data Sequence Mapping

   The data stream as a whole can be reassembled through the use of the
   Data Sequence Mapping components of the DSS option (Figure 9), which
   define the mapping from the subflow sequence number to the data
   sequence number.  This is used by the receiver to ensure in-order
   delivery to the application layer.  Meanwhile, the subflow-level
   sequence numbers (i.e., the regular sequence numbers in the TCP
   header) are only relevant to the subflow.  It is expected (but not
   mandated) that SACK [RFC2018] will be used at the subflow level to
   improve efficiency.

   The Data Sequence Mapping specifies a mapping from the subflow
   sequence space to the data sequence space.  This is expressed in
   terms of starting sequence numbers for the subflow and the data
   level, and a length of bytes for which this mapping is valid.  This
   explicit mapping for a range of data, rather than per-packet
   signaling, was chosen to assist with compatibility with situations
   where TCP/IP segmentation or coalescing is undertaken separately from
   the stack that is generating the data flow (e.g., through the use of
   TCP segmentation offloading on network interface cards, or by
   middleboxes such as Performance Enhancing Proxies (PEPs) [RFC3135]).
   It also allows a single mapping to cover many packets; this may be
   useful in bulk-transfer situations.

   A mapping is fixed, in that the subflow sequence number is bound to
   the data sequence number after the mapping has been processed.  A
   sender MUST NOT change this mapping after it has been declared;
   however, the same data sequence number can be mapped to by different
   subflows for retransmission purposes (see Section 3.3.6).  This would
   also permit the same data to be sent simultaneously on multiple
   subflows for resilience or efficiency purposes, especially in the
   case of lossy links.  Although the detailed specification of such
   operation is outside the scope of this document, an implementation
   SHOULD treat the first data that is received at a subflow for the
   data sequence space as the data that should be delivered to the
   application, and any subsequent data for that sequence space SHOULD
   be ignored.

   The data sequence number is specified as an absolute value, whereas
   the subflow sequence numbering is relative (the SYN at the start of
   the subflow has a relative subflow sequence number of 0).  This is
   done to allow middleboxes to change the Initial Sequence Number (ISN)
   of a subflow, such as firewalls that undertake ISN randomization.

   The Data Sequence Mapping also contains a checksum of the data that
   this mapping covers, if the use of checksums has been negotiated at
   the MP_CAPABLE exchange.  Checksums are used to detect if the payload
   has been adjusted in any way by a non-MPTCP-aware middlebox.  If this
   checksum fails, it will trigger a failure of the subflow, or a
   fallback to regular TCP, as documented in Section 3.7, since MPTCP
   can no longer reliably know the subflow sequence space at the
   receiver to build Data Sequence Mappings.  Without checksumming
   enabled, corrupt data may be delivered to the application if a
   middlebox alters segment boundaries, alters content, or does not
   deliver all segments covered by a Data Sequence Mapping.  It is
   therefore RECOMMENDED that checksumming be used, unless it is known
   that the network path contains no such devices.

   The checksum algorithm used is the standard TCP checksum [RFC0793],
   operating over the data covered by this mapping, along with a
   pseudo-header as shown in Figure 10.

                          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 Sequence Number (8 octets)               |
     |                                                              |
     |              Subflow Sequence Number (4 octets)              |
     |  Data-Level Length (2 octets) |        Zeros (2 octets)      |

                 Figure 10: Pseudo-Header for DSS Checksum

   Note that the data sequence number used in the pseudo-header is
   always the 64-bit value, irrespective of what length is used in the
   DSS option itself.  The standard TCP checksum algorithm has been
   chosen, since it will be calculated anyway for the TCP subflow, and
   if calculated first over the data before adding the pseudo-headers,
   it only needs to be calculated once.  Furthermore, since the TCP
   checksum is additive, the checksum for a DSN_MAP can be constructed
   by simply adding together the checksums for the data of each
   constituent TCP segment and adding the checksum for the DSS

   Note that checksumming relies on the TCP subflow containing
   contiguous data; therefore, a TCP subflow MUST NOT use the Urgent
   Pointer to interrupt an existing mapping.  Further note, however,
   that if Urgent data is received on a subflow, it SHOULD be mapped to
   the data sequence space and delivered to the application, analogous
   to Urgent data in regular TCP.

   To avoid possible deadlock scenarios, subflow-level processing should
   be undertaken separately from processing at the connection level.
   Therefore, even if a mapping does not exist from the subflow space to
   the data-level space, the data SHOULD still be ACKed at the subflow
   (if it is in-window).  This data cannot, however, be acknowledged at
   the data level (Section 3.3.2) because its data sequence numbers are
   unknown.  Implementations MAY hold onto such unmapped data for a
   short while, in the expectation that a mapping will arrive shortly.
   Such unmapped data cannot be counted as being within the connection-
   level receive window because this is relative to the data sequence
   numbers, so if the receiver runs out of memory to hold this data, it
   will have to be discarded.  If a mapping for that subflow-level
   sequence space does not arrive within a receive window of data, that
   subflow SHOULD be treated as broken, closed with a RST, and any
   unmapped data silently discarded.

   Data sequence numbers are always 64-bit quantities and MUST be
   maintained as such in implementations.  If a connection is
   progressing at a slow rate, so protection against wrapped sequence
   numbers is not required, then an implementation MAY include just the
   lower 32 bits of the data sequence number in the Data Sequence
   Mapping and/or Data ACK as an optimization, and an implementation can
   make this choice independently for each packet.  An implementation
   MUST be able to receive and process both 64-bit and 32-bit sequence
   number values, but it is not required that an implementation be able
   to send both.

   An implementation MUST send the full 64-bit data sequence number if
   it is transmitting at a sufficiently high rate that the 32-bit value
   could wrap within the Maximum Segment Lifetime (MSL) [RFC7323].  The
   lengths of the DSNs used in these values (which may be different) are
   declared with flags in the DSS option.  Implementations MUST accept a
   32-bit DSN and implicitly promote it to a 64-bit quantity by
   incrementing the upper 32 bits of the sequence number each time the
   lower 32 bits wrap.  A sanity check MUST be implemented to ensure
   that a wrap occurs at an expected time (e.g., the sequence number
   jumps from a very high number to a very low number) and is not
   triggered by out-of-order packets.

   As with the standard TCP sequence number, the data sequence number
   should not start at zero, but at a random value to make blind session
   hijacking harder.  This specification requires setting the IDSN of
   each host to the least significant 64 bits of the SHA-256 hash of the
   host's key, as described in Section 3.1.  This is also required in
   order for the receiver to know what the expected IDSN is and thus
   determine if any initial connection-level packets are missing; this
   is particularly relevant if two subflows start transmitting

   The mapping provided by a Data Sequence Mapping MUST apply to some or
   all of the subflow sequence space in the TCP segment that carries the
   option.  It does not need to be included in every MPTCP packet, as
   long as the subflow sequence space in that packet is covered by a
   mapping known at the receiver.  This can be used to reduce overhead
   in cases where the mapping is known in advance.  One such case is
   when there is a single subflow between the hosts, and another is when
   segments of data are scheduled in larger-than-packet-sized chunks.

   An "infinite" mapping can be used to fall back to regular TCP by
   mapping the subflow-level data to the connection-level data for the
   remainder of the connection (see Section 3.7).  This is achieved by
   setting the Data-Level Length field of the DSS option to the reserved
   value of 0.  The checksum, in such a case, will also be set to 0.

3.3.2.  Data Acknowledgments

   To provide full end-to-end resilience, MPTCP provides a connection-
   level acknowledgment, to act as a cumulative ACK for the connection
   as a whole.  This is done via the "Data ACK" field of the DSS option
   (Figure 9).  The Data ACK is analogous to the behavior of the
   standard TCP cumulative ACK -- indicating how much data has been
   successfully received (with no holes).  This can be compared to the
   subflow-level ACK, which acts in a fashion analogous to TCP SACK,
   given that there may still be holes in the data stream at the
   connection level.  The Data ACK specifies the next data sequence
   number it expects to receive.

   The Data ACK, as for the DSN, can be sent as the full 64-bit value or
   as the lower 32 bits.  If data is received with a 64-bit DSN, it MUST
   be acknowledged with a 64-bit Data ACK.  If the DSN received is
   32 bits, an implementation can choose whether to send a 32-bit or
   64-bit Data ACK, and an implementation MUST accept either in this

   The Data ACK proves that the data, and all required MPTCP signaling,
   have been received and accepted by the remote end.  One key use of
   the Data ACK signal is that it is used to indicate the left edge of
   the advertised receive window.  As explained in Section 3.3.4, the
   receive window is shared by all subflows and is relative to the Data
   ACK.  Because of this, an implementation MUST NOT use the RCV.WND
   field of a TCP segment at the connection level if it does not also
   carry a DSS option with a Data ACK field.  Furthermore, separating
   the connection-level acknowledgments from the subflow level allows
   processing to be done separately, and a receiver has the freedom to
   drop segments after acknowledgment at the subflow level -- for
   example, due to memory constraints when many segments arrive out of

   An MPTCP sender MUST NOT free data from the send buffer until it has
   been acknowledged by both a Data ACK received on any subflow and at
   the subflow level by all subflows on which the data was sent.  The
   former condition ensures liveness of the connection, and the latter
   condition ensures liveness and self-consistence of a subflow when
   data needs to be retransmitted.  Note, however, that if some data
   needs to be retransmitted multiple times over a subflow, there is a
   risk of blocking the send window.  In this case, the MPTCP sender can
   decide to terminate the subflow that is behaving badly by sending a
   RST, using an appropriate MP_TCPRST (Section 3.6) error code.

   The Data ACK MAY be included in all segments; however, optimizations
   SHOULD be considered in more advanced implementations, where the Data
   ACK is present in segments only when the Data ACK value advances, and
   this behavior MUST be treated as valid.  This behavior ensures that
   the send buffer is freed, while reducing overhead when the data
   transfer is unidirectional.

3.3.3.  Closing a Connection

   In regular TCP, a FIN announces to the receiver that the sender has
   no more data to send.  In order to allow subflows to operate
   independently and to keep the appearance of TCP over the wire, a FIN
   in MPTCP only affects the subflow on which it is sent.  This allows
   nodes to exercise considerable freedom over which paths are in use at
   any one time.  The semantics of a FIN remain as for regular TCP;
   i.e., it is not until both sides have ACKed each other's FINs that
   the subflow is fully closed.

   When an application calls close() on a socket, this indicates that it
   has no more data to send; for regular TCP, this would result in a FIN
   on the connection.  For MPTCP, an equivalent mechanism is needed;
   this is referred to as the DATA_FIN.

   A DATA_FIN is an indication that the sender has no more data to send,
   and as such it can be used to verify that all data has been
   successfully received.  A DATA_FIN, as with the FIN on a regular TCP
   connection, is a unidirectional signal.

   The DATA_FIN is signaled by setting the "F" flag in the DSS option
   (Figure 9) to 1.  A DATA_FIN occupies 1 octet (the final octet) of
   the connection-level sequence space.  Note that the DATA_FIN is
   included in the Data-Level Length but not at the subflow level: for
   example, a segment with a DSN value of 80 and a Data-Level Length of
   11, with DATA_FIN set, would map 10 octets from the subflow into data
   sequence space 80-89, and the DATA_FIN would be DSN 90; therefore,
   this segment, including DATA_FIN, would be acknowledged with a
   DATA_ACK of 91.

   Note that when the DATA_FIN is not attached to a TCP segment
   containing data, the DSS MUST have a subflow sequence number of 0, a
   Data-Level Length of 1, and the data sequence number that corresponds
   with the DATA_FIN itself.  The checksum in this case will only cover
   the pseudo-header.

   A DATA_FIN has the same semantics and behavior as a regular TCP FIN,
   but at the connection level.  Notably, it is only DATA_ACKed once all
   data has been successfully received at the connection level.  Note,
   therefore, that a DATA_FIN is decoupled from a subflow FIN.  It is
   only permissible to combine these signals on one subflow if there is
   no data outstanding on other subflows.  Otherwise, it may be
   necessary to retransmit data on different subflows.  Essentially, a
   host MUST NOT close all functioning subflows unless it is safe to do
   so, i.e., until all outstanding data has been DATA_ACKed or until the
   segment with the DATA_FIN flag set is the only outstanding segment.

   Once a DATA_FIN has been acknowledged, all remaining subflows MUST be
   closed with standard FIN exchanges.  Both hosts SHOULD send FINs on
   all subflows, as a courtesy, to allow middleboxes to clean up state
   even if an individual subflow has failed.  Reducing the timeouts
   (MSL) on subflows at end hosts after receiving a DATA_FIN is also
   encouraged.  In particular, any subflows where there is still
   outstanding data queued (which has been retransmitted on other
   subflows in order to get the DATA_FIN acknowledged) MAY be closed
   with a RST with an MP_TCPRST (Section 3.6) error code for "too much
   outstanding data".

   A connection is considered closed once both hosts' DATA_FINs have
   been acknowledged by DATA_ACKs.

