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draft-ietf-mptcp-multiaddressed
Internet Engineering Task Force A. Ford
Internet-Draft Roke Manor Research
Intended status: Experimental C. Raiciu
Expires: September 9, 2010 M. Handley
University College London
March 8, 2010
TCP Extensions for Multipath Operation with Multiple Addresses
draft-ford-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 - 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 to IETF in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that
other groups may also distribute working documents as Internet-
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and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
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The list of current Internet-Drafts can be accessed at
http://www.ietf.org/ietf/1id-abstracts.txt.
The list of Internet-Draft Shadow Directories can be accessed at
http://www.ietf.org/shadow.html.
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This Internet-Draft will expire on September 9, 2010.
Copyright Notice
Copyright (c) 2010 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the BSD License.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. Design Assumptions . . . . . . . . . . . . . . . . . . . . 4
1.2. Layered Representation . . . . . . . . . . . . . . . . . . 5
1.3. Operation Summary . . . . . . . . . . . . . . . . . . . . 6
1.4. Requirements Language . . . . . . . . . . . . . . . . . . 7
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 7
3. Semantic Issues . . . . . . . . . . . . . . . . . . . . . . . 8
4. MPTCP Protocol . . . . . . . . . . . . . . . . . . . . . . . . 9
4.1. Connection Initiation . . . . . . . . . . . . . . . . . . 9
4.2. Starting a New Subflow . . . . . . . . . . . . . . . . . . 11
4.3. Address Knowledge Exchange (Path Management) . . . . . . . 13
4.3.1. Address Advertisement . . . . . . . . . . . . . . . . 14
4.3.2. Remove Address . . . . . . . . . . . . . . . . . . . . 16
4.4. General MPTCP Operation . . . . . . . . . . . . . . . . . 16
4.4.1. Data Sequence Numbering . . . . . . . . . . . . . . . 17
4.4.2. Data Acknowledgements . . . . . . . . . . . . . . . . 19
4.4.3. Receiver Considerations . . . . . . . . . . . . . . . 19
4.4.4. Sender Considerations . . . . . . . . . . . . . . . . 20
4.4.5. Congestion Control Considerations . . . . . . . . . . 21
4.4.6. Subflow Policy . . . . . . . . . . . . . . . . . . . . 21
4.5. Closing a Connection . . . . . . . . . . . . . . . . . . . 22
4.6. Error Handling . . . . . . . . . . . . . . . . . . . . . . 24
5. Security Considerations . . . . . . . . . . . . . . . . . . . 24
6. Interactions with Middleboxes . . . . . . . . . . . . . . . . 25
7. Interfaces . . . . . . . . . . . . . . . . . . . . . . . . . . 28
8. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 28
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 29
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 30
11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 30
11.1. Normative References . . . . . . . . . . . . . . . . . . . 30
11.2. Informative References . . . . . . . . . . . . . . . . . . 30
Appendix A. Notes on use of TCP Options . . . . . . . . . . . . . 31
Appendix B. Signaling Control Information in the Payload . . . . 32
Appendix C. Resync Packet . . . . . . . . . . . . . . . . . . . . 32
Appendix D. Changelog . . . . . . . . . . . . . . . . . . . . . . 34
D.1. Changes since draft-ford-mptcp-multiaddressed-02 . . . . . 34
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 34
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1. Introduction
Multipath TCP (henceforth referred to as MPTCP) is set of extensions
to regular TCP [2] to allow a transport connection to operate across
multiple paths simultaneously. This document presents the protocol
changes required by Multipath 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 several others:
o Architecture [3], which explains the motivations behind Multipath
TCP and 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 endpoints are multihomed and
multiaddressed
To simplify the design we assume that the presence of multiple
addresses at an endpoint 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.
There are three aspects to the backwards-compatibility listed above
(discussed in more detail in [3]):
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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).
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.
Areas for further study:
o In theory, since this is purely a TCP extension, it should be
possible to use MPTCP with both IPv4 and IPv6 subflows for the
same connection on dual-stack hosts, thus having the additional
possible benefit of aiding transition.
o Some features of the design presented here could be extended to
work with non-multi-addressed hosts by using other packet metadata
(such as ports or flow label), packet marking, or partial
multipath (such as by using a proxy).
1.2. Layered Representation
MPTCP operates at the transport layer, and its existence aims to be
transparent to both higher and lower layers. It is a set of
additional features on top of standard TCP, and as such MPTCP is
designed to be usable by legacy applications with no changes. A
possible implementation would be for such a feature to be a system-
wide setting: "Use multipath TCP by default? Y/N". Multipath-aware
applications would be able to use an extended sockets API to have
further influence on the behaviour of MPTCP. Figure 1 illustrates
this layering.
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+-------------------------------+
| Application |
+---------------+ +-------------------------------+
| Application | | MPTCP |
+---------------+ + - - - - - - - + - - - - - - - +
| TCP | | Subflow (TCP) | Subflow (TCP) |
+---------------+ +-------------------------------+
| IP | | IP | IP |
+---------------+ +-------------------------------+
Figure 1: Comparison of Standard TCP and MPTCP Protocol Stacks
Detailed discussion of an architecture for developing a multipath TCP
implementation, especially regarding the functional separation by
which different components should be developed, is given in [3].
1.3. Operation Summary
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 4.
o To a non-MPTCP-aware application, MPTCP will be indistinguishable
from normal TCP. All MPTCP operation is handled by the MPTCP
implementation, although extended APIs could provide additional
control and influence [5]. An application begins by opening a TCP
socket in the normal way.
o An MPTCP connection begins as a single TCP session. This is
illustrated in Figure 2 as being between Addresses A1 and B1 on
Hosts A and B respectively.
o If extra paths are available, additional TCP sessions 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.
o MPTCP identifies multiple paths by the presence of multiple
addresses at endpoints. 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 TCP sessions (termed
'subflows') will be achieved through a path management method.
This document describes a mechanism by which an endpoint can
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initiate new subflows by using its additional addresses, or by
signalling its available addresses to the other endpoint.
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. Connections are terminated by connection-level FIN
packets as well as those relating to the individual subflows.
Host A Host B
------------------------ ------------------------
Address A1 Address A2 Address B1 Address B2
---------- ---------- ---------- ----------
| | | |
| (initial connection setup) | |
|----------------------------------->| |
|<-----------------------------------| |
| | | |
| (additional subflow setup) |
| |--------------------->| |
| |<---------------------| |
| | | |
| | | |
Figure 2: Example MPTCP Usage Scenario
1.4. 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].
2. 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 stream of TCP packets sent over a path. A subflow is a
component part of a connection between two endpoints.
Connection: A collection of one or more subflows, over which an
application can communicate between two endpoints. There is a
one-to-one mapping between a connection and a socket.
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Data-level: The payload data is nominally transfered 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 an endpoint. May also be referred to as a "Connection ID".
Endpoint: A host operating an MPTCP implementation, and either
initiating or terminating a MPTCP connection.
3. Semantic Issues
In order to support multipath operation, the semantics of some TCP
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. 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 the subflow
sequence number only, not the data-level sequence space. Although
data acknowledgments could be inferred from the subflow ACK, an
explicit connection-level DATA_ACK is used to ensure end-to-end
reliability in the presense of certain types of middlebox.
Receive Window: The receive window in the TCP header indicates the
amount of free buffer space for this connection (as opposed to for
this subflow) that is available at the receiver. This is a change
to the semantics of the field. With regular TCP the window is
relative to the acknowledgment number in the TCP header. This is
not meaningful for multipath TCP. Instead with multipath TCP the
receive window is relative to the DATA_ACK field, indicating the
amount of buffer space available at the data-level. This permits
the receive window to serve its original purpose and provide flow-
control of the data sent by the TCP sending application.
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.
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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.
Address List: Address management 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 has
no implication whatsoever for other connections between the same
pair of hosts.
5-tuple: The 5-tuple (protocol, local address, local port, remote
address, remote port) presented 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.
These API issues are discussed in more detail in [5].
4. 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.
These TCP Options will have option numbers allocated by IANA, as
listed in Section 10, and are defined throughout the following
subsections.
4.1. Connection Initiation
Connection Initiation begins with a SYN, SYN/ACK exchange on a single
path. Each of these packets will additionally feature the MP_CAPABLE
TCP option (Figure 3) This option declares its sender is capable of
performing multipath TCP and wishes to do so on this particular
connection). As well as this declaration, this field presents a
locally-unique token identifying this connection. This is used when
adding additional subflows to this connection.
This token is generated by the sender and has local meaning only,
hence it MUST be unique for the sender. The token MUST be difficult
for an attacker to guess, and thus it is recommended it SHOULD be
generated randomly. (However, see further discussions about security
in Section 5, including the possibility of 64-bit tokens.)
This option is only present in packets with the SYN flag set. It is
only used in the first TCP session of a connection, in order to
identify the connection; all following connections will use path
management options (see Section 4.2) to join the existing connection.
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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=MP_CAP | Length=11 | Sender Token :
+---------------+---------------+---------------+---------------+
: Sender Token (cont 4 octets) | Initial Data Sequence Number :
+-----------------------------------------------+---------------+
: Initial Data Sequence Number (cont - 6 bytes) |
+-----------------------------------------------+---------------+
Figure 3: Multipath 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 will 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 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. Once the active opener has
sent a SYN without the MP_CAPABLE option, it MUST fall back to
regular TCP behavior, even if it subsequently receives a SYN/ACK that
contains an MP_CAPABLE option. This might happen if the MP_CAPABLE
SYN and subsequent non-MP-capable SYN are reordered. This is to
ensure that the two endpoints end up in an interoperable state, no
matter what order the SYNs arrive at the passive opener. This final
state is inferred from the presence or absence of the DATA_ACK option
in the third packet of the TCP handshake.
The MPC option includes the most significant 6 bytes of the 8-byte
initial Data Sequence Number option (discussed in Section 4.4). The
least significant two bytes should be zeroed. This is also used as
an implicit mapping of the SYN to the data sequence space (and this
initial SYN counts as one octet in this space, as for a regular SYN
in single-path TCP). This will be used to ensure both ends agree on
whether the connection is multipath or standard TCP, regardless of
middlebox behaviour. This could also have some (minor) security
benefits, discussed in Section 5. To preserve option space, only the
most significant six bytes are sent in the SYN, as there is no
significant security benefit from randomizing the values of the lower
two bytes given that these fall within typical receive windows sizes.
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4.2. Starting a New Subflow
Endpoints have knowledge of their own address(es), and can become
aware of the other endpoint's addresses through signalling exchanges
as described in Section 4.3. Using this knowledge, an endpoint can
initiate a new subflow over a currently unused pair of addresses.
Either endpoint that is part of a connection can initiate the
creation of a new subflow.
A new subflow is started as a normal TCP SYN/ACK exchange. The
"Join" TCP option (Figure 4) is used to identify of which connection
the new subflow should become a part. The token used is the locally
unique token of the destination for the subflow, as defined by the
MP_CAPABLE option received in the first SYN/ACK exchange.
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=OPT_JOIN | Length = 7 |Receiver Token (4 octets total):
+---------------+---------------+----------------+--------------+
: Receiver Token (continued) | Address ID |
+-------------------------------+----------------+
Figure 4: Join Connection option
In response to a SYN with the "Join" option, if the token is valid
for an existing MPTCP connection, the recipient MUST respond with a
SYN/ACK also containing a "Join" option, with the initiator's token.
This serves two purposes: firstly, to ensure both endpoints agree on
the connection being referred to (this is particularly relevant when
both addresses being used are new to the connection); and secondly,
to ensure there are no middleboxes in the path that will drop MPTCP
options on the return path. This behaviour is illustred in Figure 5.
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Host A Host B
------------------------ ------------------------
Address A1 Address A2 Address B1 Address B2
---------- ---------- ---------- ----------
| | | |
| SYN + OPT_MPC(Token A) | |
|----------------------------------->| |
|<-----------------------------------| |
| SYN/ACK + OPT_MPC(Token B) | |
| | | |
| | SYN + OPT_JOIN(Token B) |
| |----------------------------------->|
| |<-----------------------------------|
| | SYN/ACK + OPT_JOIN(Token A) |
| | | |
Figure 5: Example use of MPTCP Tokens
If the token is unknown, the recipient MUST respond with a TCP RST in
the same way as when an unknown TCP port is used.
It should be noted that additional subflows can exist between any
pair of ports; no explicit accept calls or bind calls are required to
open additional subflows. To associate a new subflow to an existing
connection, the token supplied in the subflow's SYN exchange is used
for demultiplexing. This means that port numbers on subflow SYN
exchanges are not important, and a receiver of a SYN SHOULD allow any
values to be used, as long as the 5-tuple is unique for each host.
However the sender of a SYN containing a JOIN option SHOULD send the
SYN to the port used by the remote party for the first subflow in the
connection. The local port for such SYNs MAY be chosen locally,
either dynamically, or by the application if an API allows the
application to do so. 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.
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 JOIN option includes an "Address ID". This is an identifier,
locally unique to the sender of this option, and with only per-
connection relevance, which identifies the source address of this
packet. The key purpose of this identifier is, if an address becomes
unexpectedly unavailable on the sender, it can signal this to the
receiver via a remove address option (Section 4.3.2) without needing
to know what the source address actually is (thus allowing the use of
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NATs). It also allows correlation between new connection attempts
and address signalling (Section 4.3.1), to prevent duplicate subflow
initiation.
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.
The Address ID must 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.
This option can only be present when the SYN flag is set.
4.3. Address Knowledge Exchange (Path Management)
We use the term "path management" to refer to the exchange of
information about additional paths between endpoints, which in this
design is managed by multiple addresses at endpoints. For more
detail of the architectural thinking behind this design, see the
separate 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 4.2, where the initiator has an
additional address. The second method is described in the following
subsections, whereby addresses are signalled explicitly to the other
endpoint, to allow it to initiate new connections. This approach, of
two complementary mechanisms, has been chosen to allow addresses to
change in flight, and thus support operation through NATs, whilst
also allowing the signalling of previously unknown addresses, such as
those belonging to other address families (e.g. IPv4 and IPv6).
Here is an example of typical operation of the protocol:
o An endpoint that is multihomed starts an additional TCP session to
an address/port pair that is already in use on the other endpoint,
using a token to identify the flow (Section 4.2). (A multihomed
destination may open a new subflow from its new address to an
existing subflow's source address and port, or a multihomed source
may open a new subflow from its new address to an existing
subflow's destination and port).
o More concretely, say a connection is intiated from host "A" on
(address, port) combination A1 to destination (address, port) B1
on host "B". If host A is multihomed, it starts an additional
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connection from new (address, port) A2 to B1, using B's previously
declared token. Alternatively, if B is multhomed, it will try to
set up a new TCP connection from B2 to A1, using A's previously
declared token.
o Simultaneously (or after a timeout), an "Add Address" option
(Section 4.3.1) 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. Using the previous notation, this would be an
Add Address packet sent from A1 to B1, informing B of address A2.
o The mix of using the SYN-based option and the Add Address option,
including timeouts, is implementation-specific and can be tailored
to agree with local policy.
o If host B successfully receives the first SYN, starting a new
subflow, it can use the Address ID in the Join option to correlate
this with the Add Address option that will also arrive on an
existing subflow. Assuming the endpoint has already responded to
the SYN with a SYN/ACK, it will know to ignore the Add Address
option. Otherwise, if it has not received such a SYN, it will try
to initiate a new subflow from one or more of its addresses to
address A2 (triggered by the Add Address option). This is
intended to permit new sessions to be opened if one endpoint is
behind a NAT. A slight security improvement can be gained if a
host ensures there is a correlated Add Address option before
responding to the SYN.
Other scenarios are valid, however, such as those where entirely new
addresses are signalled, e.g. to allow an IPv6 and an IPv4 path to be
used simultaneously.
4.3.1. Address Advertisement
The Add Address TCP Option announces additional addresses on which an
endpoint can be reached (Figure 6), which allows several (ID,
address) pairs to be announced to the other endpoint. Multiple
addresses can be added 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 address removal, and
therefore endpoints must cache the mapping between ID and address.
This is also used to identify Join Connection options (Section 4.2)
relating to the same address, even when address translators are in
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use. The ID must be unique to the sender and connection, per
address, but its mechanism for allocating such IDs is implementation-
specific.
This option is shown for IPv4. For IPv6, the IPVer field will read
6, and the length of the address will be 16 octets not 4, and thus
the length of the option will be 2 + (18 * number_of_entries). If
there is sufficient TCP option space, multiple addresses can be
included, with an ID following on immediately from the previous
address, and their existance can be inferred through the option
length and version fields.
NB: by having a IPVer field, we get four free reserved bits. These
could be used in later versions of this protocol for expressing
sender policy, e.g. one bit for "use now" or similar, to
differentiate between subflows for backup purposes and those for
throughput.
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=OPT_ADDR | Length | Address ID | IPVer |(resvd)|
+---------------+---------------+---------------+-------+-------+
| Address (IPv4 - 4 octets) |
+---------------------------------------------------------------+
( ... further ID/Version/Address fields as required ... )
Figure 6: Add Address option (for IPv4)
Ideally, we'd like to ensure the Add Address (and Remove Address)
option is sent reliably and in order to the other end. This is to
ensure that we don't close the connection when remove/add addresses
are processed in reverse order, and 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.
Subflow level ACKs do not cover options, so if we want explicit
guarantees we need to build in other mechanisms. Solutions include
echoing the options and sending one option per RTT, or adding a
sequence number to the option which is explicitly acked in another
option. However, we feel these mechanisms' added complexity is not
worth the benefits they bring. There are two basic failure modes for
options: a) every new option gets stripped or b) some options get
stripped, randomly. The second option looks more like a middlebox
implementation error, so we believe it is not worth optimizing for.
In the first case, resending the option on a different subflow is the
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thing to do. To achieve similar reliability without explicit ACKs,
we propose sending all Add/Remove Address options on all existing
subflows. If ordering is needed, we should only send one add/remove
option per RTT (modulo lost packets at subflow level).
If an address index is in use, the Add Address option SHOULD be
silently ignored.
4.3.2. Remove Address
If, during the lifetime of a MPTCP connection, a previously-announced
address becomes invalid (e.g. if the interface disappears), the
affected endpoint should announce this so that the other endpoint can
remove subflows related to this address.
This is achieved through the Remove Address option (Figure 7), which
will remove a previously-added address (or list of addresses) from a
connection and terminate any subflows currently using that address.
The sending and receipt of this message should trigger the sending of
FINs by both endpoints on the affected subflow(s) (if possible), as a
courtesy to cleaning up middlebox state, but endpoints may clean up
their internal state without a long timeout.
Address removal is undertaken by ID, so as to permit the use of NATs
and other middleboxes. If there is no address at the requested ID,
the receiver will silently ignore the request.
The standard way to close a subflow (so long as it is still
functioning) is to use a FIN exchange as in regular TCP - for more
information, see Section 4.5.
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=OPT_REMADR| Length = 2+n | Address ID | ...
+---------------+---------------+---------------+
Figure 7: Remove Address option
4.4. 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
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.
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4.4.1. Data Sequence Numbering
The data stream as a whole can be reassembled through the use of the
Data Sequence Mapping (Figure 8) option, which defines the mapping
from the data sequence number to the subflow 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.
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=OPT_DSN | Length | Data Sequence Number ... :
+---------------+---------------+------------------------------+
: ... ( (length-8) octets ) | Data-level Length (2 octets) |
+-------------------------------+------------------------------+
| Subflow Sequence Number (4 octets) |
+-------------------------------+------------------------------+
Figure 8: Data Sequence Mapping option
This option specifies a full mapping from data sequence number to
subflow sequence number, informing the receiver that there is a one-
to-one correspondence between the two sequence spaces for the
specified length. 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).
The data sequence number specified in this option is absolute,
whereas the subflow sequence numbering is relative (the SYN at the
start of the subflow has subflow sequence number 1). This is to
permit middleboxes that may wish to alter sequence numbering, since
the data stream itself will not be affected.
TBD: if we used absolute sequence numbers that would make receiver
code a bit simpler, and would make it more difficult to inject data
as the attacker needs to guess both Data Sequence Number and Subflow
Sequence Number. How many middleboxes are there that change the
sequence numbers, and should we optimize for them?
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
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data sequence number can be mapped on different subflows for
retransmission purposes (see Section 4.4.4).
A receiver MUST NOT accept data for which it does not have a mapping
to the data sequence space. To do this, the receiver will not
acknowledge the unmapped data at subflow level. It is better to have
a subflow fail than to accept data in the wrong order. However, if
there was a lost packet in the subflow, the receiver SHOULD wait for
this to be retransmitted before closing the subflow, since the lost
packet may contain the necessary mapping information.
NOTE: if the subflow did ACK data for which it did not have a
mapping, it would be possible to use the DATA_ACK to detect when the
mapping was lost. This will likely not increase reliability, as the
subflow will likely drop all unknown options. In addition, the
receiver is now storing potentially useless data: what happens if the
mapping never arrives? Should the receiver have a timer to delete
this data?
Data sequence numbers are always 64-bit quantities, and should be
maintained as such in implementations. If a connection is
progressing at a slow rate, so that protection against wrapped
sequence numbers is not required, and security requirements against
blind insertion attacks are not stringent, then it is permissible to
include just the lower 32 bits of the sequence number in the OPT_DSN
option as an optimization. Implementations MUST accept this and
implicitly promote it to a 64-bit quantity. In all other cases, the
full 64 bits should be included. Security implications are discussed
in Section 5.
As with the standard TCP sequence number, the data sequence number
should not start at zero, but at a random value to make session
hijacking harder. This is done by including a Data Sequence Mapping
option along with the MP_CAPABLE option in the initial SYN (which
occupies one octet of data sequence space; see Section 4.1). In this
case, to save option space, neither the data-level length nor the
subflow sequence number fields are present in this option, so the
Length field will be the length of the Data Sequence Number, plus two
octets.
The 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 a 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 endpoints, another
is when segments of data are scheduled in larger than packet-sized
chunks.
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4.4.2. Data Acknowledgements
In a perfect world, it would be possible to make do with only
subflow-level acknowledgements, with the sender keeping track of
these acknowledgements to derive what data has been successfully
received. If there are ever cases where the subflow data is dropped
after it has been acked (which may occur if a proxy middlebox fails,
or if a buffer fills on a host), the connection will break entirely
since the sender will assume the data has been received when it
hasn't.
Therefore, MPTCP provides a connection-level acknowledgement (the
DATA_ACK) to act as a cumulative ACK for the connection as a whole.
This is analogous to the behaviour of the standard TCP cumulative ACK
in SACK - indicating how much data has been successfully received
(with no holes). This option, illustrated in Figure 9, is expected
to be included in every packet by an MPTCP host.
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=OPT_DACK | