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IETF RFC 8684
Last modified on Tuesday, March 31st, 2020
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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
 Abstract
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
https://www.rfc-editor.org/info/RFC 8684.
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
(https://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction
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
Acknowledgments
Authors' Addresses
1. Introduction
Multipath TCP (MPTCP) is a set of extensions to regular TCP [RFC 793]
to provide a Multipath TCP service [RFC 6182], 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:
* [RFC 6182] (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.
* [RFC 6356] (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.
* [RFC 6897] (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
[RFC 6824]. 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
versions.
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
multiaddressed.
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 [RFC 6356] 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 [RFC 2992]), 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 [RFC 6182]):
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 [RFC 6897]
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 [RFC 6182] 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
[RFC 6897].
+-------------------------------+
| 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 [RFC 6897]. 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,
respectively.
* 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",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
BCP 14 [RFC 2119] [RFC 8174] 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
messages.
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 ->
[flags]
<- 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 ->
[flags]
<- 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 ->
[flags]
<- 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 ->
[flags]
<- 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 ->
[Echo-flag=0,
IP#-A2,
IP#-A2's Address ID,
HMAC of IP#-A2]
<- ADD_ADDR
[Echo-flag=1,
IP#-A2,
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]
[Checksum]
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
message.
* 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 [RFC 6181], the following
steps are taken: keys are sent in the clear in the MP_CAPABLE
messages; MP_JOIN messages are secured with HMAC-SHA256 ([RFC 2104]
using the algorithm in [RFC 6234]) using those keys; and standard
TCP validity checks are made on the other messages (ensuring that
sequence numbers are in-window [RFC 5961]). Residual threats to
MPTCP v0 were identified in [RFC 7430], 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) [RFC 2018]) 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 [RFC 5681] 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
[RFC 5961], and initial subflow sequence numbers SHOULD be generated
according to the recommendations in [RFC 6528].
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
connection.
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
attempts.
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 [RFC 4086]. 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
connection.
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
service.
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 [RFC 4987]). 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
MP_CAPABLE in the SYN/ACK.
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
mappings.
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
payload).
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 [RFC 6234] 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 [RFC 6234].
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
handshake.
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
format.
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 [RFC 6234], 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 [RFC 4086].
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
time.
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 [RFC 2104] is used, along with the SHA-256
hash algorithm [RFC 6234], 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
included.
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
sent.
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
subsections.
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
specification.
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 [RFC 2018] 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) [RFC 3135]).
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 [RFC 793],
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
pseudo-header.
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) [RFC 7323]. 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
simultaneously.
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
situation.
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
order.
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
re-established.
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
level.
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
window.
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
data.
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 [RFC 6356]; 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 [RFC 6356]. 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
[RFC 6897].
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 [RFC 6182].
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
policy.
* 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 [RFC 5961], 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 [RFC 2104], using the SHA-256 hash
algorithm [RFC 6234]. 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 [RFC 1918]. 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
[RFC 5681], 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 [RFC 1122] 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 [RFC 793]. 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
follows:
* 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
document:
* 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
device.
* 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
host.
* 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
session.
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
checksums.
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
behavior:
* 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
software.
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 [RFC 2992]).
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 [RFC 6897].
5. Security Considerations
As identified in [RFC 6181], the addition of multipath capability to
TCP will bring with it a number of new classes of threats. In order
to prevent these threats, [RFC 6182] 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
setup.
* 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 [RFC 4086] 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
[RFC 7430].
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 [RFC 6181].
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 [RFC 6824] 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 [RFC 6181], 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) [RFC 3022] 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 [RFC 1918], 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) [RFC 3135] 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 [RFC 2979] 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 [RFC 6824]. 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 [RFC 6824]. Since [RFC 6824] is an Experimental RFC and
not a Standards Track RFC, and since no further entries have occurred
beyond those pointing to [RFC 6824], 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 [RFC 6824].
+-------+-----------------+----------------------+-------------+
| 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 [RFC 8126].
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 [RFC 6824].
Since [RFC 6824] is an Experimental RFC and not a Standards Track RFC,
and since no further entries have occurred beyond those pointing to
[RFC 6824], 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 [RFC 6824].