   As specified above, a standard TCP FIN on an individual subflow only
   shuts down the subflow on which it was sent.  If all subflows have
   been closed with a FIN exchange but no DATA_FIN has been received and
   acknowledged, the MPTCP connection is treated as closed only after a
   timeout.  This implies that an implementation will have TIME_WAIT
   states at both the subflow level and the connection level (see
   Appendix D).  This permits "break-before-make" scenarios where
   connectivity is lost on all subflows before a new one can be

3.3.4.  Receiver Considerations

   Regular TCP advertises a receive window in each packet, telling the
   sender how much data the receiver is willing to accept past the
   cumulative ACK.  The receive window is used to implement flow
   control, throttling down fast senders when receivers cannot keep up.

   MPTCP also uses a unique receive window, shared between the subflows.
   The idea is to allow any subflow to send data as long as the receiver
   is willing to accept it.  The alternative -- maintaining per-subflow
   receive windows -- could end up stalling some subflows while others
   would not use up their window.

   The receive window is relative to the DATA_ACK.  As in TCP, a
   receiver MUST NOT shrink the right edge of the receive window (i.e.,
   DATA_ACK + receive window).  The receiver will use the data sequence
   number to tell if a packet should be accepted at the connection

   When deciding to accept packets at the subflow level, regular TCP
   checks the sequence number in the packet against the allowed receive
   window.  With MPTCP, such a check is done using only the connection-
   level window.  A sanity check SHOULD be performed at the subflow
   level to ensure that the subflow and mapped sequence numbers meet the
   following test: SSN - SUBFLOW_ACK <= DSN - DATA_ACK, where SSN is the
   subflow sequence number of the received packet and SUBFLOW_ACK is the
   RCV.NXT (next expected sequence number) of the subflow (with the
   equivalent connection-level definitions for DSN and DATA_ACK).

   In regular TCP, once a segment is deemed in-window, it is put in
   either the in-order receive queue or the out-of-order queue.  In
   Multipath TCP, the same thing happens, but at the connection level: a
   segment is placed in the connection-level in-order or out-of-order
   queue if it is in-window at both the connection level and the subflow
   level.  The stack still has to remember, for each subflow, which
   segments were received successfully so that it can ACK them at the
   subflow level appropriately.  Typically, this will be implemented by
   keeping per-subflow out-of-order queues (containing only message
   headers -- not the payloads) and remembering the value of the
   cumulative ACK.

   It is important for implementers to understand how large a receive
   buffer is appropriate.  The lower bound for full network utilization
   is the maximum bandwidth-delay product of any one of the paths.
   However, this might be insufficient when a packet is lost on a slower
   subflow and needs to be retransmitted (see Section 3.3.6).  A tight
   upper bound would be the maximum round-trip time (RTT) of any path
   multiplied by the total bandwidth available across all paths.  This
   permits all subflows to continue at full speed while a packet is
   fast-retransmitted on the maximum RTT path.  Even this might be
   insufficient to maintain full performance in the event of a
   retransmit timeout on the maximum RTT path.  Determining the
   relationship between retransmission strategies and receive buffer
   sizing is left for future study.

3.3.5.  Sender Considerations

   The sender remembers receive window advertisements from the receiver.
   It should only update its local receive window values when the
   largest sequence number allowed (i.e., DATA_ACK + receive window)
   increases on the receipt of a DATA_ACK.  This is important for
   allowing the use of paths with different RTTs and thus different
   feedback loops.

   MPTCP uses a single receive window across all subflows, and if the
   receive window was guaranteed to be unchanged end to end, a host
   could always read the most recent receive window value.  However,
   some classes of middleboxes may alter the TCP-level receive window.
   Typically, these will shrink the offered window, although for short
   periods of time it may be possible for the window to be larger
   (however, note that this would not continue for long periods, since
   ultimately the middlebox must keep up with delivering data to the
   receiver).  Therefore, if receive window sizes differ on multiple
   subflows, when sending data MPTCP SHOULD take the largest of the most
   recent window sizes as the one to use in calculations.  This rule is
   implicit in the requirement not to reduce the right edge of the

   The sender MUST also remember the receive windows advertised by each
   subflow.  The allowed window for subflow i is (ack_i, ack_i +
   rcv_wnd_i), where ack_i is the subflow-level cumulative ACK of
   subflow i.  This ensures that data will not be sent to a middlebox
   unless there is enough buffering for the data.

   Putting the two rules together, we get the following: a sender is
   allowed to send data segments with data-level sequence numbers
   between (DATA_ACK, DATA_ACK + receive_window).  Each of these
   segments will be mapped onto subflows, as long as subflow sequence
   numbers are in the allowed windows for those subflows.  Note that
   subflow sequence numbers do not generally affect flow control if the
   same receive window is advertised across all subflows.  They will
   perform flow control for those subflows with a smaller advertised
   receive window.

   The send buffer MUST, at a minimum, be as big as the receive buffer,
   to enable the sender to reach maximum throughput.

3.3.6.  Reliability and Retransmissions

   The Data Sequence Mapping allows senders to resend data with the same
   data sequence number on a different subflow.  When doing this, a host
   MUST still retransmit the original data on the original subflow, in
   order to preserve the subflow's integrity (middleboxes could replay
   old data and/or could reject holes in subflows), and a receiver will
   ignore these retransmissions.  While this is clearly suboptimal, for
   compatibility reasons this is sensible behavior.  Optimizations could
   be negotiated in future versions of this protocol.  Note also that
   this property would also permit a sender to always send the same
   data, with the same data sequence number, on multiple subflows, if
   desired for reliability reasons.

   This protocol specification does not mandate any mechanisms for
   handling retransmissions, and much will be dependent upon local
   policy (as discussed in Section 3.3.8).  One can imagine aggressive
   connection-level retransmission policies where every packet lost at
   the subflow level is retransmitted on a different subflow (hence
   wasting bandwidth but possibly reducing application-to-application
   delays) or conservative retransmission policies where connection-
   level retransmissions are only used after a few subflow-level
   retransmission timeouts occur.

   It is envisaged that a standard connection-level retransmission
   mechanism would be implemented around a connection-level data queue:
   all segments that haven't been DATA_ACKed are stored.  A timer is set
   when the head of the connection level is ACKed at the subflow level
   but is not DATA_ACKed at the data level.  This timer will guard
   against retransmission failures by middleboxes that proactively ACK

   The sender MUST keep data in its send buffer as long as the data has
   not been acknowledged both (1) at the connection level and (2) on all
   subflows on which it has been sent.  In this way, the sender can
   always retransmit the data if needed, on the same subflow or on a
   different one.  A special case is when a subflow fails: the sender
   will typically resend the data on other working subflows after a
   timeout and will keep trying to retransmit the data on the failed
   subflow too.  The sender will declare the subflow failed after a
   predefined upper bound on retransmissions is reached (which MAY be
   lower than the usual TCP limits of the MSL) or on the receipt of an
   ICMP error, and only then delete the outstanding data segments.

   If multiple retransmissions that indicate that a subflow is
   performing badly are triggered, this MAY lead to a host resetting the
   subflow with a RST.  However, additional research is required to
   understand the heuristics of how and when to reset underperforming
   subflows.  For example, a highly asymmetric path may be misdiagnosed
   as underperforming.  A RST for this purpose SHOULD be accompanied by
   an "Unacceptable performance" MP_TCPRST option (Section 3.6).

3.3.7.  Congestion Control Considerations

   Different subflows in an MPTCP connection have different congestion
   windows.  To achieve fairness at bottlenecks and resource pooling, it
   is necessary to couple the congestion windows in use on each subflow,
   in order to push most traffic to uncongested links.  One algorithm
   for achieving this is presented in [RFC6356]; the algorithm does not
   achieve perfect resource pooling but is "safe" in that it is readily
   deployable in the current Internet.  By this we mean that it does not
   take up more capacity on any one path than if it was a single path
   flow using only that route, so this ensures fair coexistence with
   single-path TCP at shared bottlenecks.

   It is foreseeable that different congestion controllers will be
   implemented for MPTCP, each aiming to achieve different properties in
   the resource pooling / fairness / stability design space, as well as
   those for achieving different properties in quality of service,
   reliability, and resilience.

   Regardless of the algorithm used, the design of MPTCP aims to provide
   the congestion control implementations with sufficient information to
   make the right decisions; this information includes, for each
   subflow, which packets were lost and when.

3.3.8.  Subflow Policy

   Within a local MPTCP implementation, a host may use any local policy
   it wishes to decide how to share the traffic to be sent over the
   available paths.

   In the typical use case, where the goal is to maximize throughput,
   all available paths will be used simultaneously for data transfer,
   using coupled congestion control as described in [RFC6356].  It is
   expected, however, that other use cases will appear.

   For instance, one possibility is an "all-or-nothing" approach, i.e.,
   have a second path ready for use in the event of failure of the first
   path, but alternatives could include entirely saturating one path
   before using an additional path (the "overflow" case).  Such choices
   would be most likely based on the monetary cost of links but may also
   be based on properties such as the delay or jitter of links, where
   stability (of delay or bandwidth) is more important than throughput.
   Application requirements such as these are discussed in detail in

   The ability to make effective choices at the sender requires full
   knowledge of the path "cost", which is unlikely to be the case.  It
   would be desirable for a receiver to be able to signal their own
   preferences for paths, since they will often be the multihomed party
   and may have to pay for metered incoming bandwidth.

   To enable this behavior, the MP_JOIN option (see Section 3.2)
   contains the "B" bit, which allows a host to indicate to its peer
   that this path should be treated as a backup path to use only in the
   event of failure of other working subflows (i.e., a subflow where the
   receiver has indicated that B=1 SHOULD NOT be used to send data
   unless there are no usable subflows where B=0).

   In the event that the available set of paths changes, a host may wish
   to signal a change in priority of subflows to the peer (e.g., a
   subflow that was previously set as a backup should now take priority
   over all remaining subflows).  Therefore, the MP_PRIO option, shown
   in Figure 11, can be used to change the "B" flag of the subflow on
   which it is sent.

                          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
     |     Kind      |     Length    |Subtype|(rsv)|B|

            Figure 11: Change Subflow Priority (MP_PRIO) Option

   Another use of the MP_PRIO option is to set the "B" flag on a subflow
   to cleanly "retire" its use before closing it and removing it with
   REMOVE_ADDR (Section 3.4.2) -- for example, to support make-before-
   break session continuity, where new subflows are added before the
   previously used subflows are closed.

   It should be noted that the backup flag is a request from a data
   receiver to a data sender only, and the data sender SHOULD adhere to
   these requests.  A host cannot assume that the data sender will do
   so, however, since local policies -- or technical difficulties -- may
   override MP_PRIO requests.  Note also that this signal applies to a
   single direction, and so the sender of this option could choose to
   continue using the subflow to send data even if it has signaled B=1
   to the other host.

3.4.  Address Knowledge Exchange (Path Management)

   We use the term "path management" to refer to the exchange of
   information about additional paths between hosts, which in this
   design is managed by multiple addresses at hosts.  For more details
   regarding the architectural thinking behind this design, see the
   MPTCP architecture document [RFC6182].

   This design makes use of two methods of sharing such information, and
   both can be used on a connection.  The first is the direct setup of
   new subflows (described in Section 3.2), where the initiator has an
   additional address.  The second method (described in the following
   subsections) signals addresses explicitly to the other host to allow
   it to initiate new subflows.  The two mechanisms are complementary:
   the first is implicit and simple, while the second (explicit) is more
   complex but is more robust.  Together, these mechanisms allow
   addresses to change in flight (and thus support operation through
   NATs, since the source address need not be known); they also allow
   the signaling of previously unknown addresses and of addresses
   belonging to other address families (e.g., both IPv4 and IPv6).

   Here is an example of typical operation of the protocol:

   *  An MPTCP connection is initially set up between address/port A1 of
      Host A and address/port B1 of Host B.  If Host A is multihomed and
      multiaddressed, it can start an additional subflow from its
      address A2 to B1, by sending a SYN with an MP_JOIN option from A2
      to B1, using B's previously declared token for this connection.
      Alternatively, if B is multihomed, it can try to set up a new
      subflow from B2 to A1, using A's previously declared token.  In
      either case, the SYN will be sent to the port already in use for
      the original subflow on the receiving host.

   *  Simultaneously (or after a timeout), an ADD_ADDR option
      (Section 3.4.1) is sent on an existing subflow, informing the
      receiver of the sender's alternative address(es).  The recipient
      can use this information to open a new subflow to the sender's
      additional address(es).  In our example, A will send the ADD_ADDR
      option informing B of address/port A2.  The mix of using the
      SYN-based option and the ADD_ADDR option, including timeouts, is
      implementation specific and can be tailored to agree with local

   *  If subflow A2-B1 is successfully set up, Host B can use the
      Address ID in the MP_JOIN option to correlate this source address
      with the ADD_ADDR option that will also arrive on an existing
      subflow; now B knows not to open A2-B1, ignoring the ADD_ADDR.
      Otherwise, if B has not received the A2-B1 MP_JOIN SYN but
      received the ADD_ADDR, it can try to initiate a new subflow from
      one or more of its addresses to address A2.  This permits new
      sessions to be opened if one host is behind a NAT.

   Other ways of using the two signaling mechanisms are possible; for
   instance, signaling addresses in other address families can only be
   done explicitly using the Add Address (ADD_ADDR) option.

3.4.1.  Address Advertisement

   The ADD_ADDR MPTCP option announces additional addresses (and,
   optionally, ports) on which a host can be reached (Figure 12).  This
   option can be used at any time during a connection, depending on when
   the sender wishes to enable multiple paths and/or when paths become
   available.  As with all MPTCP signals, the receiver MUST undertake
   standard TCP validity checks, e.g., per [RFC5961], before acting
   upon it.

