Internet Engineering Task Force A. Ford
Internet-Draft Roke Manor Research
Intended status: Experimental C. Raiciu
Expires: September 15, 2011 M. Handley
University College London
O. Bonaventure
Universite catholique de
Louvain
March 14, 2011
TCP Extensions for Multipath Operation with Multiple Addresses
draft-ietf-mptcp-multiaddressed-03
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. reliable
bytestream), and provides the components necessary to establish and
use multiple TCP flows across potentially disjoint paths.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
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Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on September 15, 2011.
Copyright Notice
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Copyright (c) 2011 IETF Trust and the persons identified as the
document authors. All rights reserved.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. Design Assumptions . . . . . . . . . . . . . . . . . . . . 4
1.2. Multipath TCP in the Networking Stack . . . . . . . . . . 5
1.3. Terminology . . . . . . . . . . . . . . . . . . . . . . . 6
1.4. MPTCP Concept . . . . . . . . . . . . . . . . . . . . . . 6
1.5. Requirements Language . . . . . . . . . . . . . . . . . . 7
2. Operation Overview . . . . . . . . . . . . . . . . . . . . . . 8
3. MPTCP Protocol . . . . . . . . . . . . . . . . . . . . . . . . 9
3.1. Connection Initiation . . . . . . . . . . . . . . . . . . 10
3.2. Starting a New Subflow . . . . . . . . . . . . . . . . . . 14
3.3. General MPTCP Operation . . . . . . . . . . . . . . . . . 18
3.3.1. Data Sequence Mapping . . . . . . . . . . . . . . . . 20
3.3.2. Data Acknowledgements . . . . . . . . . . . . . . . . 23
3.3.3. Closing a Connection . . . . . . . . . . . . . . . . . 24
3.3.4. Receiver Considerations . . . . . . . . . . . . . . . 25
3.3.5. Sender Considerations . . . . . . . . . . . . . . . . 26
3.3.6. Reliability and Retransmissions . . . . . . . . . . . 27
3.3.7. Congestion Control Considerations . . . . . . . . . . 28
3.3.8. Subflow Policy . . . . . . . . . . . . . . . . . . . . 28
3.4. Address Knowledge Exchange (Path Management) . . . . . . . 30
3.4.1. Address Advertisement . . . . . . . . . . . . . . . . 31
3.4.2. Remove Address . . . . . . . . . . . . . . . . . . . . 33
3.5. Fallback . . . . . . . . . . . . . . . . . . . . . . . . . 34
3.6. Error Handling . . . . . . . . . . . . . . . . . . . . . . 37
3.7. Heuristics . . . . . . . . . . . . . . . . . . . . . . . . 37
3.7.1. Port Usage . . . . . . . . . . . . . . . . . . . . . . 38
3.7.2. Delayed Subflow Start . . . . . . . . . . . . . . . . 38
3.7.3. Failure Handling . . . . . . . . . . . . . . . . . . . 39
4. Semantic Issues . . . . . . . . . . . . . . . . . . . . . . . 39
5. Security Considerations . . . . . . . . . . . . . . . . . . . 41
6. Interactions with Middleboxes . . . . . . . . . . . . . . . . 42
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 45
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8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 45
9. References . . . . . . . . . . . . . . . . . . . . . . . . . . 46
9.1. Normative References . . . . . . . . . . . . . . . . . . . 46
9.2. Informative References . . . . . . . . . . . . . . . . . . 46
Appendix A. Notes on use of TCP Options . . . . . . . . . . . . . 48
Appendix B. Control Blocks . . . . . . . . . . . . . . . . . . . 49
B.1. MPTCP Control Block . . . . . . . . . . . . . . . . . . . 50
B.1.1. Authentication and Metadata . . . . . . . . . . . . . 50
B.1.2. Sending Side . . . . . . . . . . . . . . . . . . . . . 50
B.1.3. Receiving Side . . . . . . . . . . . . . . . . . . . . 51
B.2. TCP Control Blocks . . . . . . . . . . . . . . . . . . . . 51
B.2.1. Sending Side . . . . . . . . . . . . . . . . . . . . . 51
B.2.2. Receiving Side . . . . . . . . . . . . . . . . . . . . 51
Appendix C. Changelog . . . . . . . . . . . . . . . . . . . . . . 52
C.1. Changes since draft-ietf-mptcp-multiaddressed-02 . . . . . 52
C.2. Changes since draft-ietf-mptcp-multiaddressed-01 . . . . . 52
C.3. Changes since draft-ietf-mptcp-multiaddressed-00 . . . . . 52
C.4. Changes since draft-ford-mptcp-multiaddressed-03 . . . . . 52
C.5. Changes since draft-ford-mptcp-multiaddressed-02 . . . . . 53
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 53
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1. Introduction
MPTCP is a set of extensions to regular TCP [2] to provide a
Multipath TCP [3] service, 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 signalling 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:
o Architecture [3], which explains the motivations behind Multipath
TCP, contains a discussion of high-level design decisions on which
this design is based, and an explanation of a functional
separation through which an extensible MPTCP implementation can be
developed.
o Congestion Control [4], presenting a safe congestion control
algorithm for coupling the behaviour of the multiple paths in
order to "do no harm" to other network users.
o Application Considerations [5], discussing 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.
1.1. Design Assumptions
In order to limit the potentially huge design space, the authors
imposed two key constraints on the multipath TCP design presented in
this document:
o It must be backwards-compatible with current, regular TCP, to
increase its chances of deployment
o 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
utilisation and resilience to a subset of node failures. The
congestion control algorithms as discussed in [4] ensure this does
not act detrimentally.
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There are three aspects to the backwards-compatibility listed above
(discussed in more detail in [3]):
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 the
segments it sends on the wire arrive unmodified at the
destination: they may be split or coalesced; options may be
removed or duplicated.
Application Constraints: The protocol must be usable with no change
to existing applications that use the standard 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.
Fall-back: 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.
Further discussion of the design constraints and associated design
decisions are given in the MPTCP Architecture document [3].
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
[5].
+-------------------------------+
| Application |
+---------------+ +-------------------------------+
| Application | | MPTCP |
+---------------+ + - - - - - - - + - - - - - - - +
| TCP | | Subflow (TCP) | Subflow (TCP) |
+---------------+ +-------------------------------+
| IP | | IP | IP |
+---------------+ +-------------------------------+
Figure 1: Comparison of Standard TCP and MPTCP Protocol Stacks
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1.3. Terminology
Path: A sequence of links between a sender and a receiver, defined
in this context by a source and destination address pair.
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: A end host operating an MPTCP implementation, and either
initiating or accepting an MPTCP connection.
1.4. MPTCP Concept
This section provides a high-level summary of normal operation of
MPTCP, and is illustrated by the scenario shown in Figure 2. A
detailed description of operation is given in Section 3.
o To a non-MPTCP-aware application, MPTCP will behave the same as
normal TCP. Extended APIs could provide additional control to
MPTCP-aware applications [5]. An application begins by opening a
TCP socket in the normal way. MPTCP signaling and operation is
handled by the MPTCP implementation.
o An MPTCP connection begins similarly to a regular TCP connection.
This is illustrated in Figure 2 where a TCP connection is
established between addresses A1 and B1 on Hosts A and B
respectively.
o If extra paths are available, additional TCP sessions (termed
"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.
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o 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.
o 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 signalling its available addresses
to the other host.
o MPTCP adds connection-level sequence numbers to allow the
reassembly of the in-order data stream from multiple subflows
which may deliver packets out-of-order due to differing network
delays.
o Subflows are terminated as regular TCP connections, with a four
way FIN handshake. The MPTCP connection is terminated by a
connection-level FIN.
Host A Host B
------------------------ ------------------------
Address A1 Address A2 Address B1 Address B2
---------- ---------- ---------- ----------
| | | |
| (initial connection setup) | |
|----------------------------------->| |
|<-----------------------------------| |
| | | |
| (additional subflow setup) |
| |--------------------->| |
| |<---------------------| |
| | | |
| | | |
Figure 2: Example MPTCP Usage Scenario
1.5. Requirements Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [1].
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2. Operation Overview
This section presents a single description of standard MPTCP
operation, with reference to the protocol operation. The detailed
protocol specification follows in Section 3.
To understand the operation of Multipath TCP, let us consider a very
simple case where a client having two addresses, A1 and A2
establishes an MPTCP connection with a dual homed server having
addresses B1 and B2, as illustrated in Figure 2 in the previous
section. MPTCP offers the same bidirectional bytestream service as
regular TCP.
To open an MPTCP connection, the client sends a SYN segment from one
of its addresses (say A1) to one of the server's addresses (say B1).
This SYN segment contains the MP_CAPABLE option that indicates that
the client supports MPTCP and contains the client's key for this
MPTCP connection. The server replies with a SYN segment that also
contains the MP_CAPABLE option to confirm that it supports MPTCP.
The MP_CAPABLE option returned by the server includes the server's
key. The client are server keys are used for different purposes by
MPTCP. First, each host derives a 32 bits token that uniquely
identifies the MPTCP connection on this host. Second, the keys are
used to authenticate the utilisation of other addresses. Additional
details about the utilisation of the MP_CAPABLE option may be found
in Section 3.1.
To enable the client and the server to use their multiple addresses
to support the same MPTCP connection, MPTCP allows the client and the
server to open additional subflows. These subflows are TCP
connections that are linked to the MPTCP connection and can be used
to send and receive data. The client can open an additional subflow
by sending a SYN segment from another address (e.g. A2) with the
MP_JOIN option to the server. The MP_JOIN option contains the
server's token that uniquely identifies the MPTCP connection to which
the subflow must be associated and a random number. To accept the
subflow, the server replies by sending a SYN+ACK segment with the
MP_JOIN option that contains a random number chosen by the server and
a HMAC computed over the client and server's random numbers with the
client and server keys. This HMAC authenticates the server to the
client. Upon reception of this SYN+ACK segment, the client replies
with an ACK segment that contains an MP_JOIN option that includes
another HMAC that authenticates the client to the server. Additional
details about the utilisation of the MP_JOIN option may be found in
Section 3.2.
The server may also establish one or more subflows with the client by
sending SYN segments with the MP_JOIN option that has been briefly
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described above. Furthermore, a host my also inform the other host
of the IP addresses that it owns. MPTCP uses two options for this
purpose. The ADD_ADDR option allows a host to indicates that it owns
another address. For example, in the above scenario, the server
could use the ADD_ADDR option to indicate that it also owns address
B2. If a host becomes unable to use a previously advertised address,
it uses the REMOVE_ADDR option to indicate the address that it lost
to its peer. Additional details about the utilisation of the
ADD_ADDR and REMOVE_ADDR options may be found in Section 3.4.
The data produced by the client and the server can be sent over any
of the subflows that compose an MPTCP connection, and if a subflow
fails, data may need to be retransmitted over another subflow. For
this, MPTCP relies on two principles. First, each subflow is
equivalent to a normal TCP connection with its own 32-bits sequence
numbering space. This enables MPTCP to traverse complex middle-boxes
like transparent proxies or traffic normalizers. Second, MPTCP
maintains a 64-bits data sequence numbering space. The DSS MPTCP
option is used to send the data sequence numbers and data sequence
acknowledgements. When a host sends a TCP segment over one subflow,
it indicates inside the segment, by using the DSS option, the mapping
between the 64-bits data sequence number and the 32-bits sequence
number used by the subflow. Thanks to this mapping, the receiving
host can reorder the data received, possibly out-of-sequence over the
different subflows. In MPTCP, a received segment is acknowledged at
two different levels. First, the TCP cumulative or selective
acknowledgements are used to acknowledge the reception of the data on
each subflow. Second, the acknowledgements field in the DSS option
is returned by the receiving host to provide cumulative
acknowledgements at the data sequence level. When a segment is lost,
the receiver detects the gap in the received 32-bits sequence number
and traditional TCP retransmission mechanisms are triggered to
recover from the loss. When a subflow fails, MPTCP detects the
failure and retransmits the unacknowledged data over another subflow
that is still active. The DSS option also includes an optional
checksum that covers data at the MPTCP connection level to enable a
receiver to detect whether an middlebox has inserted, deleted or
modified data on-the-fly. The transmission of data by MPTCP is
discussed in details in Section 3.3.
3. MPTCP Protocol
This section describes the operation of the MPTCP protocol, and is
subdivided into sections for each key part of the protocol operation.
All MPTCP operations are signalled using optional TCP header fields.
A single TCP option number will be assigned by IANA (see Section 8),
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and then individual messages will be determined by a "sub-type", the
values of which will also be stored in an IANA registry (and are also
listed in Section 8). This sub-type is a four-bit field - the first
four 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 must be
included on packets with the SYN flag set. Additionally, there is
one MPTCP option for signalling metadata to ensure 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 signalling additional addresses.
Whilst 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 [6]) on a single
packet. Therefore, an implementation may choose to send duplicate
ACKs containing the additional signalling information. This changes
the semantics of a duplicate ACK, these are usually only sent as a
signal of a lost segment [7] in regular TCP. Therefore, an MPTCP
implementation receiving a duplicate ACK which 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 signalling purposes, so as to ensure no middleboxes
misinterpret this as a sign of congestion.
Furthermore, standard TCP validity checks (such as ensuring the
Sequence Number and Acknowledgement Number are within window) MUST be
undertaken before processing any MPTCP signals, as described in [8].
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)
TCP option (Figure 4). This option declares its sender is capable of
performing multipath TCP and wishes to do so on this particular
connection.
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This option contains a 64-bit key that is used to authenticate the
addition of future subflows. This is the only time the key will be
sent in clear on the wire; all future subflows will identify the
connection using a 32-bit "token". This token is a cryptographic
hash of this key. This will be a truncated (most significant 32
bits) SHA-1 hash [9]. A different, 64-bit truncation (the least
significant 64 bits) of the hash of the key will be used as the
Initial Data Sequence Number.
This key is generated by its sender and has local meaning only, and
its method of generation is implementation-specific. The key MUST be
hard to guess, and it MUST be unique for the sending host at any one
time. Recommendations for generating random keys are given in [10].
Connections will be indexed at each host by the token (the truncated
SHA-1 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 very small risk that two different keys will hash to the
same token. An implementation SHOULD check its list of connection
tokens to ensure there is not a collision before sending its key in
the SYN/ACK. 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, so in the worst case if there was a token
collision, it just means that the second connection cannot support
multiple subflows, but will otherwise provide a regular TCP service.
The MP_CAPABLE option is carried on the SYN, SYN/ACK, and ACK packets
that start the first subflow of an MPTCP connection. The data
carried by each packet is as follows, where A = initiator and B =
listener.
o SYN (A->B): A's Key.
o SYN/ACK (B->A): B's Key.
o ACK (A->B): Both A's Key and B's Key.
The contents of the option is determined by the SYN and ACK flags of
the packet, verified by the option's length field. For the diagram
shown in Figure 4, "sender" and "receiver" refer to the sender or
receiver of the TCP packet (which can be either host). If the SYN
flag is set, a single key is included; if only an ACK flag is set,
both keys are present.
The keys are echoed in the ACK in order to allow the listener (host
B) to act statelessly until the TCP connection reaches the
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ESTABLISHED state. If the listener acts in this way, however, it
MUST generate its key in a verifiable fashion, allowing it to verify
that it generated the key when it is echoed in the ACK.
Furthermore, in order to ensure reliable delivery of the ACK
containing the MP_CAPABLE option, a server MUST respond with an ACK
segment on receipt of this, which may contain data, or will be a pure
ACK if it does not have any data to send immediately. If the
initiator does not receive this ACK within the RTO, it MUST re-send
the ACK containing MP_CAPABLE. In effect, an MPTCP connection is in
a "PRE_ESTABLISHED" state while awaiting this ACK, and only upon
receipt of the ACK will it move to the ESTABLISHED state.
The first four bits of the first octet in the MP_CAPABLE option
(Figure 4) define the MPTCP option subtype (see Section 8; for
MP_CAPABLE, this is 0), and the remaining four bits of this octet
specifies the MPTCP version in use (for this specification, this is
0).
The second octet is reserved for flags. The leftmost bit - labeled C
- indicates "Checksum required", and SHOULD be set to 1 unless
specifically overridden (for example, if the system administrator has
decided that checksums are not required - see Section 3.3 for more
discussion). The remaining bits are used for crypto algorithm
negotiation. Currently only the rightmost bit - labeled S - is
assigned, and indicates the use of HMAC-SHA1 (as defined in
Section 3.2). An implementation that only supports this method MUST
set this bit to 1 and all other currently reserved bits to zero. If
none of these flags are set, the MP_CAPABLE option MUST be treated as
invalid and ignored (i.e. it must be treated as a regular TCP
handshake).
These bits negotiate capabilities in similar ways. For the 'C' bit,
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 C=0. The decision whether to use checksums
will be stored by an implementation in a per-connection binary state
variable.
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
S will be always set to 1). The responder responds with only one bit
set - this is the chosen algorithm. The rationale for this behaviour
is that the responder will typically be a server with potentially
many thousands of connections, so may wish to choose an algorithm
with minimal computational complexity, depending on load. If a
responder does not support (or does not want to support) any of the
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initiator's proposals, it can respond without an MP_CAPABLE option,
thus forcing a fall-back to regular TCP.
The MP_CAPABLE option is only used in the first subflow of a
connection, in order to identify the connection; all following
subflows will use the "Join" option (see Section 3.2) to join the
existing 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|C| (reservd) |S|
+---------------+---------------+-------+-------+-+-----------+-+
| Sender's Key |
| (64 bits) |
| |
+---------------------------------------------------------------+
| Receiver's Key (64 bits) |
| (if Length==20) |
| |
+---------------------------------------------------------------+
Figure 4: Multipath Capable (MP_CAPABLE) option
If a SYN contains an MP_CAPABLE option but the SYN/ACK does not, it
is assumed that the passive opener is not multipath capable and thus
the MPTCP session MUST operate as regular, single-path TCP. If a SYN
does not contain a 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
regular, single-path TCP. This is to maintain compatibility with
middleboxes on the path that drop some or all TCP options.
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 re-ordered 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 3.5.
The initial Data Sequence Number (IDSN) is generated as a hash from
the Key, in the same way as the token, i.e. IDSN-A = Hash(Key-A) and
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IDSN-B = Hash(Key-B). The Hash mechanism here provides the least
significant 64 bits of the SHA-1 hash of the key. The SYN with
MP_CAPABLE occupies the first octet of Data Sequence Space.
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 signalling 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 permitted 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 3.7 for heuristics).
A new subflow is started as a normal TCP SYN/ACK exchange. The Join
Connection (MP_JOIN) TCP 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 behaviour of MP_JOIN using the HMAC-SHA1
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.
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). The tokens presented in this
option are generated by the SHA-1 [9] algorithm, 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).
The MP_JOIN SYN not only sends the token (which is static for a
connection) but also Random Numbers (nonces) that are used to prevent
replay attacks on the authentication method.
The MP_JOIN option includes an "Address ID". This is an identifier
that only has significance within a single connection, where it
identifies the source address of this packet, even if the address
itself has been changed in transit by a middlebox. This allows
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address removal without needing to know what the source address at
the receiver is, thus this allows address removal through NATs. The
sender can signal this to the receiver via the REMOVE_ADDR option
(Section 3.4.2). It also allows correlation between new subflow
setup attempts and address signalling (Section 3.4.1), to prevent
setting up duplicate subflows on the same path.
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 Address IDs associated with all
established subflows.
The MP_JOIN option on SYNs also includes 4 bits of flags, 3 of which
are currently reserved and MUST be set to zero by the sender. The
final bit, labelled 'B', indicates whether the initiator wishes this
subflow to be used purely as a backup path (B=1) in the event of
failure of other paths, or whether it wants it to be used as part of
the connection immediately. Subflow policy is discussed in more
detail in Section 3.3.8.
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| |B| Address ID |
+---------------+---------------+-------+-----+-+---------------+
| Receiver's Token (32 bits) |
+---------------------------------------------------------------+
| Sender's Random Number (32 bits) |
+---------------------------------------------------------------+
Figure 5: Join Connection (MP_JOIN) option (for initial SYN)
When receiving a SYN with a 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) MAC. 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
an RST, analogous to an unknown port in TCP. Although cryptographic
calculations are required in the SYN/ACK, it is felt that the 32-bit
token gives sufficient protection against blind state exhaustion
attacks and therefore there is no need to provide mechanisms to allow
a responder to operate statelessly at the MP_JOIN stage.
An MAC 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). This is to allow both hosts to have exchanged random
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data to be used as the message before generating the MAC. In both
cases, the MAC algorithm is HMAC as defined in [11], using the SHA-1
hash algorithm [9] (thus generating a 160-bit / 20 octet HMAC). Due
to option space limitations, the MAC included in the SYN/ACK is
truncated to the leftmost 64 bits, but this is acceptable since while
in an attacker-initiated attack, the attacker can retry many times;
if the attacker is the responder, he only has one chance to get the
MAC correct.
The initiator's authentication information is sent in its first ACK,
and is shown in Figure 7. The same reliability algorithm for this
packet as for the MP_CAPABLE ACK is applied: receipt of this packet
MUST trigger an 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.
The reserved bits in this option MUST be set to zero by the sender.
The key for the MAC 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 in each case is
the concatenations of Random Number 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.
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| |B| Address ID |
+---------------+---------------+-------+-----+-+---------------+
| |
| Sender's Truncated MAC (64 bits) |
| |
+---------------------------------------------------------------+
| Sender's Random Number (32 bits) |
+---------------------------------------------------------------+
Figure 6: Join Connection (MP_JOIN) option (for responding SYN/ACK)
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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 MAC (160 bits SHA-1) |
| |
| |
+---------------------------------------------------------------+
Figure 7: Join Connection (MP_JOIN) option (for third ACK)
These various TCP 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(Key-A) |
|--------------------------------------------->|
|<---------------------------------------------|
| SYN/ACK + MP_CAPABLE(Key-B) |
| | |
| ACK + MP_CAPABLE(Key-A, Key-B) |
|--------------------------------------------->|
| | |
| | SYN + MP_JOIN(Token-B, R-A) |
| |------------------------------->|
| |<-------------------------------|
| | SYN/ACK + MP_JOIN(MAC-B, R-B) |
| | |
| | ACK + MP_JOIN(MAC-A) |
| |------------------------------->|
| | |
MAC-A = MAC(Key=(Key-A+Key-B), Msg=(R-A+R-B))
MAC-B = MAC(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.
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If the token is accepted at Host B, but the MAC returned to Host A
does not match the one expected, Host A MUST close the subflow with a
TCP RST.
If Host B does not receive the expected MAC, or the MP_JOIN option is
missing from the ACK, it MUST close the subflow with a TCP RST.
If the MACs are verified as correct, then both hosts have
authenticated each other as being the same peers as 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 RST.
This covers all cases of the loss of an MP_JOIN. In more detail, if
MP_JOIN is stripped from the SYN on the path from A to B, and Host B
does not have a passive opener on the relevant port, it will respond
with an 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
(either since it was stripped on the return path, or it was stripped
on the outgoing path but the passive opener on Host B responded as if
it were 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 3.7 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. A consequence is that it is
possible to allow any port pairs to be used for a connection.
Deumultiplexing 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 setup, demultiplexing
packets is done using the five-tuple, as in traditional TCP. The
five-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. General MPTCP Operation
This section discusses 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
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sufficient control information to allow it to be reassembled and
delivered reliably and in-order to the recipient application. The
following subsections define this behaviour in detail.
During normal MPTCP operation, the Data Sequence Signal (DSS) TCP
option (shown in Figure 9) is used to signal the data required to
enable multipath transport. This data comprises: the Data Sequence
Mapping (DSM), which defines how the sequence space on the subflow
maps to the connection level; and the Data ACK, for acknowledging
receipt of data at the connection level. These functions are
described in more detail in the following two subsections.
Either or both of the Data Sequence Mapping or the Data ACK can be
signalled in the DSS option, dependent on the flags set.
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:
o A = Data ACK present
o a = Data ACK is 8 octets (if not set, Data ACK is 4 octets)
o M = Data Sequence Number, Subflow Sequence Number, Data-level
Length, and Checksum present
o 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
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connection-level equivalent to the FIN flag in single-path TCP. The
purpose of the DATA FIN, along with 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 zero 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.
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) have subflow-only relevance. It is expected (but not
mandated) that SACK [6] is used at the subflow level to improve
efficiency.
The Data Sequence Mapping specifies a full mapping from subflow
sequence space to data sequence space, for the specified length
(number of bytes of data) starting at the specified Subflow and Data
Sequence Numbers. The purpose of the explicit mapping is 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). It also allows a single mapping to cover many
packets, which may be useful in bulk transfer situations.
A mapping is unique, in that the subflow sequence number is bound to
the data sequence number after the mapping has been processed. It is
not possible to change this mapping afterwards; however, the same
data sequence number can be mapped to different subflows for
retransmission purposes (see Section 3.3.6). It would also permit
the same data to be sent simultaneously on multiple subflows for
resilience purposes, although the detailed specification of such
operation is outside the scope of this document.
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 relative subflow sequence number 0). This is to
allow middleboxes to change the Initial Sequence Number of a subflow,
such as firewalls that undertake ISN randomization.
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The data sequence mapping also contains a checksum of the data that
this mapping covers. This is 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 3.5, since MPTCP can no longer
reliably know the subflow sequence space at the receiver to build
data sequence mappings.
The checksum algorithm used is the standard TCP checksum [2],
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, and 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 that at 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
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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 an RST, and an 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 it is permissible to include just the
lower 32 bits of the data sequence number in the Data Sequence
Mapping and/or Data ACK as an optimization. An implementation MUST
send the full 64 bit Data Sequence Number if it is transmitting at a
sufficiently high rate that it could wrap within the MSL [12]. 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 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 is done by setting the initial data sequence
number (IDSN) of each host to the least significant 64 bits of the
SHA-1 hash of the host's key, as described in Section 3.1.
A Data Sequence Mapping 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,
another is when segments of data are scheduled in larger than packet-
sized chunks. An "infinite" mapping can be used to fallback to
regular TCP by mapping the subflow-level data to the connection-level
data for the remainder of the connection (see Section 3.5). This is
achieved by setting the data-level length field to the reserved value
of 0. The checksum, in such a case, will also be set to zero.
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3.3.2. Data Acknowledgements
To provide full end-to-end resilience, MPTCP provides a connection-
level acknowledgement, to act as a cumulative ACK for the connection
as a whole. This is the "Data ACK" field of the DSS option
(Figure 9). The Data ACK is analogous to the behaviour of the
standard TCP cumulative ACK in TCP SACK - indicating how much data
has been successfully received (with no holes). 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, it is valid for the implementation to choose whether to send
a 32 bit or 64 bit Data ACK.
The rationale for the inclusion of the Data ACK includes the
existence of certain middleboxes that pro-actively ACK packets, and
thus might cause deadlock conditions if data were acked at the
subflow level but then fails to reach the receiver. This sort of bad
interaction might be especially prevalent when the receiver is
mobile. The Data ACK ensures the data has been delivered to the
receiver. Furthermore, separating the connection-level
acknowledgements from the subflow-level allows processing to be done
separately, and a receiver has the freedom to drop segments after
acknowledgement at the subflow level, for example due to memory
constraints when many segments arrive out-of-order.
Another reason for including the Data ACK is that it indicates 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 connection-level if it
does not also carry a DSS option with a Data ACK field.
An MPTCP sender MUST only free data from the send buffer when it has
been acknowledged by both a Data ACK received on any subflow and at
the subflow level by any subflows the data was sent on. 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 restransmited. Note, however, that if some data needs to be
retransmitted multiple times over a subflow, there is a risk of
blocking the sending window. In this case, the MPTCP sender can
decide to cancel the subflow that is behaving badly by sending a RST.
The Data ACK MAY be included in all segments, however optimisations
SHOULD be considered in more advanced implementations, where the Data
ACK is present in segments only when the Data ACK value advances, and
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this behaviour MUST be treated as valid. This behaviour ensures the
sender buffer is freed, while reducing overhead when the data
transfer is unidirectional.
3.3.3. Closing a Connection
In regular TCP a FIN announces 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, and for regular TCP this would result in a
FIN on the connection. For MPTCP, an equivalent mechanism is needed,
and 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 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 signalled by setting the 'F' flag in the Data
Sequence Signal option (Figure 9) to 1. A DATA FIN occupies one
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 DSN 80, and length 11,
with DATA FIN set, would map 10 octets from the subflow into data
sequnce space 80-89, the DATA FIN is DSN 90, and 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 Data Sequence Mapping MUST have Subflow Sequence
Number of 0, a 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 semantics and behaviour 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 permissable to combine these signals on one subflow if there is
no data oustanding on other subflows. Otherwise, it may be necessary
to retransmit data on different subflows. Essentially, a host MUST
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NOT FIN all subflows unless it is safe to do so, i.e. until all data
has been DATA ACKed, or that the segment with the 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, as
a courtesy to allow middleboxes to clean up state even if the subflow
has failed. It is also encouraged to reduce the timeouts (Maximum
Segment Life) on subflows at end hosts. 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 an RST.
A connection is considered closed once both hosts' DATA FINs have
been acknowledged by DATA ACKs.
Note that a host may also send a FIN on an individual subflow to shut
it down, but this impact is limited to the subflow in question. 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 and connection levels.
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 connection level.
When deciding to accept packets at subflow level, normal TCP uses the
sequence number in the packet and checks it against the allowed
receive window. With multipath, such a check is done using only the
connection level window. A sanity check SHOULD be performed at
subflow level to ensure that the subflow and mapped sequence numbers
meet the following test: SSN - SUBFLOW_ACK <= DSN - DATA_ACK.
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In regular TCP, once a segment is deemed in-window, it is either put
in the in-order receive queue or in the out-of-order queue. In
multipath TCP, the same happens but at connection-level: a segment is
placed in the connection level in-order or out-of-order queue if it
is in-window at both connection and subflow level. The stack still
has to remember, for each subflow, which segments were received
succesfully so that it can ACK them at 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 receiver
buffer is appropriate. The lower bound for full network utilization
is the maximum bandwidth-delay product of any 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 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. It is for future study to determine the relationship between
retransmission strategies and receive buffer sizing.
3.3.5. Sender Considerations
The sender remembers receiver 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. This is important to allow using 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 also remembers the receive windows advertised by each
subflow. The allowed window for subflow i is (ack_i, ack_i +
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rcv_wnd_i), where ack_i is the subflow-level cumulative ack of
subflow i. This ensures 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 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 be, at the minimum, 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 re-send 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 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 the best behaviour. Optimisations
could be negotiated in future versions of this protocol.
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 retransmissions policies where every packet lost at
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
retransmits 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 subflow level but
its corresponding data is not ACKed at data level. This timer will
guard against failures in re-transmission by middleboxes that pro-
active ACK data.
The sender MUST keep data in its send buffer as long as the data has
not been acknowledged at both connection level and on all subflows it
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has been sent on. 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 Maximum Segment Life), or on the receipt of an ICMP
error, and only then delete the outstanding data segments.
Multiple retransmissions are triggers that will indicate that a
subflow performs badly and could lead to a host resetting the subflow
with an RST. However, additional research is required to understand
the heuristics of how and when to reset underperforming subflows.
For example, subflows that perform highly asymmetrically may be mis-
diagnosed as underperforming.
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 [4]; 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 the MPTCP protocol
aims to provide the congestion control implementations sufficient
information to take 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 maximise throughput,
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all available paths will be used simultaneously for data transfer,
using coupled congestion control as described in [4]. It is
expected, however, that other use cases will appear.
For instance, a 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 [5].
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.
Whilst fine-grained control may be the most powerful solution, that
would require some mechanism such as overloading the ECN signal [13],
which is undesirable, and it is felt that there would not be
sufficient benefit to justify an entirely new signal. Therefore 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 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. 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
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
+---------------+---------------+-------+-----+-+
| Kind | Length |Subtype| |B|
+---------------+---------------+-------+-----+-+
Figure 11: MP_PRIO option
It should be noted that the backup flag is a request from the
receiver to the sender only, and the sender SHOULD adhere to these
requests. The receiver, however, may continue using the subflow to
send data even if it has signalled B=1 to the other host.
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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 detail of
the architectural thinking behind this design, see the separate
architecture document [3].
This design makes use of two methods of sharing such information,
used simultaneously. The first is the direct setup of new subflows,
already 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 explicit is more complex
but is more robust. Together, the mechanisms allow addresses to
change in flight (and thus support operation through NATs, since the
source address need not be known), and also allow the signalling 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:
o 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
multi-addressed, it can start an additional subflow from its
address A2 to B1, by sending a SYN with a 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.
o 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. In our example, A will send 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.
o If subflow A2-B1 is succesfully setup, host B can use the Address
ID in the Join option to correlate this 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
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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 option.
3.4.1. Address Advertisement
The Add Address (ADD_ADDR) TCP Option announces additional addresses
(and optionally, ports) on which a host can be reached (Figure 12).
Multiple instances of this TCP option can be added in a single
message if there is sufficient TCP option space, otherwise multiple
TCP messages containing this option will be sent. 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.
Every address has an ID which can be used for uniquely identifying
the address within a connection, for address removal. This 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 ID MUST
uniquely identify the address to the sender (within the scope of the
connection), but the mechanism for allocating such IDs is
implementation-specific.
All address IDs learnt 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 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 for
avoiding the required 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 MAY choose to refresh advertisements periodically.
This option is shown in Figure 12. The illustration is sized for
IPv4 addresses (IPVer = 4). For IPv6, the IPVer field will read 6,
and the length of the address will be 16 octets (instead of 4).
The presence of the final two octets, specifying the TCP port number
to use, are optional and 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 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
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the explicit specification of a different port is required. If no
port is specified, MPTCP SHOULD attempt to connect to the specified
address on same port as is already in use by the signalling subflow,
and this is discussed in more detail in Section 3.7.
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| IPVer | Address ID |
+---------------+---------------+-------+-------+---------------+
| Address (IPv4 - 4 octets / IPv6 - 16 octets) |
+-------------------------------+---------------+---------------+
| Port (2 octets, optional) |
+-------------------------------+
Figure 12: Add Address (ADD_ADDR) option (shown for IPv4)
Due to the proliferation of NATs, it is reasonably likely that one
host may attempt to advertise private addresses [14]. We do not wish
to blanket prohibit this, since there may be cases where both hosts
have additional interfaces on the same private network. We must
ensure, however, that such advertisements do not cause harm. The
standard mechanism to create a new subflow (Section 3.2) contains a
32-bit token that uniquely identifies the connection to the receiving
host. If the token is unknown, the host will return with a RST. In
the unlikely event that the token is known, subflow setup will
continue, but the MAC exchange must occur for authentication. This
will fail, and will provide sufficient protection against two
unconnected hosts accidentally setting up a new subflow upon the
signal of a private address.
Ideally, we'd like to ensure the ADD_ADDR and REMOVE_ADDR options are
sent reliably, and in order, to the other end. This is to ensure
that we do not unnecessarily cause an outage in the connection when
remove/add addresses are processed in reverse order, and also to
ensure that all possible paths are used. We note, however, that
losing reliability and ordering it will not break the multipath
connections; they will just reduce the opportunity to open multipath
paths and to survive different patterns of path failures.
Therefore, implementing reliability signals for these TCP options is
not necessary. In order to minimise the impact of the loss of these
options, however, it is RECOMMENDED that a sender should send these
options on all available subflows. If these options need to be
received in-order, an implementation SHOULD only send one ADD_ADDR/
REMOVE_ADDR option per RTT, to minimise the risk of misordering.
When receiving an ADD_ADDR message with an Address ID already in use
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for a live subflow within the connection, the receiver SHOULD
silently ignore the ADD_ADDR. If the Address ID is not in use on a
live subflow, but is stored by the receiver, a new ADD_ADDR SHOULD
take precedence and replace the stored address.
A host that receives an ADD_ADDR but finds a connection setup to that
address is unsuccessful SHOULD NOT perform further connection
attempts to this address for this connection. A sender that wants to
trigger a new incoming connection attempt on a previously advertised
address can therefore refresh ADD_ADDR information by sending the
option again.
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 MPTCP options as
indications of congestion [7], and an MPTCP implementation SHOULD NOT
send more than two duplicate ACKs in a row for signalling 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),
the affected host SHOULD announce this so that the peer can remove
subflows related to this address.
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.
For security purposes, if a host receives a REMOVE_ADDR option, it
must ensure the affected path(s) are no longer in use before it
instigates closure. The receipt of REMOVE_ADDR SHOULD first trigger
the sending of a TCP Keepalive [15] on the path, and if a response is
received the path is not removed. Typical TCP validity tests on the
subflow (e.g. ensuring sequence and ack numbers are correct) MUST
also be undertaken.
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 cleaning up
middlebox state, before cleaning up any local state.
Address removal is undertaken by ID, so as to permit the use of NATs
and other middleboxes that rewrite source addresses. If there is no
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address at the requested ID, the receiver will silently ignore the
request.
A subflow that is still functioning MUST be closed with a FIN
exchange as in regular TCP - for more information, see Section 3.3.3.
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| | Address ID | ...
+---------------+---------------+-------+-------+---------------+
Figure 13: Remove Address (REMOVE_ADDR) option
3.5. Fallback
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 TCP options can reach each host. The handshake as
described in Section 3.1 will fall back to regular TCP if either of
the SYN messages do not have the MPTCP options: this is the same, and
desired, behaviour 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, the TCP options will not get through on the SYNs and the
subflow will be closed.
There is, however, another corner case which should be addressed.
That is one of MPTCP options getting through on the SYN, but not on
regular packets. This can be resolved if the subflow is the first
subflow, and thus all data in flight is contiguous, 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
acknowledgement, 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 Data ACK in a DSS option, the sender determines 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 an RST. In the case of the first subflow (i.e. that
started with MP_CAPABLE), it MUST drop out of a MPTCP mode back to
regular TCP. The sender will send one final Data Sequence Mapping,
with the length value of 0 indicating an infinite mapping (in case
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the path drops options in one direction only), and then revert to
sending data on the single subflow without any MPTCP options.
Note that this rule essentially prohibits the sending of data on the
third packet of a MP_CAPABLE or MP_JOIN handshake, since both that
option and a DSS cannot fit in TCP option space. If the initiator is
to send first, another segment must be sent that contains the data
and DSS. Note also that an additional subflow cannot be used until
the initial path has been verified as MPTCP-capable.
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. If lost options on data packets
occur on any other subflow apart from the the initial subflow, it
should be treated as a standard path failure. The data would not be
DATA_ACKed (since there is no mapping for the data), and the subflow
can be closed with an RST. (Note that these rules do not apply if an
infinite mapping is included from the start - in which case, each end
will send DSS options declaring the infinite mapping.)
The case described above is a specialised case of fallback. More
generally, fallback to regular TCP can become necessary at any point
during a connection if a non-MPTCP-aware middlebox changes the data
stream.
As described in Section 3.3, each portion of data for which there is
a mapping is protected by a checksum. 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. This will also detect if 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
cannot signal any further mappings. Furthermore, in addition to the
possibility of payload modifications that are valid at the
application layer, there is the possibility that false-positives
could be hit across MPTCP segment boundaries, corrupting the data.
Therefore, all data from the start of the segment that failed the
checksum onwards is not trustworthy.
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 the adjustment has 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
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must be immediately closed with an RST, featuring a MP_FAIL option
(Figure 14), which defines the Data Sequence Number at the start of
the segment (defined by the Data Sequence Mapping) which had the
checksum failure.
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) :
+--------------------------------------------------------------+
: Data Sequence Number (continued) |
+--------------------------------------------------------------+
Figure 14: Fallback (MP_FAIL) option
Failed data will not be DATA_ACKed and so will be re-transmitted on
other subflows (Section 3.3.6).
A special case is when there is a single subflow and it fails with a
checksum error. Here, MPTCP should be able to recover and continue
sending data. There are two possible mechanisms to support this.
The first and simplest is to establish a new subflow as part of the
same MPTCP connection, and then close the original one with a RST.
Since it is known that the path may be compromised, it is not
desirable to use MPTCP's segmentation on this path any longer. The
new subflow will begin and will signal an infinite mapping (indicated
by length=0 in the Data Sequence Mapping option, Section 3.3) from
the data sequence number of the segment that failed the checksum.
This connection will then continue to appear as a regular TCP
session, and a middlebox may change the payload without causing
unintentional harm.
An optimisation is possible, however. If it is known that all
unacknowledged data in flight is contiguous, an infinite mapping
could be applied to the subflow without the need to close it first,
and essentially turn off all further MPTCP signalling. 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.
The sender will receive this, and if all unacknowledged data in
flight is contiguous, will signal an infinite mapping (if the data is
not contiguous, the sender MUST send an RST). 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 last segment that
was known to be delivered intact. From that point onwards data can
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be altered by a middlebox without affecting MPTCP, as the data stream
is equivalent to a regular, legacy TCP session.
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 receive SHOULD stop generating Data ACKs after it receives
an infinite mapping.
When a connection is in fallback mode, only one subflow can send data
at a time. Otherwise, the receiver would not know how to reorder the
data. However, subflows can be opened and closed as necessary, as
long as a single one is active at any point.
It should be emphasised that we are not attempting to prevent the use
of middleboxes that want to adjust the payload. An MPTCP-aware
middlebox to provide such functionality could be designed that would
re-write checksums if needed, and additionally would be able to parse
the data sequence mappings, and thus not hit false positives though
not knowing where data boundaries lie.
3.6. Error Handling
In addition to the fallback mechanism as 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 an RST - are covered in Section 4. Where possible, we do not want
to deviate from regular TCP behaviour.
The following list covers possible errors and the appropriate MPTCP
behaviour:
o Unknown token in MP_JOIN (or MAC failure in MP_JOIN ACK, or
missing MP_JOIN in SYN/ACK response): send RST (analogous to TCP's
behaviour on an unknown port)
o DSN out of Window (during normal operation): drop the data, do not
send Data ACKs.
o Remove request for unknown address ID: silently ignore
3.7. Heuristics
There are a number of heuristics that are needed for performance or
deployment but which are not required for protocol correctness. In
this section we detail such heuristics. Note that discussion of
buffering and certain sender and receiver window behaviours are
presented in Section 3.3.4 and Section 3.3.5, as well as
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retransmission in Section 3.3.6.
3.7.1. Port Usage
Under typical operation an MPTCP implementation SHOULD use the same
ports as already in use. In other words, the destination port of a
SYN containing a 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 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 the passive opener wishes to
signal to the other host that a specific port should be used, and
this facility is provided in the Add Address 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).
3.7.2. Delayed Subflow Start
Many TCP connections are short-lived and consist only of a few
segments, and so the overheads of using MPTCP outweigh any benefits.
A heuristic is required, therefore, to decide when to start using
additional subflows in an MPTCP connection. We expect that
experience gathered from deployments will provide further guidance on
this, and will be affected by particular application characteristics
(which are likely to change over time). However, a suggested
general-purpose heuristic that an implementation MAY choose to employ
is as follows. Results from experimental deployments are needed in
order to verify the correctness of this proposal.
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 which will never
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fill its buffers, and thus never use MPTCP. 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.
This section has shown some of the considerations than an implementer
should give when developing MPTCP heuristics, but is not intended to
be prescriptive.
3.7.3. Failure Handling
Requirements for MPTCP's handling of unexpected signals have been
given in Section 3.6. There are other failure cases, however, where
a hosts can choose appropriate behaviour.
For example, Section 3.1 suggests that a host should fall back to
trying regular TCP SYNs after several failures of MPTCP SYNs. 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
failing.
Another failure could occur when the MP_JOIN handshake fails.
Section 3.6 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 during
the lifetime of the connection, unless the other host refreshes the
information with a REMOVE_ADDR and then an ADD_ADDR for the same
address.
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 these. Behaviour
could also be learnt for particularly badly performing subflows or
subflows that regularly fail during use, in order to temporarily
choose not to use these paths.
4. Semantic Issues
In order to support multipath operation, the semantics of some TCP
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components have changed. To aid clarity, this section collects these
semantic changes as a 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 one octet of sequence space. There is an explicit mapping
of data sequence space to subflow sequence space, which is
signalled 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
acknowledgement from the subflow ACKs. Instead an explicit data-
level ACK is used. This avoids possible deadlock scenarios when a
non-TCP-aware middlebox pro-actively ACKs at the subflow level,
and separates subflow- and connection-level processing at an end
host.
Duplicate ACK: A duplicate ACK that includes MPTCP signalling MUST
NOT be treated as a signal of congestion. To avoid any non-MPTCP-
aware entities also mistakenly seeing duplicate ACKs in such
cases, MPTCP SHOULD NOT send more than two duplicate ACKs
containing 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 for this subflow) that is available at the
receiver. This is the same semantics as 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 flow control role is preserved. Note that some
middleboxes may change the receive window, and so a host must use
the maximum value of those recently seen on the constituent
subflows for the connection-level receive window, and also need 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. A connection is
considered reset if a RST is received on every subflow.
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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 Address message, or implicitly
through a Join) has no implication 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 [5].
5. Security Considerations
As identified in [16], the addition of multipath capability to TCP
will bring with it a number of new classes of threat. In order to
prevent these, [3] 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, and the key security
requirements are:
o Provide a mechanism to confirm that the parties in a subflow
handshake are the same as in the original connection setup.
o Provide verification that the peer can receive traffic at a new
address before using it as part of a connection.
o 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 documented in Section 3.1 and Section 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 never again in
the clear. To ease demultiplexing whilst not giving away any
cryptographic material, future subflows use a truncated SHA-1 hash of
this key as the connection identification "token". The keys are
combined and used as keys in a MAC, and this should verify that the
parties in the handshake are the same as 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, and therefore the handshakes use single-use
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random numbers (nonces) at both ends - this ensures the MAC will
never be the same on two handshakes. The use of crypto capability
bits in the initial connection handshake to negotiate use of a
particular algorithm will allow the deployment of additional crypto
mechanisms in the future. Note that this would be susceptible to
bid-down attacks only if the attacker was on-path (and thus would be
able to modify the data anyway). The security mechanism presented in
this draft should therefore protect against all forms of flooding and
hijacking attacks suggested in [16].
6. Interactions with Middleboxes
Multipath TCP was designed to be deployable in the present world.
Its design takes into account "reasonable" existing middlebox
behaviour. 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 has made
to accomodate the different middleboxes.
A primary concern is our use of a new TCP option. Most middleboxes
should just forward packets with new options unchanged, yet there are
some that don't. These we expect will either 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 "Kind", and all message types are
defined by "subtype" values (see Section 8). This should reduce the
chances of only some types of MPTCP options being passed, and 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 or the return path, the initiating
host can fall back to regular TCP, as illustred in Figure 15 and
discussed in Section 3.1.
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 either with SYN/ACK or RST (subflow
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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.
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 15: Connection Setup with Middleboxes that Strip Options from
Packets
We now examine data flow with MPTCP, assuming the flow is correctly
setup, which implies 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 has been discussed
in the Fallback section. If a fraction of options are stripped,
behaviour 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, as discussed
in Section 3.3. 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 behaviour
and the elements in the MPTCP design that allow operation through
such middleboxes. Issues surrounding dropping packets with options
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or stripping options were discussed above, and are not included here:
o NAT [17]: Network Address (and Port) Translators change the source
address (and often source port) of packets. This means that a
host will not know its public-facing address for signalling in
MPTCP. Therefore, MPTCP permits implicit address addition via the
MP_JOIN option, and the handshake mechanism ensures that
connection attempts to private addresses [14] do not cause
problems. Explicit address removal is undertaken by an ID number
to allow no knowledge of the source address.
o Performance Enhancing Proxies (PEPs) [18]: might pro-actively ACK
data to increase performance. Problems will occur if a PEP ACKs
data and then fails before sending it on to the receiver, of if
the receiver is mobile and moves away before proactively ACKed
data is forwarded on. If subflow ACKs were used to control send
buffering, the data could be lost and never be retransmitted, thus
causing the subflow to permanently stall. MPTCP therefore uses
the DATA_ACK to make progress when one of its subflows fails in
this way. This is why MPTCP does not use subflow ACKs to infer
connection level ACKs.
o Traffic Normalizers [19]: may not allow holes in sequence numbers,
and 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.
o Firewalls [20]: might perform initial sequence number
randomization on TCP connections. MPTCP uses relative sequence
numbers in data sequence mapping to cope with this. Like NATs,
firewalls will not permit many incoming connections, so MPTCP
supports address signalling (ADD_ADDR) so that a multi-addressed
host can invite its peer behind the firewall/NAT to connect out to
its additional interface.
o Intrusion Detection Systems: look out for traffic patterns and
content that could threaten a network. Multipath will mean that
such data is potentially spread, so it is more difficult for an
IDS to analyse the whole traffic, and potentially increases the
risk of false positives. However, for an MPTCP-aware IDS, tokens
can be read by such systems to correlate multiple subflows and re-
assemble for analysis.
o Application level middleboxes: such as content-aware firewalls may
alter the payload within a subflow, such as re-writing URIs in
HTTP traffic. MPTCP will detect these using the checksum and
close the affected subflow(s), if there are other subflows that
can be used. If all subflows are affected multipath will fallback
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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:
o 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 capabilities. If such a packet does not get
through, MPTCP will end up falling back to regular TCP.
o 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.
o The Receive Window: may be shrunk by some middleboxes at the
subflow level. MPTCP will use the maximum window at data-level,
but will also obey subflow specific windows.
7. Acknowledgements
The authors are supported by Trilogy
(http://www.trilogy-project.org), a research project (ICT-216372)
partially funded by the European Community under its Seventh
Framework Program. The views expressed here are those of the
author(s) only. The European Commission is not liable for any use
that may be made of the information in this document.
The authors gratefully acknowledge significant input into this
document from Sebastien Barre, Christoph Paasch 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 and Bob Briscoe.
8. IANA Considerations
This document will make a request to IANA to allocate a new TCP
option value for MPTCP. This value will be the value of the "Kind"
field seen in all MPTCP options in this document.
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This document will also request IANA operates a registry for MPTCP
option subtype values. The values as defined by this specification
are as follows:
+-------------+-----------------------------+---------------+-------+
| Symbol | Name | Ref | Value |
+-------------+-----------------------------+---------------+-------+
| MP_CAPABLE | Multipath Capable | Section 3.1 | 0x0 |
| MP_JOIN | Join Connection | Section 3.2 | 0x1 |
| DSS | Data Sequence Signal (Data | Section 3.3 | 0x2 |
| | ACK and Data Sequence | | |
| | Mapping) | | |
| ADD_ADDR | Add Address | Section 3.4.1 | 0x3 |
| REMOVE_ADDR | Remove Address | Section 3.4.2 | 0x4 |
| MP_PRIO | Change Subflow Priority | Section 3.3.8 | 0x5 |
| MP_FAIL | Fallback | Section 3.5 | 0x6 |
+-------------+-----------------------------+---------------+-------+
Table 1: MPTCP Option Subtypes
This document also requests that IANA keeps a registry of
cryptographic handshake algorithms based on the flags in MP_CAPABLE
(Section 3.1). This document specifies only one algorithm:
+-------+-----------+----------------------------+
| Flags | Algorithm | Document |
+-------+-----------+----------------------------+
| 0x1 | HMAC-SHA1 | This document, Section 3.2 |
+-------+-----------+----------------------------+
Table 2: MPTCP Handshake Algorithms
9. References
9.1. Normative References
[1] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
9.2. Informative References
[2] Postel, J., "Transmission Control Protocol", STD 7, RFC 793,
September 1981.
[3] Ford, A., Raiciu, C., Handley, M., Barre, S., and J. Iyengar,
"Architectural Guidelines for Multipath TCP Development",
draft-ietf-mptcp-architecture-05 (work in progress),
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January 2011.
[4] Raiciu, C., Handley, M., and D. Wischik, "Coupled Congestion
Control for Multipath Transport Protocols",
draft-ietf-mptcp-congestion-01 (work in progress),
January 2011.
[5] Scharf, M. and A. Ford, "MPTCP Application Interface
Considerations", draft-ietf-mptcp-api-00 (work in progress),
November 2010.
[6] Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP
Selective Acknowledgment Options", RFC 2018, October 1996.
[7] Allman, M., Paxson, V., and W. Stevens, "TCP Congestion
Control", RFC 2581, April 1999.
[8] Gont, F., "Security Assessment of the Transmission Control
Protocol (TCP)", draft-ietf-tcpm-tcp-security-02 (work in
progress), January 2011.
[9] Eastlake, D. and T. Hansen, "US Secure Hash Algorithms (SHA and
HMAC-SHA)", RFC 4634, July 2006.
[10] Eastlake, D., Schiller, J., and S. Crocker, "Randomness
Requirements for Security", BCP 106, RFC 4086, June 2005.
[11] Krawczyk, H., Bellare, M., and R. Canetti, "HMAC: Keyed-Hashing
for Message Authentication", RFC 2104, February 1997.
[12] Jacobson, V., Braden, B., and D. Borman, "TCP Extensions for
High Performance", RFC 1323, May 1992.
[13] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition of
Explicit Congestion Notification (ECN) to IP", RFC 3168,
September 2001.
[14] Rekhter, Y., Moskowitz, R., Karrenberg, D., Groot, G., and E.
Lear, "Address Allocation for Private Internets", BCP 5,
RFC 1918, February 1996.
[15] Braden, R., "Requirements for Internet Hosts - Communication
Layers", STD 3, RFC 1122, October 1989.
[16] Bagnulo, M., "Threat Analysis for TCP Extensions for Multi-path
Operation with Multiple Addresses", draft-ietf-mptcp-threat-08
(work in progress), January 2011.
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[17] Srisuresh, P. and K. Egevang, "Traditional IP Network Address
Translator (Traditional NAT)", RFC 3022, January 2001.
[18] Border, J., Kojo, M., Griner, J., Montenegro, G., and Z.
Shelby, "Performance Enhancing Proxies Intended to Mitigate
Link-Related Degradations", RFC 3135, June 2001.
[19] Handley, M., Paxson, V., and C. Kreibich, "Network Intrusion
Detection: Evasion, Traffic Normalization, and End-to-End
Protocol Semantics", Usenix Security 2001, 2001, <http://
www.usenix.org/events/sec01/full_papers/handley/handley.pdf>.
[20] Freed, N., "Behavior of and Requirements for Internet
Firewalls", RFC 2979, October 2000.
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 have 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 we propose using in this draft.
SYN packets typically include MSS (4 bytes), window scale (3 bytes),
SACK permitted (2 bytes) and timestamp (10 bytes) options. Together
these sum to 19 bytes. Some operating systems appear to pad each
option up to a word boundary, thus using 24 bytes (a brief survey
suggests Windows XP and Mac OS X do this, whereas Linux does not).
Optimistically, therefore, we have 21 bytes spare, or 16 if it has to
be word-aligned. In either case, however, the SYN versions of
Multipath Capable (12 bytes) and Join (12 or 16 bytes) options will
fit in this remaining space.
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 Data Sequence Signal (DSS) option varies in
length depending on whether the Data Sequence Mapping and DATA ACK
are included, and whether the sequence numbers in use are 4 or 8
octets. 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 bi-
directional and high-bandwidth, it is unlikely that all that option
space will be required on each DSS option.
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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 following packet). Other options include: alternating between 4
and 8 byte sequence numbers in each option; and sending the DATA_ACK
on a duplicate subflow-level ACK (although note that this must not be
taken as a signal of congestion).
On subflow and connection setup, an MPTCP option is also set on the
third packet (an ACK). These are 20 bytes (for Multipath Capable)
and 24 bytes (for Join) - both of which will fit in the available
option space.
Pure ACKs in TCP typically contain only timestamps (10B). Here,
multipath TCP typically needs to encode only the DATA ACK (maximum of
12 octets). 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 accomodate 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 8 and 22 bytes, depending on
whether IPv4 or IPv6 is used, and whether the port number is present
or not. It is unlikely that such signalling would fit in a data
packet (although if there is space, it is fine to include it). It is
recommended to use duplicate ACKs with no other payload or options in
order to transmit these rare signals. Note this is the reason for
mandating that duplicate ACKs with MPTCP options are not taken as a
signal of congestion.
Finally, there are issues with reliable delivery of options. As
options can also be sent on pure ACKs, these are not reliably sent.
This is not an issue for DATA_ACK due to their cumulative nature, but
may be an issue for ADD_ADDR/REMOVE_ADDR options. Here, it is
recommended to send these options redundantly (whether on multiple
paths, or on the same path on a number of ACKs - but interspersed
with data in order to avoid interpretation as congestion). The cases
where options are stripped by middleboxes are discussed in Section 6.
Appendix B. Control Blocks
Conceptually, an MPTCP connection can be represented as an MPTCP
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control block 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.
RFC793 [2] specifies several state variables. Whenever possible, we
reuse the same terminology as RFC793 to describe the state variables
that are maintained by MPTCP.
B.1. MPTCP Control Block
The MPTCP control block contains the following variable per-
connection.
B.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, 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 C bit the MP_CAPABLE options exchanged
during connection establishment, and is set to false otherwise.
If this flag is set, the checksum must be computed in all DSS
options.
B.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 with RFC1323, 16 bits without): This is the sending
window. MPTCP maintains the sending 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
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SEQ.WND value which is sent in each transmitted segment.
B.1.3. Receiving Side
RCV.NXT (64 bits): This is the Data Sequence Number of the next byte
which 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 which is sent in the DSS option on
all subflows.
RCV.WND (32bits with RFC1323, 16 bits otherwise): This is the
connection-level receive window, which is the maximum of the
RCV.WND on all the subflows.
B.2. TCP Control Blocks
The MPTCP control block also contains a list of the TCP control
blocks that are associated to 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:
B.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 acknowledgement 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.
B.2.2. Receiving Side
RCV.NXT (32 bits): This is the sequence number of the next byte
which 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 with RFC1323, 16 bits otherwise): This is the
subflow-level receive window which is updated with the window
field from the segments received on this subflow.
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Appendix C. Changelog
This section maintains logs of significant changes made to this
document between versions.
C.1. Changes since draft-ietf-mptcp-multiaddressed-02
o Changed to using a single TCP option with a sub-type field.
o Merged Data Sequence Number, DATA ACK, and DATA FIN.
o Changed DATA FIN behaviour (separated from subflow FIN).
o Added crypto agility and checksum negotiation.
o Redefined MP_JOIN handshake to use only three TCP options.
o Added pseudo-header to checksum.
o Many clarifications and re-structuring.
o Added more discussion on heuristics.
C.2. Changes since draft-ietf-mptcp-multiaddressed-01
o Added proposal for hash-based security mechanism.
o Added receiver subflow policy control (backup path flags and
MP_PRIO option).
o Changed DSN_MAP checksum to use the TCP checksum algorithm.
C.3. Changes since draft-ietf-mptcp-multiaddressed-00
o Various clarifications and minor re-structuring in response to
comments.
C.4. Changes since draft-ford-mptcp-multiaddressed-03
o Clarified handshake mechanism, especially with regard to error
cases (Section 3.2).
o Added optional port to ADD_ADDR and clarified situation with
private addresses (Section 3.4.1).
o Added path liveness check to REMOVE_ADDR (Section 3.4.2).
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o Added chunk checksumming to DSN_MAP (Section 3.3.1) to detect
payload-altering middleboxes, and defined fallback mechanism
(Section 3.5).
o Major clarifications to receive window discussion (Section 3.3.5).
o Various textual clarifications, especially in examples.
C.5. Changes since draft-ford-mptcp-multiaddressed-02
o Remove Version and Address ID in MP_CAPABLE in Section 3.1, and
make ISN be 6 bytes.
o Data sequence numbers are now always 8 bytes. But in some cases
where it is unambiguous it is permissible to only send the lower 4
bytes if space is at a premium.
o Clarified behaviour of MP_JOIN in Section 3.2.
o Added DATA_ACK to Section 3.3.
o Clarified fallback to non-multipath once a non-MP-capable SYN is
sent.
Authors' Addresses
Alan Ford
Roke Manor Research
Old Salisbury Lane
Romsey, Hampshire SO51 0ZN
UK
Phone: +44 1794 833 465
Email: alan.ford@roke.co.uk
Costin Raiciu
University College London
Gower Street
London WC1E 6BT
UK
Email: c.raiciu@cs.ucl.ac.uk
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Mark Handley
University College London
Gower Street
London WC1E 6BT
UK
Email: m.handley@cs.ucl.ac.uk
Olivier Bonaventure
Universite catholique de Louvain
Pl. Ste Barbe, 2
Louvain-la-Neuve 1348
Belgium
Email: olivier.bonaventure@uclouvain.be
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