+----------+---------------------------------+-------------+
| 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 [RFC 8126]. 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 [RFC 8126]. 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 [RFC 8126], 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
[RFC 793] Postel, J., "Transmission Control Protocol", STD 7,
RFC 793, DOI 10.17487/RFC 793, September 1981,
<https://www.rfc-editor.org/info/RFC 793>.
[RFC 2104] Krawczyk, H., Bellare, M., and R. Canetti, "HMAC: Keyed-
Hashing for Message Authentication", RFC 2104,
DOI 10.17487/RFC 2104, February 1997,
<https://www.rfc-editor.org/info/RFC 2104>.
[RFC 2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC 2119, March 1997,
<https://www.rfc-editor.org/info/RFC 2119>.
[RFC 5961] Ramaiah, A., Stewart, R., and M. Dalal, "Improving TCP's
Robustness to Blind In-Window Attacks", RFC 5961,
DOI 10.17487/RFC 5961, August 2010,
<https://www.rfc-editor.org/info/RFC 5961>.
[RFC 6234] Eastlake 3rd, D. and T. Hansen, "US Secure Hash Algorithms
(SHA and SHA-based HMAC and HKDF)", RFC 6234,
DOI 10.17487/RFC 6234, May 2011,
<https://www.rfc-editor.org/info/RFC 6234>.
[RFC 8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC 8174,
May 2017, <https://www.rfc-editor.org/info/RFC 8174>.
8.2. Informative References
[deployments]
Bonaventure, O. and S. Seo, "Multipath TCP Deployments",
IETF Journal 2016, November 2016,
<https://www.ietfjournal.org/multipath-tcp-deployments/>.
[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,
<https://www.usenix.org/conference/nsdi12/technical-
sessions/presentation/raiciu>.
[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,
<https://www.usenix.org/legacy/events/sec01/full_papers/
handley/handley.pdf>.
[RFC 1122] Braden, R., Ed., "Requirements for Internet Hosts -
Communication Layers", STD 3, RFC 1122,
DOI 10.17487/RFC 1122, October 1989,
<https://www.rfc-editor.org/info/RFC 1122>.
[RFC 1918] 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/RFC 1918,
February 1996, <https://www.rfc-editor.org/info/RFC 1918>.
[RFC 2018] Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP
Selective Acknowledgment Options", RFC 2018,
DOI 10.17487/RFC 2018, October 1996,
<https://www.rfc-editor.org/info/RFC 2018>.
[RFC 2979] Freed, N., "Behavior of and Requirements for Internet
Firewalls", RFC 2979, DOI 10.17487/RFC 2979, October 2000,
<https://www.rfc-editor.org/info/RFC 2979>.
[RFC 2992] Hopps, C., "Analysis of an Equal-Cost Multi-Path
Algorithm", RFC 2992, DOI 10.17487/RFC 2992, November 2000,
<https://www.rfc-editor.org/info/RFC 2992>.
[RFC 3022] Srisuresh, P. and K. Egevang, "Traditional IP Network
Address Translator (Traditional NAT)", RFC 3022,
DOI 10.17487/RFC 3022, January 2001,
<https://www.rfc-editor.org/info/RFC 3022>.
[RFC 3135] 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/RFC 3135, June 2001,
<https://www.rfc-editor.org/info/RFC 3135>.
[RFC 4086] Eastlake 3rd, D., Schiller, J., and S. Crocker,
"Randomness Requirements for Security", BCP 106, RFC 4086,
DOI 10.17487/RFC 4086, June 2005,
<https://www.rfc-editor.org/info/RFC 4086>.
[RFC 4987] Eddy, W., "TCP SYN Flooding Attacks and Common
Mitigations", RFC 4987, DOI 10.17487/RFC 4987, August 2007,
<https://www.rfc-editor.org/info/RFC 4987>.
[RFC 5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, DOI 10.17487/RFC 5681, September 2009,
<https://www.rfc-editor.org/info/RFC 5681>.
[RFC 6181] Bagnulo, M., "Threat Analysis for TCP Extensions for
Multipath Operation with Multiple Addresses", RFC 6181,
DOI 10.17487/RFC 6181, March 2011,
<https://www.rfc-editor.org/info/RFC 6181>.
[RFC 6182] Ford, A., Raiciu, C., Handley, M., Barre, S., and J.
Iyengar, "Architectural Guidelines for Multipath TCP
Development", RFC 6182, DOI 10.17487/RFC 6182, March 2011,
<https://www.rfc-editor.org/info/RFC 6182>.
[RFC 6356] Raiciu, C., Handley, M., and D. Wischik, "Coupled
Congestion Control for Multipath Transport Protocols",
RFC 6356, DOI 10.17487/RFC 6356, October 2011,
<https://www.rfc-editor.org/info/RFC 6356>.
[RFC 6528] Gont, F. and S. Bellovin, "Defending against Sequence
Number Attacks", RFC 6528, DOI 10.17487/RFC 6528, February
2012, <https://www.rfc-editor.org/info/RFC 6528>.
[RFC 6824] Ford, A., Raiciu, C., Handley, M., and O. Bonaventure,
"TCP Extensions for Multipath Operation with Multiple
Addresses", RFC 6824, DOI 10.17487/RFC 6824, January 2013,
<https://www.rfc-editor.org/info/RFC 6824>.
[RFC 6897] Scharf, M. and A. Ford, "Multipath TCP (MPTCP) Application
Interface Considerations", RFC 6897, DOI 10.17487/RFC 6897,
March 2013, <https://www.rfc-editor.org/info/RFC 6897>.
[RFC 7323] Borman, D., Braden, B., Jacobson, V., and R.
Scheffenegger, Ed., "TCP Extensions for High Performance",
RFC 7323, DOI 10.17487/RFC 7323, September 2014,
<https://www.rfc-editor.org/info/RFC 7323>.
[RFC 7413] Cheng, Y., Chu, J., Radhakrishnan, S., and A. Jain, "TCP
Fast Open", RFC 7413, DOI 10.17487/RFC 7413, December 2014,
<https://www.rfc-editor.org/info/RFC 7413>.
[RFC 7430] 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/RFC 7430, July 2015,
<https://www.rfc-editor.org/info/RFC 7430>.
[RFC 8041] Bonaventure, O., Paasch, C., and G. Detal, "Use Cases and
Operational Experience with Multipath TCP", RFC 8041,
DOI 10.17487/RFC 8041, January 2017,
<https://www.rfc-editor.org/info/RFC 8041>.
[RFC 8126] Cotton, M., Leiba, B., and T. Narten, "Guidelines for
Writing an IANA Considerations Section in RFCs", BCP 26,
RFC 8126, DOI 10.17487/RFC 8126, June 2017,
<https://www.rfc-editor.org/info/RFC 8126>.
[TCPLO] Ramaiah, A., "TCP option space extension", Work in
Progress, Internet-Draft, draft-ananth-tcpm-tcpoptext-00,
26 March 2012, <https://tools.ietf.org/html/draft-ananth-
tcpm-tcpoptext-00>.
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
[RFC 7323]), 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
option.
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
DATA_ACK).
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
[RFC 7413], 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. [RFC 7413]
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 [RFC 7413].
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 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
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 [RFC 793] 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 [RFC 6824],
which specifies MPTCP v0; and this document, which obsoletes
[RFC 6824] and specifies MPTCP v1. Note that this specification is
not backward compatible with [RFC 6824].
* This document incorporates lessons learned from the various
implementations, deployments, and experiments gathered in the
documents "Use Cases and Operational Experience with Multipath
TCP" [RFC 8041] and the IETF Journal article "Multipath TCP
Deployments" [deployments].
* Connection initiation, through the exchange of the MP_CAPABLE
MPTCP option, is different from [RFC 6824]. 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 [RFC 6824]
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: alan.ford@gmail.com
Costin Raiciu
University Politehnica of Bucharest
Splaiul Independentei 313
Bucharest
Romania
Email: costin.raiciu@cs.pub.ro
Mark Handley
University College London
Gower Street
London
WC1E 6BT
United Kingdom
Email: m.handley@cs.ucl.ac.uk
Olivier Bonaventure
Université catholique de Louvain
Pl. Ste Barbe, 2
1348 Louvain-la-Neuve
Belgium
Email: olivier.bonaventure@uclouvain.be
Christoph Paasch
Apple, Inc.
Cupertino, CA
United States of America
Email: cpaasch@apple.com
RFC TOTAL SIZE: 202587 bytes
PUBLICATION DATE: Tuesday, March 31st, 2020
LEGAL RIGHTS: The IETF Trust (see BCP 78)
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