                          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
     |     Kind      |     Length    |Subtype|(rsv)|E|  Address ID   |
     |           Address (IPv4: 4 octets / IPv6: 16 octets)          |
     |   Port (2 octets, optional)   |                               |
     +-------------------------------+                               |
     |                Truncated HMAC (8 octets, if E=0)              |
     |                               +-------------------------------+
     |                               |

                  Figure 12: Add Address (ADD_ADDR) Option

   Every address has an Address ID that can be used for uniquely
   identifying the address within a connection for address removal.  The
   Address ID is also used to identify MP_JOIN options (see Section 3.2)
   relating to the same address, even when address translators are in
   use.  The Address ID MUST uniquely identify the address for the
   sender of the option (within the scope of the connection); the
   mechanism for allocating such IDs is implementation specific.

   All Address IDs learned via either MP_JOIN or ADD_ADDR SHOULD be
   stored by the receiver in a data structure that gathers all the
   Address-ID-to-address mappings for a connection (identified by a
   token pair).  In this way, there is a stored mapping between the
   Address ID, observed source address, and token pair for future
   processing of control information for a connection.  Note that an
   implementation MAY discard incoming address advertisements at will --
   for example, to avoid updating mapping state or because advertised
   addresses are of no use to it (for example, IPv6 addresses when it
   has IPv4 only).  Therefore, a host MUST treat address advertisements
   as soft state, and it MAY choose to refresh advertisements
   periodically.  Note also that an implementation MAY choose to cache
   these address advertisements even if they are not currently relevant
   but may be relevant in the future, such as IPv4 addresses when IPv6
   connectivity is available but IPv4 is awaiting DHCP.

   This option is shown in Figure 12.  The illustration is sized for
   IPv4 addresses.  For IPv6, the length of the address will be
   16 octets (instead of 4).

   The 2 octets that specify the TCP port number to use are optional,
   and their presence can be inferred from the length of the option.
   Although it is expected that the majority of use cases will use the
   same port pairs as those used for the initial subflow (e.g., port 80
   remains port 80 on all subflows, as does the ephemeral port at the
   client), there may be cases (such as port-based load balancing) where
   the explicit specification of a different port is required.  If no
   port is specified, MPTCP SHOULD attempt to connect to the specified
   address on the same port as the port that is already in use by the
   subflow on which the ADD_ADDR signal was sent; this is discussed in
   more detail in Section 3.9.

   The Truncated HMAC parameter present in this option is the rightmost
   64 bits of an HMAC, negotiated and calculated in the same way as for
   MP_JOIN as described in Section 3.2.  For this specification of
   MPTCP, as there is only one hash algorithm option specified, this
   will be HMAC as defined in [RFC2104], using the SHA-256 hash
   algorithm [RFC6234].  In the same way as for MP_JOIN, the key for the
   HMAC algorithm, in the case of the message transmitted by Host A,
   will be Key-A followed by Key-B, and in the case of Host B, Key-B
   followed by Key-A.  These are the keys that were exchanged in the
   original MP_CAPABLE handshake.  The message for the HMAC is the
   Address ID, IP address, and port that precede the HMAC in the
   ADD_ADDR option.  If the port is not present in the ADD_ADDR option,
   the HMAC message will nevertheless include 2 octets of value zero.
   The rationale for the HMAC is to prevent unauthorized entities from
   injecting ADD_ADDR signals in an attempt to hijack a connection.
   Note that, additionally, the presence of this HMAC prevents the
   address from being changed in flight unless the key is known by an
   intermediary.  If a host receives an ADD_ADDR option for which it
   cannot validate the HMAC, it SHOULD silently ignore the option.

   A set of four flags is present after the subtype and before the
   Address ID.  Only the rightmost bit -- labeled "E" -- is assigned in
   this specification.  The other bits are currently unassigned; they
   MUST be set to 0 by a sender and MUST be ignored by the receiver.

   The "E" flag exists to provide reliability for this option.  Because
   this option will often be sent on pure ACKs, there is no guarantee of
   reliability.  Therefore, a receiver receiving a fresh ADD_ADDR option
   (where E=0) will send the same option back to the sender, but not
   including the HMAC and with E=1, to indicate receipt.  According to
   local policy, the lack of this type of "echo" can indicate to the
   initial ADD_ADDR sender that the ADD_ADDR needs to be retransmitted.

   Due to the proliferation of NATs, it is reasonably likely that one
   host may attempt to advertise private addresses [RFC1918].  It is not
   desirable to prohibit this behavior, since there may be cases where
   both hosts have additional interfaces on the same private network,
   and a host MAY advertise such addresses.  The MP_JOIN handshake to
   create a new subflow (Section 3.2) provides mechanisms to minimize
   security risks.  The MP_JOIN message contains a 32-bit token that
   uniquely identifies the connection to the receiving host.  If the
   token is unknown, the host will respond with a RST.  In the unlikely
   event that the token is valid at the receiving host, subflow setup
   will continue, but the HMAC exchange must occur for authentication.
   The HMAC exchange will fail and will provide sufficient protection
   against two unconnected hosts accidentally setting up a new subflow
   upon the signal of a private address.  Further security
   considerations around the issue of ADD_ADDR messages that
   accidentally misdirect, or maliciously direct, new MP_JOIN attempts
   are discussed in Section 5.

   A host that receives an ADD_ADDR but finds that a connection set up
   to that IP address and port number is unsuccessful SHOULD NOT perform
   further connection attempts to this address/port combination for this
   connection.  A sender that wants to trigger a new incoming connection
   attempt on a previously advertised address/port combination can
   therefore refresh ADD_ADDR information by sending the option again.

   A host can therefore send an ADD_ADDR message with an already-
   assigned Address ID, but the address MUST be the same as the address
   previously assigned to this Address ID.  A new ADD_ADDR may have the
   same port number or a different port number.  If the port number is
   different, the receiving host SHOULD try to set up a new subflow to
   this new address/port combination.

   A host wishing to replace an existing Address ID MUST first remove
   the existing one (Section 3.4.2).

   During normal MPTCP operation, it is unlikely that there will be
   sufficient TCP option space for ADD_ADDR to be included along with
   those for data sequence numbering (Section 3.3.1).  Therefore, it is
   expected that an MPTCP implementation will send the ADD_ADDR option
   on separate ACKs.  As discussed earlier, however, an MPTCP
   implementation MUST NOT treat duplicate ACKs with any MPTCP option,
   with the exception of the DSS option, as indications of congestion
   [RFC5681], and an MPTCP implementation SHOULD NOT send more than two
   duplicate ACKs in a row for signaling purposes.

3.4.2.  Remove Address

   If, during the lifetime of an MPTCP connection, a previously
   announced address becomes invalid (e.g., if the interface disappears
   or an IPv6 address is no longer preferred), the affected host SHOULD
   announce this situation so that the peer can remove subflows related
   to this address.  Even if an address is not in use by an MPTCP
   connection, if it has been previously announced, an implementation
   SHOULD announce its removal.  A host MAY also choose to announce that
   a valid IP address should not be used any longer -- for example, for
   make-before-break session continuity.

   This is achieved through the Remove Address (REMOVE_ADDR) option
   (Figure 13), which will remove a previously added address (or list of
   addresses) from a connection and terminate any subflows currently
   using that address.

                        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
   |     Kind      |Length = 3 + n |Subtype|(resvd)|   Address ID  | ...
                              (followed by n-1 Address IDs, if required)

               Figure 13: Remove Address (REMOVE_ADDR) Option

   For security purposes, if a host receives a REMOVE_ADDR option, it
   must ensure that the affected path or paths are no longer in use
   before it instigates closure.  The receipt of REMOVE_ADDR SHOULD
   first trigger the sending of a TCP keepalive [RFC1122] on the path,
   and if a response is received, the path SHOULD NOT be removed.  If
   the path is found to still be alive, the receiving host SHOULD no
   longer use the specified address for future connections, but it is
   the responsibility of the host that sent the REMOVE_ADDR to shut down
   the subflow.  Before the address is removed, the requesting host MAY
   also use MP_PRIO (Section 3.3.8) to request that a path no longer be
   used.  Typical TCP validity tests on the subflow (e.g., ensuring that
   sequence and ACK numbers are correct) MUST also be undertaken.  An
   implementation can use indications of these test failures as part of
   intrusion detection or error logging.

   The sending and receipt (if no keepalive response was received) of
   this message SHOULD trigger the sending of RSTs by both hosts on the
   affected subflow(s) (if possible), as a courtesy, to allow the
   cleanup of middlebox state before cleaning up any local state.

   Address removal is undertaken according to the Address ID, so as to
   permit the use of NATs and other middleboxes that rewrite source
   addresses.  If an Address ID is not known, the receiver will silently
   ignore the request.

   A subflow that is still functioning MUST be closed with a FIN
   exchange as in regular TCP, rather than using this option.  For more
   information, see Section 3.3.3.

3.5.  Fast Close

   Regular TCP has the means of sending a RST signal to abruptly close a
   connection.  With MPTCP, a regular RST only has the scope of the
   subflow; it will only close the applicable subflow and will not
   affect the remaining subflows.  MPTCP's connection will stay alive at
   the data level, in order to permit break-before-make handover between
   subflows.  It is therefore necessary to provide an MPTCP-level
   "reset" to allow the abrupt closure of the whole MPTCP connection;
   this is done via the MP_FASTCLOSE option.

   MP_FASTCLOSE is used to indicate to the peer that the connection will
   be abruptly closed and no data will be accepted anymore.  The reasons
   for triggering an MP_FASTCLOSE are implementation specific.  Regular
   TCP does not allow the sending of a RST while the connection is in a
   synchronized state [RFC0793].  Nevertheless, implementations allow
   the sending of a RST in this state if, for example, the operating
   system is running out of resources.  In these cases, MPTCP should
   send the MP_FASTCLOSE.  This option is illustrated in Figure 14.

                          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
     |     Kind      |    Length     |Subtype|      (reserved)       |
     |                      Option Receiver's Key                    |
     |                            (64 bits)                          |
     |                                                               |

                Figure 14: Fast Close (MP_FASTCLOSE) Option

   If Host A wants to force the closure of an MPTCP connection, it can
   do so via two options:

   *  Option A (ACK): Host A sends an ACK containing the MP_FASTCLOSE
      option on one subflow, containing the key of Host B as declared in
      the initial connection handshake.  On all the other subflows,
      Host A sends a regular TCP RST to close these subflows and tears
      them down.  Host A now enters FASTCLOSE_WAIT state.

   *  Option R (RST): Host A sends a RST containing the MP_FASTCLOSE
      option on all subflows, containing the key of Host B as declared
      in the initial connection handshake.  Host A can tear down the
      subflows and the connection immediately.

   If Host A decides to force the closure by using Option A and sending
   an ACK with the MP_FASTCLOSE option, the connection shall proceed as

   *  Upon receipt of an ACK with MP_FASTCLOSE by Host B, containing the
      valid key, Host B answers on the same subflow with a TCP RST and
      tears down all subflows also through sending TCP RST signals.
      Host B can now close the whole MPTCP connection (it transitions
      directly to CLOSED state).

   *  As soon as Host A has received the TCP RST on the remaining
      subflow, it can close this subflow and tear down the whole
      connection (transition from FASTCLOSE_WAIT state to CLOSED state).
      If Host A receives an MP_FASTCLOSE instead of a TCP RST, both
      hosts attempted fast closure simultaneously.  Host A should reply
      with a TCP RST and tear down the connection.

   *  If Host A does not receive a TCP RST in reply to its MP_FASTCLOSE
      after one retransmission timeout (RTO) (the RTO of the subflow
      where the MP_FASTCLOSE has been sent), it SHOULD retransmit the
      MP_FASTCLOSE.  To keep this connection from being retained for a
      long time, the number of retransmissions SHOULD be limited; this
      limit is implementation specific.  A RECOMMENDED number is 3.  If
      no TCP RST is received in response, Host A SHOULD send a TCP RST
      with the MP_FASTCLOSE option itself when it releases state in
      order to clear any remaining state at middleboxes.

   If, however, Host A decides to force the closure by using Option R
   and sending a RST with the MP_FASTCLOSE option, Host B will act as
   follows: upon receipt of a RST with MP_FASTCLOSE, containing the
   valid key, Host B tears down all subflows by sending a TCP RST.
   Host B can now close the whole MPTCP connection (it transitions
   directly to CLOSED state).

3.6.  Subflow Reset

   An implementation of MPTCP may also need to send a regular TCP RST to
   force the closure of a subflow.  A host sends a TCP RST in order to
   close a subflow or reject an attempt to open a subflow (MP_JOIN).  In
   order to let the receiving host know why a subflow is being closed or
   rejected, the TCP RST packet MAY include the MP_TCPRST option
   (Figure 15).  The host MAY use this information to decide, for
   example, whether it tries to re-establish the subflow immediately,
   later, or never.

                          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
     |     Kind      |    Length     |Subtype|U|V|W|T|    Reason     |

                Figure 15: TCP RST Reason (MP_TCPRST) Option

   The MP_TCPRST option contains a reason code that allows the sender of
   the option to provide more information about the reason for the
   termination of the subflow.  Using 12 bits of option space, the first
   4 bits are reserved for flags (only one of which is currently
   defined), and the remaining octet is used to express a reason code
   for this subflow termination, from which a receiver MAY infer
   information about the usability of this path.

   The "T" flag is used by the sender to indicate whether the error
   condition that is reported is Transient ("T" bit set to 1) or
   Permanent ("T" bit set to 0).  If the error condition is considered
   to be Transient by the sender of the RST segment, the recipient of
   this segment MAY try to re-establish a subflow for this connection
   over the failed path.  The time at which a receiver may try to
   re-establish this subflow is implementation specific but SHOULD take
   into account the properties of the failure as defined by the provided
   reason code.  If the error condition is considered to be Permanent,
   the receiver of the RST segment SHOULD NOT try to re-establish a
   subflow for this connection over this path.  The "U", "V", and "W"
   flags are not defined by this specification and are reserved for
   future use.  An implementation of this specification MUST set these
   flags to 0, and a receiver MUST ignore them.

   "Reason" is an 8-bit field that indicates the reason code for the
   termination of the subflow.  The following codes are defined in this

   *  Unspecified error (code 0x00).  This is the default error; it
      implies that the subflow is no longer available.  The presence of
      this option shows that the RST was generated by an MPTCP-aware

   *  MPTCP-specific error (code 0x01).  An error has been detected in
      the processing of MPTCP options.  This is the usual reason code to
      return in the cases where a RST is being sent to close a subflow
      because of an invalid response.

   *  Lack of resources (code 0x02).  This code indicates that the
      sending host does not have enough resources to support the
      terminated subflow.

   *  Administratively prohibited (code 0x03).  This code indicates that
      the requested subflow is prohibited by the policies of the sending

   *  Too much outstanding data (code 0x04).  This code indicates that
      there is an excessive amount of data that needs to be transmitted
      over the terminated subflow while having already been acknowledged
      over one or more other subflows.  This may occur if a path has
      been unavailable for a short period and it is more efficient to
      reset and start again than it is to retransmit the queued data.

   *  Unacceptable performance (code 0x05).  This code indicates that
      the performance of this subflow was too low compared to the other
      subflows of this Multipath TCP connection.

   *  Middlebox interference (code 0x06).  Middlebox interference has
      been detected over this subflow, making MPTCP signaling invalid.
      For example, this may be sent if the checksum does not validate.

3.7.  Fallback

   Sometimes, middleboxes will exist on a path that could prevent the
   operation of MPTCP.  MPTCP has been designed to cope with many
   middlebox modifications (see Section 6), but there are still some
   cases where a subflow could fail to operate within the MPTCP
   requirements.  Notably, these cases are the following: the loss of
   MPTCP options on a path and the modification of payload data.  If
   such an event occurs, it is necessary to "fall back" to the previous,
   safe operation.  This may be either falling back to regular TCP or
   removing a problematic subflow.

   At the start of an MPTCP connection (i.e., the first subflow), it is
   important to ensure that the path is fully MPTCP capable and the
   necessary MPTCP options can reach each host.  The handshake as
   described in Section 3.1 SHOULD fall back to regular TCP if either of
   the SYN messages does not have the MPTCP options: this is the same,
   and desired, behavior in the case where a host is not MPTCP capable
   or the path does not support the MPTCP options.  When attempting to
   join an existing MPTCP connection (Section 3.2), if a path is not
   MPTCP capable and the MPTCP options do not get through on the SYNs,
   the subflow will be closed according to the MP_JOIN logic.

   There is, however, another corner case that should be addressed: the
   case where MPTCP options get through on the SYN but not on regular
   packets.  If the subflow is the first subflow and thus all data in
   flight is contiguous, this situation can be resolved by using the
   following rules:

   *  A sender MUST include a DSS option with Data Sequence Mapping in
      every segment until one of the sent segments has been acknowledged
      with a DSS option containing a Data ACK.  Upon reception of the
      acknowledgment, the sender has the confirmation that the DSS
      option passes in both directions and may choose to send fewer DSS
      options than once per segment.

   *  If, however, an ACK is received for data (not just for the SYN)
      without a DSS option containing a Data ACK, the sender determines
      that the path is not MPTCP capable.  In the case of this occurring
      on an additional subflow (i.e., one started with MP_JOIN), the
      host MUST close the subflow with a RST, which SHOULD contain an
      MP_TCPRST option (Section 3.6) with a "Middlebox interference"
      reason code.

   *  In the case of such an ACK being received on the first subflow
      (i.e., that started with MP_CAPABLE), before any additional
      subflows are added, the implementation MUST drop out of MPTCP mode
      and fall back to regular TCP.  The sender will send one final Data
      Sequence Mapping, with the Data-Level Length value of 0 indicating
      an infinite mapping (to inform the other end in case the path
      drops options in one direction only), and then revert to sending
      data on the single subflow without any MPTCP options.

   *  If a subflow breaks during operation, e.g., if it is rerouted and
      MPTCP options are no longer permitted, then once this is detected
      (by the subflow-level receive buffer filling up, since there is no
      mapping available in order to DATA_ACK this data), the subflow
      SHOULD be treated as broken and closed with a RST, since no data
      can be delivered to the application layer and no fallback signal
      can be reliably sent.  This RST SHOULD include the MP_TCPRST
      option (Section 3.6) with a "Middlebox interference" reason code.

   These rules should cover all cases where such a failure could happen
   -- whether it's on the forward or reverse path and whether the server
   or the client first sends data.

   So far, this section has discussed the loss of MPTCP options, either
   initially or during the course of the connection.  As described in
   Section 3.3, each portion of data for which there is a mapping is
   protected by a checksum, if checksums have been negotiated.  This
   mechanism is used to detect if middleboxes have made any adjustments
   to the payload (added, removed, or changed data).  A checksum will
   fail if the data has been changed in any way.  The use of a checksum
   will also detect whether the length of data on the subflow is
   increased or decreased, and this means the Data Sequence Mapping is
   no longer valid.  The sender no longer knows what subflow-level
   sequence number the receiver is genuinely operating at (the middlebox
   will be faking ACKs in return), and it cannot signal any further
   mappings.  Furthermore, in addition to the possibility of payload
   modifications that are valid at the application layer, it is possible
   that such modifications could be triggered across MPTCP segment
   boundaries, corrupting the data.  Therefore, all data from the start
   of the segment that failed the checksum onward is not trustworthy.

   Note that if checksum usage has not been negotiated, this fallback
   mechanism cannot be used unless there is some higher-layer or
   lower-layer signal to inform the MPTCP implementation that the
   payload has been tampered with.

   When multiple subflows are in use, the data in flight on a subflow
   will likely involve data that is not contiguously part of the
   connection-level stream, since segments will be spread across the
   multiple subflows.  Due to the problems identified above, it is not
   possible to determine what adjustments have been done to the data
   (notably, any changes to the subflow sequence numbering).  Therefore,
   it is not possible to recover the subflow, and the affected subflow
   must be immediately closed with a RST that includes an MP_FAIL option
   (Figure 16), which defines the data sequence number at the start of
   the segment (defined by the Data Sequence Mapping) that had the
   checksum failure.  Note that the MP_FAIL option requires the use of
   the full 64-bit sequence number, even if 32-bit sequence numbers are
   normally in use in the DSS signals on the path.

                          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
     |     Kind      |   Length=12   |Subtype|      (reserved)      |
     |                                                              |
     |                 Data Sequence Number (8 octets)              |
     |                                                              |

                    Figure 16: Fallback (MP_FAIL) Option

   The receiver of this option MUST discard all data following the data
   sequence number specified.  Failed data MUST NOT be DATA_ACKed and so
   will be retransmitted on other subflows (Section 3.3.6).

   A special case is when there is a single subflow and it fails with a
   checksum error.  If it is known that all unacknowledged data in
   flight is contiguous (which will usually be the case with a single
   subflow), an infinite mapping can be applied to the subflow without
   the need to close it first, essentially turning off all further MPTCP
   signaling.  In this case, if a receiver identifies a checksum failure
   when there is only one path, it will send back an MP_FAIL option on
   the subflow-level ACK, referring to the data-level sequence number of
   the start of the segment on which the checksum error was detected.
   The sender will receive this information and, if all unacknowledged
   data in flight is contiguous, will signal an infinite mapping.  This
   infinite mapping will be a DSS option (Section 3.3) on the first new
   packet, containing a Data Sequence Mapping that acts retroactively,
   referring to the start of the subflow sequence number of the most
   recent segment that was known to be delivered intact (i.e., was
   successfully DATA_ACKed).  From that point onward, data can be
   altered by a middlebox without affecting MPTCP, as the data stream is
   equivalent to a regular, legacy TCP session.  While in theory paths
   may only be damaged in one direction -- and the MP_FAIL signal
   affects only one direction of traffic -- for simplicity of
   implementation, the receiver of an MP_FAIL MUST also respond with an
   MP_FAIL in the reverse direction and entirely revert to a regular TCP

   In the rare case that the data is not contiguous (which could happen
   when there is only one subflow but it is retransmitting data from a
   subflow that has recently been uncleanly closed), the receiver MUST
   close the subflow with a RST with MP_FAIL.  The receiver MUST discard
   all data that follows the data sequence number specified.  The sender
   MAY attempt to create a new subflow belonging to the same connection
   and, if it chooses to do so, SHOULD immediately place the single
   subflow in single-path mode by setting an infinite Data Sequence
   Mapping.  This mapping will begin from the data-level sequence number
   that was declared in the MP_FAIL.

   After a sender signals an infinite mapping, it MUST only use subflow
   ACKs to clear its send buffer.  This is because Data ACKs may become
   misaligned with the subflow ACKs when middleboxes insert or delete
   data.  The receiver SHOULD stop generating Data ACKs after it
   receives an infinite mapping.

   When a connection has fallen back with an infinite mapping, only one
   subflow can send data; otherwise, the receiver would not know how to
   reorder the data.  In practice, this means that all MPTCP subflows
   will have to be terminated except one.  Once MPTCP falls back to
   regular TCP, it MUST NOT revert to MPTCP later in the connection.

   It should be emphasized that MPTCP is not attempting to prevent the
   use of middleboxes that want to adjust the payload.  An MPTCP-aware
   middlebox could provide such functionality by also rewriting

3.8.  Error Handling

   In addition to the fallback mechanism described above, the standard
   classes of TCP errors may need to be handled in an MPTCP-specific
   way.  Note that changing semantics -- such as the relevance of a RST
   -- are covered in Section 4.  Where possible, we do not want to
   deviate from regular TCP behavior.

   The following list covers possible errors and the appropriate MPTCP

   *  Unknown token in MP_JOIN (or HMAC failure in MP_JOIN ACK, or
      missing MP_JOIN in SYN/ACK response): send RST (analogous to TCP's
      behavior on an unknown port)

   *  DSN out of window (during normal operation): drop the data; do not
      send Data ACKs

   *  Remove request for unknown Address ID: silently ignore

3.9.  Heuristics

   There are a number of heuristics that are needed for performance or
   deployment but that are not required for protocol correctness.  In
   this section, we detail such heuristics.  Note that discussions of
   buffering and certain sender and receiver window behaviors are
   presented in Sections 3.3.4 and 3.3.5, and retransmission is
   discussed in Section 3.3.6.

3.9.1.  Port Usage

   Under typical operation, an MPTCP implementation SHOULD use the same
   ports as the ports that are already in use.  In other words, the
   destination port of a SYN containing an MP_JOIN option SHOULD be the
   same as the remote port of the first subflow in the connection.  The
   local port for such SYNs SHOULD also be the same as the port for the
   first subflow (and as such, an implementation SHOULD reserve
   ephemeral ports across all local IP addresses), although there may be
   cases where this is infeasible.  This strategy is intended to
   maximize the probability of the SYN being permitted by a firewall or
   NAT at the recipient and to avoid confusing any network-monitoring

   There may also be cases, however, where a host wishes to signal that
   a specific port should be used; this facility is provided in the
   ADD_ADDR option as documented in Section 3.4.1.  It is therefore
   feasible to allow multiple subflows between the same two addresses
   but using different port pairs, and such a facility could be used to
   allow load balancing within the network based on 5-tuples (e.g., some
   ECMP implementations [RFC2992]).

3.9.2.  Delayed Subflow Start and Subflow Symmetry

   Many TCP connections are short-lived and consist only of a few
   segments, and so the overhead of using MPTCP outweighs any benefits.
   A heuristic is required, therefore, to decide when to start using
   additional subflows in an MPTCP connection.  Experimental deployments
   have shown that MPTCP can be applied in a range of scenarios, so an
   implementation will likely need to take into account such factors as
   the type of traffic being sent and the duration of the session; this
   information MAY be signaled by the application layer.

   However, for standard TCP traffic, a suggested general-purpose
   heuristic that an implementation MAY choose to employ is as follows.

   If a host has data buffered for its peer (which implies that the
   application has received a request for data), the host opens one
   subflow for each initial window's worth of data that is buffered.

   Consideration should also be given to limiting the rate of adding new
   subflows, as well as limiting the total number of subflows open for a
   particular connection.  A host may choose to vary these values based
   on its load or knowledge of traffic and path characteristics.

   Note that this heuristic alone is probably insufficient.  Traffic for
   many common applications, such as downloads, is highly asymmetric,
   and the host that is multihomed may well be the client that will
   never fill its buffers and thus never use MPTCP according to this
   heuristic.  Advanced APIs that allow an application to signal its
   traffic requirements would aid in these decisions.

   An additional time-based heuristic could be applied, opening
   additional subflows after a given period of time has passed.  This
   would alleviate the above issue and also provide resilience for
   low-bandwidth but long-lived applications.

   Another issue is that both communicating hosts may simultaneously try
   to set up a subflow between the same pair of addresses.  This leads
   to an inefficient use of resources.

   If the same ports are used on all subflows, as recommended above,
   then standard TCP simultaneous-open logic should take care of this
   situation and only one subflow will be established between the
   address pairs.  However, this relies on the same ports being used at
   both end hosts.  If a host does not support TCP simultaneous open, it
   is RECOMMENDED that some element of randomization be applied to the
   time to wait before opening new subflows, so that only one subflow is
   created between a given address pair.  If, however, hosts signal
   additional ports to use (for example, for leveraging ECMP on-path),
   this heuristic is not appropriate.

   This section has shown some of the factors that an implementer should
   consider when developing MPTCP heuristics, but it is not intended to
   be prescriptive.

3.9.3.  Failure Handling

   Requirements for MPTCP's handling of unexpected signals are given in
   Section 3.8.  There are other failure cases, however, where hosts can
   choose appropriate behavior.

   For example, Section 3.1 suggests that a host SHOULD fall back to
   trying regular TCP SYNs after one or more failures of MPTCP SYNs for
   a connection.  A host may keep a system-wide cache of such
   information, so that it can back off from using MPTCP, firstly for
   that particular destination host and, eventually, on a whole
   interface, if MPTCP connections continue to fail.  The duration of
   such a cache would be implementation specific.

   Another failure could occur when the MP_JOIN handshake fails.
   Section 3.8 specifies that an incorrect handshake MUST lead to the
   subflow being closed with a RST.  A host operating an active
   intrusion-detection system may choose to start blocking MP_JOIN
   packets from the source host if multiple failed MP_JOIN attempts are
   seen.  From the connection initiator's point of view, if an MP_JOIN
   fails, it SHOULD NOT attempt to connect to the same IP address and
   port during the lifetime of the connection, unless the other host
   refreshes the information with another ADD_ADDR option.  Note that
   the ADD_ADDR option is informational only and does not guarantee that
   the other host will attempt a connection.

   In addition, an implementation may learn, over a number of
   connections, that certain interfaces or destination addresses
   consistently fail and may default to not trying to use MPTCP for such
   interfaces or addresses.  The behavior of subflows that perform
   particularly badly or subflows that regularly fail during use could
   also be learned, so that an implementation can temporarily choose not
   to use these paths.

4.  Semantic Issues

   In order to support multipath operation, the semantics of some TCP
   components have changed.  To help clarify, this section lists these
   semantic changes as a point of reference.

   Sequence number:  The (in-header) TCP sequence number is specific to
      the subflow.  To allow the receiver to reorder application data,
      an additional data-level sequence space is used.  In this
      data-level sequence space, the initial SYN and the final DATA_FIN
      occupy 1 octet of sequence space.  This is done to ensure that
      these signals are acknowledged at the connection level.  There is
      an explicit mapping of data sequence space to subflow sequence
      space, which is signaled through TCP options in data packets.

   ACK:  The ACK field in the TCP header acknowledges only the subflow
      sequence number -- not the data-level sequence space.
      Implementations SHOULD NOT attempt to infer a data-level
      acknowledgment from the subflow ACKs.  This separates subflow-
      level and connection-level processing at an end host.

   Duplicate ACK:  A duplicate ACK that includes any MPTCP signaling
      (with the exception of the DSS option) MUST NOT be treated as a
      signal of congestion.  To limit the chances of non-MPTCP-aware
      entities mistakenly interpreting duplicate ACKs as a signal of
      congestion, MPTCP SHOULD NOT send more than two duplicate ACKs
      containing (non-DSS) MPTCP signals in a row.

   Receive Window:  The receive window in the TCP header indicates the
      amount of free buffer space for the whole data-level connection
      (as opposed to the amount of space for this subflow) that is
      available at the receiver.  The semantics are the same as for
      regular TCP, but to maintain these semantics the receive window
      must be interpreted at the sender as relative to the sequence
      number given in the DATA_ACK rather than the subflow ACK in the
      TCP header.  In this way, the original role of flow control is
      preserved.  Note that some middleboxes may change the receive
      window, and so a host SHOULD use the maximum value of those
      recently seen on the constituent subflows for the connection-level
      receive window and also needs to maintain a subflow-level window
      for subflow-level processing.

   FIN:  The FIN flag in the TCP header applies only to the subflow it
      is sent on -- not to the whole connection.  For connection-level
      FIN semantics, the DATA_FIN option is used.

   RST:  The RST flag in the TCP header applies only to the subflow it
      is sent on -- not to the whole connection.  The MP_FASTCLOSE
      option provides the Fast Close functionality of a RST at the MPTCP
      connection level.

   Address List:  Address list management (i.e., knowledge of the local
      and remote hosts' lists of available IP addresses) is handled on a
      per-connection basis (as opposed to per subflow, per host, or per
      pair of communicating hosts).  This permits the application of
      per-connection local policy.  Adding an address to one connection
      (either explicitly through an ADD_ADDR message or implicitly
      through an MP_JOIN) has no implications for other connections
      between the same pair of hosts.

   5-tuple:  The 5-tuple (protocol, local address, local port, remote
      address, remote port) presented by kernel APIs to the application
      layer in a non-multipath-aware application is that of the first
      subflow, even if the subflow has since been closed and removed
      from the connection.  This decision, and other related API issues,
      are discussed in more detail in [RFC6897].

5.  Security Considerations

   As identified in [RFC6181], the addition of multipath capability to
   TCP will bring with it a number of new classes of threats.  In order
   to prevent these threats, [RFC6182] presents a set of requirements
   for a security solution for MPTCP.  The fundamental goal is for the
   security of MPTCP to be "no worse" than regular TCP today.  The key
   security requirements are as follows:

   *  Provide a mechanism to confirm that the parties in a subflow
      handshake are the same as the parties in the original connection

   *  Provide verification that the peer can receive traffic at a new
      address before using it as part of a connection.

   *  Provide replay protection, i.e., ensure that a request to
      add/remove a subflow is "fresh".

   In order to achieve these goals, MPTCP includes a hash-based
   handshake algorithm, as documented in Sections 3.1 and 3.2.

   The security of the MPTCP connection hangs on the use of keys that
   are shared once at the start of the first subflow and are never sent
   again over the network (unless used in the Fast Close mechanism
   (Section 3.5)).  To ease demultiplexing while not giving away any
   cryptographic material, future subflows use a truncated cryptographic
   hash of this key as the connection identification "token".  The keys
   are concatenated and used as keys for creating Hash-based Message
   Authentication Codes (HMACs) used on subflow setup, in order to
   verify that the parties in the handshake are the same as the parties
   in the original connection setup.  It also provides verification that
   the peer can receive traffic at this new address.  Replay attacks
   would still be possible when only keys are used; therefore, the
   handshakes use single-use random numbers (nonces) at both ends --
   this ensures that the HMAC will never be the same on two handshakes.
   Guidance on generating random numbers suitable for use as keys is
   given in [RFC4086] and discussed in Section 3.1.  The nonces are
   valid for the lifetime of the TCP connection attempt.  HMAC is also
   used to secure the ADD_ADDR option, due to the threats identified in

   The use of crypto capability bits in the initial connection handshake
   to negotiate the use of a particular algorithm allows the deployment
   of additional crypto mechanisms in the future.  This negotiation
   would nevertheless be susceptible to a bid-down attack by an on-path
   active attacker who could modify the crypto capability bits in the
   response from the receiver to use a less secure crypto mechanism.
   The security mechanism presented in this document should therefore
   protect against all forms of flooding and hijacking attacks discussed
   in [RFC6181].

   The version negotiation specified in Section 3.1, if differing MPTCP
   versions shared a common negotiation format, would allow an on-path
   attacker to apply a theoretical bid-down attack.  Since the v1 and v0
   protocols have a different handshake, such an attack would require
   that the client re-establish the connection using v0 and that the
   server support v0.  Note that an on-path attacker would have access
   to the raw data, negating any other TCP-level security mechanisms.
   As also noted in Appendix E, this document specifies the removal of
   the AddrID field [RFC6824] in the MP_PRIO option (Section 3.3.8).
   This change eliminates the possibility of a theoretical attack where
   a subflow could be placed in "backup" mode by an attacker.

   During normal operation, regular TCP protection mechanisms (such as
   ensuring that sequence numbers are in-window) will provide the same
   level of protection against attacks on individual TCP subflows as the
   level of protection that exists for regular TCP today.
   Implementations will introduce additional buffers compared to regular
   TCP, to reassemble data at the connection level.  The application of
   window sizing will minimize the risk of denial-of-service attacks
   consuming resources.

   As discussed in Section 3.4.1, a host may advertise its private
   addresses, but these might point to different hosts in the receiver's
   network.  The MP_JOIN handshake (Section 3.2) will ensure that this
   does not succeed in setting up a subflow to the incorrect host.
   However, it could still create unwanted TCP handshake traffic.  This
   feature of MPTCP could be a target for denial-of-service exploits,
   with malicious participants in MPTCP connections encouraging the
   recipient to target other hosts in the network.  Therefore,
   implementations should consider heuristics (Section 3.9) at both the
   sender and receiver to reduce the impact of this.

   To further protect against malicious ADD_ADDR messages sent by an
   off-path attacker, the ADD_ADDR includes an HMAC using the keys
   negotiated during the handshake.  This effectively prevents an
   attacker from diverting an MPTCP connection through an off-path
   ADD_ADDR injection into the stream.

   A small security risk could theoretically exist with key reuse, but
   in order to accomplish a replay attack, both the sender and receiver
   keys, and the sender and receiver random numbers, in the MP_JOIN
   handshake (Section 3.2) would have to match.

   While this specification defines a "medium" security solution,
   meeting the criteria specified at the start of this section and in
   the threat analysis document [RFC6181], since attacks only ever get
   worse, it is likely that a future version of MPTCP would need to be
   able to support stronger security.  There are several ways the
   security of MPTCP could potentially be improved; some of these would
   be compatible with MPTCP as defined in this document, while others
   may not be.  For now, the best approach is to gain experience with
   the current approach, establish what might work, and check that the
   threat analysis is still accurate.

   Possible ways of improving MPTCP security could include:

   *  defining a new MPTCP cryptographic algorithm, as negotiated in
      MP_CAPABLE.  If an implementation was being deployed in a
      controlled environment where additional assumptions could be made,
      such as the ability for the servers to store state during the TCP
      handshake, then it may be possible to use a stronger cryptographic
      algorithm than would otherwise be possible.

   *  defining how to secure data transfer with MPTCP, while not
      changing the signaling part of the protocol.

   *  defining security that requires more option space, perhaps in
      conjunction with a "long options" proposal for extending the TCP
      option space (such as those surveyed in [TCPLO]), or perhaps
      building on the current approach with a second stage of security
      based on MPTCP options.

   *  revisiting the working group's decision to exclusively use TCP
      options for MPTCP signaling and instead looking at the possibility
      of using TCP payloads as well.

   MPTCP has been designed with several methods available to indicate a
   new security mechanism, including:

   *  available flags in MP_CAPABLE (Figure 4).

   *  available subtypes in the MPTCP option (Figure 3).

   *  the Version field in MP_CAPABLE (Figure 4).

6.  Interactions with Middleboxes

   Multipath TCP was designed to be deployable in the present world.
   Its design takes into account "reasonable" existing middlebox
   behavior.  In this section, we outline a few representative
   middlebox-related failure scenarios and show how Multipath TCP
   handles them.  Next, we list the design decisions Multipath TCP has
   made to accommodate the different middleboxes.

   A primary concern is our use of a new TCP option.  Middleboxes should
   forward packets with unknown options unchanged, yet there are some
   that don't.  We expect these middleboxes to strip options and pass
   the data, drop packets with new options, copy the same option into
   multiple segments (e.g., when doing segmentation), or drop options
   during segment coalescing.

   MPTCP uses a single new TCP option called "Kind", and all message
   types are defined by "subtype" values (see Section 7).  This should
   reduce the chances of only some types of MPTCP options being passed;
   instead, the key differing characteristics are different paths and
   the presence of the SYN flag.

   MPTCP SYN packets on the first subflow of a connection contain the
   MP_CAPABLE option (Section 3.1).  If this is dropped, MPTCP SHOULD
   fall back to regular TCP.  If packets with the MP_JOIN option
   (Section 3.2) are dropped, the paths will simply not be used.

   If a middlebox strips options but otherwise passes the packets
   unchanged, MPTCP will behave safely.  If an MP_CAPABLE option is
   dropped on either the outgoing path or the return path, the
   initiating host can fall back to regular TCP, as illustrated in
   Figure 17 and discussed in Section 3.1.

                Host A                              Host B
                  |              Middlebox M            |
                  |                   |                 |
                  | SYN (MP_CAPABLE)  |        SYN      |
                  |                SYN/ACK              |
              a) MP_CAPABLE option stripped on outgoing path

                Host A                                Host B
                  |           SYN (MP_CAPABLE)            |
                  |             Middlebox M               |
                  |                  |                    |
                  |    SYN/ACK       |SYN/ACK (MP_CAPABLE)|
              b) MP_CAPABLE option stripped on return path

      Figure 17: Connection Setup with Middleboxes That Strip Options
                                from Packets

   Subflow SYNs contain the MP_JOIN option.  If this option is stripped
   on the outgoing path, the SYN will appear to be a regular SYN to
   Host B.  Depending on whether there is a listening socket on the
   target port, Host B will reply with either a SYN/ACK or a RST
   (subflow connection fails).  When Host A receives the SYN/ACK, it
   sends a RST because the SYN/ACK does not contain the MP_JOIN option
   and its token.  Either way, the subflow setup fails but otherwise
   does not affect the MPTCP connection as a whole.

   We now examine data flow with MPTCP, assuming that the flow is
   correctly set up, which implies that the options in the SYN packets
   were allowed through by the relevant middleboxes.  If options are
   allowed through and there is no resegmentation or coalescing to TCP
   segments, Multipath TCP flows can proceed without problems.

   The case when options get stripped on data packets is discussed in
   Section 3.7.  If only some MPTCP options are stripped, behavior is
   not deterministic.  If some Data Sequence Mappings are lost, the
   connection can continue so long as mappings exist for the subflow-
   level data (e.g., if multiple maps have been sent that reinforce each
   other).  If some subflow-level space is left unmapped, however, the
   subflow is treated as broken and is closed, using the process
   described in Section 3.7.  MPTCP should survive with a loss of some
   Data ACKs, but performance will degrade as the fraction of stripped
   options increases.  We do not expect such cases to appear in
   practice, though: most middleboxes will either strip all options or
   let them all through.

   We end this section with a list of middlebox classes, their behavior,
   and the elements in the MPTCP design that allow operation through
   such middleboxes.  Issues surrounding dropping packets with options
   or stripping options were discussed above and are not included here:

   *  NATs (Network Address (and port) Translators) [RFC3022] change the
      source address (and often the source port) of packets.  This means
      that a host will not know its public-facing address for signaling
      in MPTCP.  Therefore, MPTCP permits implicit address addition via
      the MP_JOIN option, and the handshake mechanism ensures that
      connection attempts to private addresses [RFC1918], since they are
      authenticated, will only set up subflows to the correct hosts.
      Explicit address removal is undertaken by an Address ID to allow
      no knowledge of the source address.

   *  Performance Enhancing Proxies (PEPs) [RFC3135] might proactively
      ACK data to increase performance.  MPTCP, however, relies on
      accurate congestion control signals from the end host, and
      non-MPTCP-aware PEPs will not be able to provide such signals.
      MPTCP will, therefore, fall back to single-path TCP or close the
      problematic subflow (see Section 3.7).

   *  Traffic normalizers [norm] may not allow holes in sequence
      numbers, and they may cache packets and retransmit the same data.
      MPTCP looks like standard TCP on the wire and will not retransmit
      different data on the same subflow sequence number.  In the event
      of a retransmission, the same data will be retransmitted on the
      original TCP subflow even if it is additionally retransmitted at
      the connection level on a different subflow.

   *  Firewalls [RFC2979] might perform Initial Sequence Number (ISN)
      randomization on TCP connections.  MPTCP uses relative sequence
      numbers in Data Sequence Mappings to cope with this.  Like NATs,
      firewalls will not permit many incoming connections, so MPTCP
      supports address signaling (ADD_ADDR) so that a multiaddressed
      host can invite its peer behind the firewall/NAT to connect out to
      its additional interface.

   *  Intrusion Detection Systems / Intrusion Prevention Systems
      (IDSs/IPSs) observe packet streams for patterns and content that
      could threaten a network.  MPTCP may require the instrumentation
      of additional paths, and an MPTCP-aware IDS or IPS would need to
      read MPTCP tokens to correlate data from multiple subflows to
      maintain comparable visibility into all of the traffic between
      devices.  Without such changes, an IDS would get an incomplete
      view of the traffic, increasing the risk of missing traffic of
      interest (false negatives) and increasing the chances of
      erroneously identifying a subflow as a risk due to only seeing
      partial data (false positives).

   *  Application-level middleboxes such as content-aware firewalls may
      alter the payload within a subflow -- for example, rewriting URIs
      in HTTP traffic.  MPTCP will detect such changes using the
      checksum and close the affected subflow(s), if there are other
      subflows that can be used.  If all subflows are affected, MPTCP
      will fall back to TCP, allowing such middleboxes to change the
      payload.  MPTCP-aware middleboxes should be able to adjust the
      payload and MPTCP metadata in order not to break the connection.

   In addition, all classes of middleboxes may affect TCP traffic in the
   following ways:

   *  TCP options may be removed, or packets with unknown options
      dropped, by many classes of middleboxes.  It is intended that the
      initial SYN exchange, with a TCP option, will be sufficient to
      identify the path's capabilities.  If such a packet does not get
      through, MPTCP will end up falling back to regular TCP.

   *  Segmentation/coalescing (e.g., TCP segmentation offloading) might
      copy options between packets and might strip some options.
      MPTCP's Data Sequence Mapping includes the relative subflow
      sequence number instead of using the sequence number in the
      segment.  In this way, the mapping is independent of the packets
      that carry it.

   *  The receive window may be shrunk by some middleboxes at the
      subflow level.  MPTCP will use the maximum window at the data
      level but will also obey subflow-specific windows.

7.  IANA Considerations

   This document obsoletes [RFC6824].  As such, IANA has updated several
   registries to point to this document.  In addition, this document
   creates one new registry.  These topics are described in the
   following subsections.

7.1.  TCP Option Kind Numbers

   IANA has updated the "TCP Option Kind Numbers" registry to point to
   this document for Multipath TCP, as shown in Table 1:

           | Kind | Length |        Meaning        | Reference |
           |  30  |   N    | Multipath TCP (MPTCP) |  RFC 8684 |

                      Table 1: TCP Option Kind Numbers

7.2.  MPTCP Option Subtypes

   The 4-bit MPTCP subtype in the "MPTCP Option Subtypes" subregistry
   under the "Transmission Control Protocol (TCP) Parameters" registry
   was defined in [RFC6824].  Since [RFC6824] is an Experimental RFC and
   not a Standards Track RFC, and since no further entries have occurred
   beyond those pointing to [RFC6824], IANA has replaced the existing
   registry with the contents of Table 2 and with the following
   explanatory note.

   Note: This registry specifies the MPTCP Option Subtypes for MPTCP v1,
   which obsoletes the Experimental MPTCP v0.  For the MPTCP v0
   subtypes, please refer to [RFC6824].

     | Value |      Symbol     |         Name         |  Reference  |
     |  0x0  |    MP_CAPABLE   |  Multipath Capable   |  RFC 8684,  |
     |       |                 |                      | Section 3.1 |
     |  0x1  |     MP_JOIN     |   Join Connection    |  RFC 8684,  |
     |       |                 |                      | Section 3.2 |
     |  0x2  |       DSS       | Data Sequence Signal |  RFC 8684,  |
     |       |                 |  (Data ACK and Data  | Section 3.3 |
     |       |                 |  Sequence Mapping)   |             |
     |  0x3  |     ADD_ADDR    |     Add Address      |  RFC 8684,  |
     |       |                 |                      |   Section   |
     |       |                 |                      |    3.4.1    |
     |  0x4  |   REMOVE_ADDR   |    Remove Address    |  RFC 8684,  |
     |       |                 |                      |   Section   |
     |       |                 |                      |    3.4.2    |
     |  0x5  |     MP_PRIO     |    Change Subflow    |  RFC 8684,  |
     |       |                 |       Priority       |   Section   |
     |       |                 |                      |    3.3.8    |
     |  0x6  |     MP_FAIL     |       Fallback       |  RFC 8684,  |
     |       |                 |                      | Section 3.7 |
     |  0x7  |   MP_FASTCLOSE  |      Fast Close      |  RFC 8684,  |
     |       |                 |                      | Section 3.5 |
     |  0x8  |    MP_TCPRST    |    Subflow Reset     |  RFC 8684,  |
     |       |                 |                      | Section 3.6 |
     |  0xf  | MP_EXPERIMENTAL | Reserved for Private |             |
     |       |                 |         Use          |             |

                      Table 2: MPTCP Option Subtypes

   Values 0x9 through 0xe are currently unassigned.  Option 0xf is
   reserved for use by private experiments.  Its use may be formalized
   in a future specification.  Future assignments in this registry are
   to be defined by Standards Action as defined by [RFC8126].
   Assignments consist of the MPTCP subtype's symbolic name, its
   associated value, and a reference to its specification.

7.3.  MPTCP Handshake Algorithms

   The "MPTCP Handshake Algorithms" subregistry under the "Transmission
   Control Protocol (TCP) Parameters" registry was defined in [RFC6824].
   Since [RFC6824] is an Experimental RFC and not a Standards Track RFC,
   and since no further entries have occurred beyond those pointing to
   [RFC6824], IANA has replaced the existing registry with the contents
   of Table 3 and with the following explanatory note.

   Note: This registry specifies the MPTCP Handshake Algorithms for
   MPTCP v1, which obsoletes the Experimental MPTCP v0.  For the MPTCP
   v0 subtypes, please refer to [RFC6824].

       | Flag Bit |             Meaning             |  Reference  |
       |    A     |        Checksum required        |  RFC 8684,  |
       |          |                                 | Section 3.1 |
       |    B     |          Extensibility          |  RFC 8684,  |
       |          |                                 | Section 3.1 |
       |    C     | Do not attempt to establish new |  RFC 8684,  |
       |          | subflows to the source address. | Section 3.1 |
       |   D-G    |            Unassigned           |             |
       |    H     |           HMAC-SHA256           |  RFC 8684,  |
       |          |                                 | Section 3.2 |

                   Table 3: MPTCP Handshake Algorithms

   Note that the meanings of bits "D" through "H" can be dependent upon
   bit "B", depending on how the Extensibility parameter is defined in
   future specifications; see Section 3.1 for more information.

   Future assignments in this registry are also to be defined by
   Standards Action as defined by [RFC8126].  Assignments consist of the
   value of the flags, a symbolic name for the algorithm, and a
   reference to its specification.

7.4.  MP_TCPRST Reason Codes

   IANA has created a further subregistry, "MPTCP MP_TCPRST Reason
   Codes" under the "Transmission Control Protocol (TCP) Parameters"
   registry, based on the reason code in the MP_TCPRST (Section 3.6)
   message.  Initial values for this registry are given in Table 4;
   future assignments are to be defined by Specification Required as
   defined by [RFC8126].  Assignments consist of the value of the code,
   a short description of its meaning, and a reference to its
   specification.  The maximum value is 0xff.

      | Code |           Meaning           |       Reference       |
      | 0x00 |      Unspecified error      | RFC 8684, Section 3.6 |
      | 0x01 |     MPTCP-specific error    | RFC 8684, Section 3.6 |
      | 0x02 |      Lack of resources      | RFC 8684, Section 3.6 |
      | 0x03 | Administratively prohibited | RFC 8684, Section 3.6 |
      | 0x04 |  Too much outstanding data  | RFC 8684, Section 3.6 |
      | 0x05 |   Unacceptable performance  | RFC 8684, Section 3.6 |
      | 0x06 |    Middlebox interference   | RFC 8684, Section 3.6 |

                  Table 4: MPTCP MP_TCPRST Reason Codes

   As guidance to the designated expert [RFC8126], assignments should
   not normally be refused unless codepoint space is becoming scarce,
   provided that there is a clear distinction from other, already-
   existing codes and also provided that there is sufficient guidance
   for implementers both sending and receiving these codes.

8.  References

8.1.  Normative References

   [RFC0793]  Postel, J., "Transmission Control Protocol", STD 7,
              RFC 793, DOI 10.17487/RFC0793, September 1981,

   [RFC2104]  Krawczyk, H., Bellare, M., and R. Canetti, "HMAC: Keyed-
              Hashing for Message Authentication", RFC 2104,
              DOI 10.17487/RFC2104, February 1997,

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,

   [RFC5961]  Ramaiah, A., Stewart, R., and M. Dalal, "Improving TCP's
              Robustness to Blind In-Window Attacks", RFC 5961,
              DOI 10.17487/RFC5961, August 2010,

   [RFC6234]  Eastlake 3rd, D. and T. Hansen, "US Secure Hash Algorithms
              (SHA and SHA-based HMAC and HKDF)", RFC 6234,
              DOI 10.17487/RFC6234, May 2011,

   [RFC8174]  Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
              2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
              May 2017, <>.

8.2.  Informative References

              Bonaventure, O. and S. Seo, "Multipath TCP Deployments",
              IETF Journal 2016, November 2016,

   [howhard]  Raiciu, C., Paasch, C., Barre, S., Ford, A., Honda, M.,
              Duchene, F., Bonaventure, O., and M. Handley, "How Hard
              Can It Be? Designing and Implementing a Deployable
              Multipath TCP", Usenix Symposium on Networked Systems
              Design and Implementation 2012, April 2012,

   [norm]     Handley, M., Paxson, V., and C. Kreibich, "Network
              Intrusion Detection: Evasion, Traffic Normalization, and
              End-to-End Protocol Semantics", Usenix Security
              Symposium 2001, August 2001,

   [RFC1122]  Braden, R., Ed., "Requirements for Internet Hosts -
              Communication Layers", STD 3, RFC 1122,
              DOI 10.17487/RFC1122, October 1989,

   [RFC1918]  Rekhter, Y., Moskowitz, B., Karrenberg, D., de Groot, G.
              J., and E. Lear, "Address Allocation for Private
              Internets", BCP 5, RFC 1918, DOI 10.17487/RFC1918,
              February 1996, <>.

   [RFC2018]  Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP
              Selective Acknowledgment Options", RFC 2018,
              DOI 10.17487/RFC2018, October 1996,

   [RFC2979]  Freed, N., "Behavior of and Requirements for Internet
              Firewalls", RFC 2979, DOI 10.17487/RFC2979, October 2000,

   [RFC2992]  Hopps, C., "Analysis of an Equal-Cost Multi-Path
              Algorithm", RFC 2992, DOI 10.17487/RFC2992, November 2000,

   [RFC3022]  Srisuresh, P. and K. Egevang, "Traditional IP Network
              Address Translator (Traditional NAT)", RFC 3022,
              DOI 10.17487/RFC3022, January 2001,

   [RFC3135]  Border, J., Kojo, M., Griner, J., Montenegro, G., and Z.
              Shelby, "Performance Enhancing Proxies Intended to
              Mitigate Link-Related Degradations", RFC 3135,
              DOI 10.17487/RFC3135, June 2001,

   [RFC4086]  Eastlake 3rd, D., Schiller, J., and S. Crocker,
              "Randomness Requirements for Security", BCP 106, RFC 4086,
              DOI 10.17487/RFC4086, June 2005,

   [RFC4987]  Eddy, W., "TCP SYN Flooding Attacks and Common
              Mitigations", RFC 4987, DOI 10.17487/RFC4987, August 2007,

   [RFC5681]  Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
              Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,

   [RFC6181]  Bagnulo, M., "Threat Analysis for TCP Extensions for
              Multipath Operation with Multiple Addresses", RFC 6181,
              DOI 10.17487/RFC6181, March 2011,

   [RFC6182]  Ford, A., Raiciu, C., Handley, M., Barre, S., and J.
              Iyengar, "Architectural Guidelines for Multipath TCP
              Development", RFC 6182, DOI 10.17487/RFC6182, March 2011,

   [RFC6356]  Raiciu, C., Handley, M., and D. Wischik, "Coupled
              Congestion Control for Multipath Transport Protocols",
              RFC 6356, DOI 10.17487/RFC6356, October 2011,

   [RFC6528]  Gont, F. and S. Bellovin, "Defending against Sequence
              Number Attacks", RFC 6528, DOI 10.17487/RFC6528, February
              2012, <>.

   [RFC6824]  Ford, A., Raiciu, C., Handley, M., and O. Bonaventure,
              "TCP Extensions for Multipath Operation with Multiple
              Addresses", RFC 6824, DOI 10.17487/RFC6824, January 2013,

   [RFC6897]  Scharf, M. and A. Ford, "Multipath TCP (MPTCP) Application
              Interface Considerations", RFC 6897, DOI 10.17487/RFC6897,
              March 2013, <>.

   [RFC7323]  Borman, D., Braden, B., Jacobson, V., and R.
              Scheffenegger, Ed., "TCP Extensions for High Performance",
              RFC 7323, DOI 10.17487/RFC7323, September 2014,

   [RFC7413]  Cheng, Y., Chu, J., Radhakrishnan, S., and A. Jain, "TCP
              Fast Open", RFC 7413, DOI 10.17487/RFC7413, December 2014,

   [RFC7430]  Bagnulo, M., Paasch, C., Gont, F., Bonaventure, O., and C.
              Raiciu, "Analysis of Residual Threats and Possible Fixes
              for Multipath TCP (MPTCP)", RFC 7430,
              DOI 10.17487/RFC7430, July 2015,

   [RFC8041]  Bonaventure, O., Paasch, C., and G. Detal, "Use Cases and
              Operational Experience with Multipath TCP", RFC 8041,
              DOI 10.17487/RFC8041, January 2017,

   [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,

   [TCPLO]    Ramaiah, A., "TCP option space extension", Work in
              Progress, Internet-Draft, draft-ananth-tcpm-tcpoptext-00,
              26 March 2012, <

Appendix A.  Notes on Use of TCP Options

   The TCP option space is limited due to the length of the Data Offset
   field in the TCP header (4 bits), which defines the TCP header length
   in 32-bit words.  With the standard TCP header being 20 bytes, this
   leaves a maximum of 40 bytes for options, and many of these may
   already be used by options such as timestamp and SACK.

   We performed a brief study on the commonly used TCP options in SYN,
   data, and pure ACK packets and found that there is enough room to fit
   all the options discussed in this document.

   SYN packets typically include the following options: Maximum Segment
   Size (MSS) (4 bytes), window scale (3 bytes), SACK permitted
   (2 bytes), and timestamp (10 bytes).  The sum of these options is
   19 bytes.  Some operating systems appear to pad each option up to a
   word boundary, thus using 24 bytes (a brief survey suggests that
   Windows XP and Mac OS X do this, whereas Linux does not).
   Optimistically, therefore, we have 21 bytes available, or 16 if
   options have to be word-aligned.  In either case, however, the SYN
   versions of MP_CAPABLE (12 bytes) and MP_JOIN (12 or 16 bytes) will
   fit in this remaining space.

   Note that due to the use of a 64-bit data-level sequence space, it is
   feasible that MPTCP will not require the timestamp option for
   protection against wrapped sequence numbers (per the Protection
   Against Wrapped Sequences (PAWS) mechanism, as described in
   [RFC7323]), since the data-level sequence space has far less chance
   of wrapping.  Confirmation of the validity of this optimization is
   left for further study.

   TCP data packets typically carry timestamp options in every packet,
   taking 10 bytes (or 12, with padding).  That leaves 30 bytes (or 28,
   if word-aligned).  The DSS option varies in length, depending on
   (1) whether the Data Sequence Mapping, DATA_ACK, or both are
   included, (2) whether the sequence numbers in use are 4 or 8 octets,
   and (3) whether the checksum is present.  The maximum size of the DSS
   option is 28 bytes, so even that will fit in the available space.
   But unless a connection is both bidirectional and high-bandwidth, it
   is unlikely that all that option space will be required on each DSS

   Within the DSS option, it is not necessary to include the Data
   Sequence Mapping and DATA_ACK in each packet, and in many cases it
   may be possible to alternate their presence (so long as the mapping
   covers the data being sent in the subsequent packet).  It would also
   be possible to alternate between 4-byte and 8-byte sequence numbers
   in each option.

   On subflow and connection setup, an MPTCP option is also set on the
   third packet (an ACK).  These are 20 bytes (for MP_CAPABLE) and
   24 bytes (for MP_JOIN), both of which will fit in the available
   option space.

   Pure ACKs in TCP typically contain only timestamps (10 bytes).  Here,
   Multipath TCP typically needs to encode only the DATA_ACK (maximum of
   12 bytes).  Occasionally, ACKs will contain SACK information.
   Depending on the number of lost packets, SACK may utilize the entire
   option space.  If a DATA_ACK had to be included, then it is probably
   necessary to reduce the number of SACK blocks to accommodate the
   DATA_ACK.  However, the presence of the DATA_ACK is unlikely to be
   necessary in a case where SACK is in use, since until at least some
   of the SACK blocks have been retransmitted, the cumulative data-level
   ACK will not be moving forward (or if it does, due to retransmissions
   on another path, then that path can also be used to transmit the new

   The ADD_ADDR option can be between 16 and 30 bytes, depending on
   (1) whether IPv4 or IPv6 is used and (2) whether or not the port
   number is present.  It is unlikely that such signaling would fit in a
   data packet (although if there is space, it is fine to include it).
   It is recommended that duplicate ACKs not be used with any other
   payload or options, in order to transmit these rare signals.  Note
   that this is the reason for mandating that duplicate ACKs with MPTCP
   options not be taken as a signal of congestion.

Appendix B.  TCP Fast Open and MPTCP

   TCP Fast Open (TFO) is an experimental TCP extension, described in
   [RFC7413], which has been introduced to allow the sending of data one
   RTT earlier than with regular TCP.  This is considered a valuable
   gain, as very short connections are very common, especially for HTTP
   request/response schemes.  It achieves this by sending the SYN
   segment together with the application's data and allowing the
   listener to reply immediately with data after the SYN/ACK.  [RFC7413]
   secures this mechanism by using a new TCP option that includes a
   cookie that is negotiated in a preceding connection.

   When using TFO in conjunction with MPTCP, there are two key points to
   take into account, as detailed below.

B.1.  TFO Cookie Request with MPTCP

   When a TFO initiator first connects to a listener, it cannot
   immediately include data in the SYN for security reasons [RFC7413].
   Instead, it requests a cookie that will be used in subsequent
   connections.  This is done with the TCP cookie request/response
   options, of 2 bytes and 6-18 bytes, respectively (depending on the
   chosen cookie length).

   TFO and MPTCP can be combined, provided that the total length of all
   the options does not exceed the maximum 40 bytes possible in TCP:

   *  In the SYN: MPTCP uses a 4-byte MP_CAPABLE option.  The sum of the
      MPTCP and TFO options is 6 bytes.  With typical TCP options using
      up to 19 bytes in the SYN (24 bytes if options are padded at a
      word boundary), there is enough space to combine the MP_CAPABLE
      with the TFO cookie request.

   *  In the SYN + ACK: MPTCP uses a 12-byte MP_CAPABLE option, but now
      the TFO option can be as long as 18 bytes.  Since the maximum
      option length may be exceeded, it is up to the listener to avoid
      this problem by using a shorter cookie.  As an example, if we
      consider that 19 bytes are used for classical TCP options, the
      maximum possible cookie length would be 7 bytes.  Note that, for
      the SYN packet, the same limitation applies to subsequent
      connections (because the initiator then echoes the cookie back to
      the listener).  Finally, if the security impact of reducing the
      cookie size is not deemed acceptable, the listener can reduce the
      amount of space used by other TCP options by omitting the TCP
      timestamps (as outlined in Appendix A).

B.2.  Data Sequence Mapping under TFO

   In the TCP establishment phase, MPTCP uses a key exchange that is
   used to generate the Initial Data Sequence Numbers (IDSNs).  In
   particular, the SYN with MP_CAPABLE occupies the first octet of data
   sequence space.  With TFO, one way to handle the data sent together
   with the SYN would be to consider an implicit DSS mapping that covers
   that SYN segment (since there is not enough space in the SYN to
   include a DSS option).  The problem with that approach is that if a
   middlebox modifies the TFO data, this will not be noticed by MPTCP
   because of the absence of a DSS checksum.  For example, a TCP-aware
   (but not MPTCP-aware) middlebox could insert bytes at the beginning
   of the stream and adapt the TCP checksum and sequence numbers
   accordingly.  With an implicit mapping, this information would give
   to the initiator and listener a different view of the DSS mapping;
   there would be no way to detect this inconsistency, because the DSS
   checksum is not present.

   To solve this issue, the TFO data must not be considered part of the
   data sequence number space: the SYN with MP_CAPABLE still occupies
   the first octet of data sequence space, but then the first non-TFO
   data byte occupies the second octet.  This guarantees that, if the
   use of the DSS checksum is negotiated, all data in the data sequence
   number space is checksummed.  We also note that this does not entail
   a loss of functionality, because TFO data is always only sent on the
   initial subflow, before any attempt to create additional subflows.

B.3.  Connection Establishment Examples

   A few examples of possible "TFO + MPTCP" establishment scenarios are
   shown below.

   Before an initiator can send data together with the SYN, it must
   request a cookie from the listener, as shown in Figure 18.  (Note:
   The sequence number and length are annotated in Figure 18 as
   Seq(Length) (e.g., "S. 0(0)") and used as such in the subsequent
   figures (e.g., "S  0(20)" in Figure 19).)  This is done by simply
   combining the TFO and MPTCP options.

   initiator                                                    listener
       |                                                           |
       |   S Seq=0(Length=0) <MP_CAPABLE>, <TFO cookie request>    |
       | --------------------------------------------------------> |
       |                                                           |
       |   S. 0(0) ack 1 <MP_CAPABLE>, <TFO cookie>                |
       | <-------------------------------------------------------- |
       |                                                           |
       |   .  0(0) ack 1 <MP_CAPABLE>                              |
       | --------------------------------------------------------> |
       |                                                           |

                         Figure 18: Cookie Request

   Once this is done, the received cookie can be used for TFO, as shown
   in Figure 19.  In this example, the initiator first sends 20 bytes in
   the SYN.  The listener immediately replies with 100 bytes following
   the SYN-ACK, to which the initiator replies with 20 more bytes.  Note
   that the last segment in the figure has a TCP sequence number of 21,
   while the DSS subflow sequence number is 1 (because the TFO data is
   not part of the data sequence number space, as explained in
   Appendix B.2.

   initiator                                                    listener 
       |                                                           |
       |    S  0(20) <MP_CAPABLE>, <TFO cookie>                    |
       | --------------------------------------------------------> |
       |                                                           |
       |    S. 0(0) ack 21 <MP_CAPABLE>                            |
       | <-------------------------------------------------------- |
       |                                                           |
       |    .  1(100) ack 21 <DSS seq=1 ssn=1 dlen=100, M=1, A=0>  |
       | <-------------------------------------------------------- |
       |                                                           |
       |    .  21(0) ack 1 <MP_CAPABLE>                            |
       | --------------------------------------------------------> |
       |                                                           |
       |    .  21(20) ack 101 <DSS ack=101 seq=1 ssn=1 dlen=20>    |
       | --------------------------------------------------------> |
       |                                                           |

                    Figure 19: The Listener Supports TFO
EID 6609 (Verified) is as follows:

Section: B.3

Original Text:

   initiator                                                    listener
       |                                                           |
       |    S  0(20) <MP_CAPABLE>, <TFO cookie>                    |
       | --------------------------------------------------------> |
       |                                                           |
       |    S. 0(0) ack 21 <MP_CAPABLE>                            |
       | <-------------------------------------------------------- |
       |                                                           |
       |    .  1(100) ack 21 <DSS ack=1 seq=1 ssn=1 dlen=100>      |
       | <-------------------------------------------------------- |
       |                                                           |
       |    .  21(0) ack 1 <MP_CAPABLE>                            |
       | --------------------------------------------------------> |
       |                                                           |
       |    .  21(20) ack 101 <DSS ack=101 seq=1 ssn=1 dlen=20>    |
       | --------------------------------------------------------> |
       |                                                           |

                    Figure 19: The Listener Supports TFO

Corrected Text:

initiator                                                    listener
       |                                                           |
       |    S  0(20) <MP_CAPABLE>, <TFO cookie>                    |
       | --------------------------------------------------------> |
       |                                                           |
       |    S. 0(0) ack 21 <MP_CAPABLE>                            |
       | <-------------------------------------------------------- |
       |                                                           |
       |    .  1(100) ack 21 <DSS seq=1 ssn=1 dlen=100, M=1, A=0>  |
       | <-------------------------------------------------------- |
       |                                                           |
       |    .  21(0) ack 1 <MP_CAPABLE>                            |
       | --------------------------------------------------------> |
       |                                                           |
       |    .  21(20) ack 101 <DSS ack=101 seq=1 ssn=1 dlen=20>    |
       | --------------------------------------------------------> |
       |                                                           |

                    Figure 19: The Listener Supports TFO
The example for TCP Fastopen with MPTCPv1 in RFC8684, Appendix-B.3 shows the listener sending an incorrect data-ack in its first data packet to the initiator sent immediately after its SYN-ACK.
At this point, the listener has not received the initiator's key, so it cannot generate the data-ack in it's response packets.
The correct behaviour would be to set flag 'A' to 0 and exclude sending data-ack until the initiator's key is received.
In Figure 20, the listener does not support TFO. The initiator detects that no state is created in the listener (as no data is ACKed) and now sends the MP_CAPABLE in the third packet, in order for the listener to build its MPTCP context at the end of the establishment. Now, the TFO data, when retransmitted, becomes part of the Data Sequence Mapping because it is effectively sent (in fact re-sent) after the establishment. initiator listener | | | S 0(20) <MP_CAPABLE>, <TFO cookie> | | --------------------------------------------------------> | | | | S. 0(0) ack 1 <MP_CAPABLE> | | <-------------------------------------------------------- | | | | . 1(0) ack 1 <MP_CAPABLE> | | --------------------------------------------------------> | | | | . 1(20) ack 1 <DSS ack=1 seq=1 ssn=1 dlen=20> | | --------------------------------------------------------> | | | | . 0(0) ack 21 <DSS ack=21 seq=1 ssn=1 dlen=0> | | <-------------------------------------------------------- | | | Figure 20: The Listener Does Not Support TFO It is also possible that the listener acknowledges only part of the TFO data, as illustrated in Figure 21. The initiator will simply retransmit the missing data together with a DSS mapping. initiator listener | | | S 0(1000) <MP_CAPABLE>, <TFO cookie> | | --------------------------------------------------------> | | | | S. 0(0) ack 501 <MP_CAPABLE> | | <-------------------------------------------------------- | | | | . 501(0) ack 1 <MP_CAPABLE> | | --------------------------------------------------------> | | | | . 501(500) ack 1 <DSS ack=1 seq=1 ssn=1 dlen=500> | | --------------------------------------------------------> | | | Figure 21: Partial Data Acknowledgment Appendix C. Control Blocks Conceptually, an MPTCP connection can be represented as an MPTCP protocol control block (PCB) that contains several variables that track the progress and the state of the MPTCP connection and a set of linked TCP control blocks that correspond to the subflows that have been established. RFC 793 [RFC0793] specifies several state variables. Whenever possible, we reuse the same terminology as RFC 793 to describe the state variables that are maintained by MPTCP. C.1. MPTCP Control Block The MPTCP control block contains the following variables per connection. C.1.1. Authentication and Metadata Local.Token (32 bits): This is the token chosen by the local host on this MPTCP connection. The token must be unique among all established MPTCP connections and is generated from the local key. Local.Key (64 bits): This is the key sent by the local host on this MPTCP connection. Remote.Token (32 bits): This is the token chosen by the remote host on this MPTCP connection, generated from the remote key. Remote.Key (64 bits): This is the key chosen by the remote host on this MPTCP connection. MPTCP.Checksum (flag): This flag is set to true if at least one of the hosts has set the "A" bit in the MP_CAPABLE options exchanged during connection establishment; otherwise, it is set to false. If this flag is set, the checksum must be computed in all DSS options. C.1.2. Sending Side SND.UNA (64 bits): This is the data sequence number of the next byte to be acknowledged, at the MPTCP connection level. This variable is updated upon reception of a DSS option containing a DATA_ACK. SND.NXT (64 bits): This is the data sequence number of the next byte to be sent. SND.NXT is used to determine the value of the DSN in the DSS option. SND.WND (32 bits): This is the send window. 32 bits if the features in RFC 7323 are used; 16 bits otherwise. MPTCP maintains the send window at the MPTCP connection level, and the same window is shared by all subflows. All subflows use the MPTCP connection- level SND.WND to compute the SEQ.WND value that is sent in each transmitted segment. C.1.3. Receiving Side RCV.NXT (64 bits): This is the data sequence number of the next byte that is expected on the MPTCP connection. This state variable is modified upon reception of in-order data. The value of RCV.NXT is used to specify the DATA_ACK that is sent in the DSS option on all subflows. RCV.WND (32 bits): This is the connection-level receive window, which is the maximum of the RCV.WND on all the subflows. 32 bits if the features in RFC 7323 are used; 16 bits otherwise. C.2. TCP Control Blocks The MPTCP control block also contains a list of the TCP control blocks that are associated with the MPTCP connection. Note that the TCP control block on the TCP subflows does not contain the RCV.WND and SND.WND state variables, as these are maintained at the MPTCP connection level and not at the subflow level. Inside each TCP control block, the following state variables are defined. C.2.1. Sending Side SND.UNA (32 bits): This is the sequence number of the next byte to be acknowledged on the subflow. This variable is updated upon reception of each TCP acknowledgment on the subflow. SND.NXT (32 bits): This is the sequence number of the next byte to be sent on the subflow. SND.NXT is used to set the value of SEG.SEQ upon transmission of the next segment. C.2.2. Receiving Side RCV.NXT (32 bits): This is the sequence number of the next byte that is expected on the subflow. This state variable is modified upon reception of in-order segments. The value of RCV.NXT is copied to the SEG.ACK field of the next segments transmitted on the subflow. RCV.WND (32 bits): This is the subflow-level receive window that is updated with the window field from the segments received on this subflow. 32 bits if the features in RFC 7323 are used; 16 bits otherwise. Appendix D. Finite State Machine The diagram in Figure 22 shows the Finite State Machine for connection-level closure. This illustrates how the DATA_FIN connection-level signal (indicated in the diagram as the DFIN flag on a DATA_ACK) (1) interacts with subflow-level FINs and (2) permits break-before-make handover between subflows. +---------+ | M_ESTAB | +---------+ M_CLOSE | | rcv DATA_FIN ------- | | ------- +---------+ snd DATA_FIN / \ snd DATA_ACK[DFIN] +-------+ | M_FIN |<----------------- ------------------->|M_CLOSE| | WAIT-1 |--------------------------- | WAIT | +---------+ rcv DATA_FIN \ +-------+ | rcv DATA_ACK[DFIN] ------- | M_CLOSE | | -------------- snd DATA_ACK | ------- | | CLOSE all subflows | snd DATA_FIN | V V V +-----------+ +-----------+ +----------+ |M_FINWAIT-2| | M_CLOSING | |M_LAST-ACK| +-----------+ +-----------+ +----------+ | rcv DATA_ACK[DFIN] | rcv DATA_ACK[DFIN] | | rcv DATA_FIN -------------- | -------------- | | ------- CLOSE all subflows | CLOSE all subflows | | snd DATA_ACK[DFIN] V delete MPTCP PCB V \ +-----------+ +--------+ ------------------------>|M_TIME WAIT|---------------->|M_CLOSED| +-----------+ +--------+ All subflows in CLOSED ------------ delete MPTCP PCB Figure 22: Finite State Machine for Connection Closure Appendix E. Changes from RFC 6824 This appendix lists the key technical changes between [RFC6824], which specifies MPTCP v0; and this document, which obsoletes [RFC6824] and specifies MPTCP v1. Note that this specification is not backward compatible with [RFC6824]. * This document incorporates lessons learned from the various implementations, deployments, and experiments gathered in the documents "Use Cases and Operational Experience with Multipath TCP" [RFC8041] and the IETF Journal article "Multipath TCP Deployments" [deployments]. * Connection initiation, through the exchange of the MP_CAPABLE MPTCP option, is different from [RFC6824]. The SYN no longer includes the initiator's key, to allow the MP_CAPABLE option on the SYN to be shorter in length and to avoid duplicating the sending of keying material. * This also ensures reliable delivery of the key on the MP_CAPABLE option by allowing its transmission to be combined with data and thus using TCP's built-in reliability mechanism. If the initiator does not immediately have data to send, the MP_CAPABLE option with the keys will be repeated on the first data packet. If the other end is the first to send, then the presence of the DSS option implicitly confirms the receipt of the MP_CAPABLE. * In the Flags field of MP_CAPABLE, "C" is now assigned to mean that the sender of this option will not accept additional MPTCP subflows to the source address and port. This improves efficiency -- for example, in cases where the sender is behind a strict NAT. * In the Flags field of MP_CAPABLE, "H" now indicates the use of HMAC-SHA256 (rather than HMAC-SHA1). * Connection initiation also defines the procedure for version negotiation, for implementations that support both v0 [RFC6824] and v1 (this document). * The HMAC-SHA256 (rather than HMAC-SHA1) algorithm is used, as it provides better security. It is used to generate the token in the MP_JOIN and ADD_ADDR messages and to set the IDSN. * A new subflow-level option exists to signal reasons for sending a RST on a subflow (MP_TCPRST (Section 3.6)); this can help an implementation decide whether to attempt later reconnection. * The MP_PRIO option (Section 3.3.8), which is used to signal a change of priority for a subflow, no longer includes the AddrID field. Its purpose was to allow the changed priority to be applied on a subflow other than the one it was sent on. However, it was determined that this could be used by a man-in-the-middle to divert all traffic onto its own path, and MP_PRIO does not include a token or other type of security mechanism. * The ADD_ADDR option (Section 3.4.1), which is used to inform the other host about another potential address, is different in several ways. It now includes an HMAC of the added address, for enhanced security. In addition, reliability for the ADD_ADDR option has been added: the IPVer field is replaced with a flag field, and one flag is assigned ("E") that is used as an "echo" so a host can indicate that it has received the option. * This document describes an additional way of performing a Fast Close -- by sending an MP_FASTCLOSE option on a RST on all subflows. This allows the host to tear down the subflows and the connection immediately. * IANA has reserved the MPTCP option subtype of value 0xf for Private Use (Section 7.2). This document doesn't define how to use that value. * This document adds a new appendix (Appendix B), which discusses the usage of both MPTCP options and TFO options on the same packet. Acknowledgments The authors gratefully acknowledge significant input into this document from Sebastien Barre and Andrew McDonald. The authors also wish to acknowledge reviews and contributions from Iljitsch van Beijnum, Lars Eggert, Marcelo Bagnulo, Robert Hancock, Pasi Sarolahti, Toby Moncaster, Philip Eardley, Sergio Lembo, Lawrence Conroy, Yoshifumi Nishida, Bob Briscoe, Stein Gjessing, Andrew McGregor, Georg Hampel, Anumita Biswas, Wes Eddy, Alexey Melnikov, Francis Dupont, Adrian Farrel, Barry Leiba, Robert Sparks, Sean Turner, Stephen Farrell, Martin Stiemerling, Gregory Detal, Fabien Duchene, Xavier de Foy, Rahul Jadhav, Klemens Schragel, Mirja Kühlewind, Sheng Jiang, Alissa Cooper, Ines Robles, Roman Danyliw, Adam Roach, Eric Vyncke, and Ben Kaduk. Authors' Addresses Alan Ford Pexip Email: Costin Raiciu University Politehnica of Bucharest Splaiul Independentei 313 Bucharest Romania Email: Mark Handley University College London Gower Street London WC1E 6BT United Kingdom Email: Olivier Bonaventure Université catholique de Louvain Pl. Ste Barbe, 2 1348 Louvain-la-Neuve Belgium Email: Christoph Paasch Apple, Inc. Cupertino, CA United States of America Email: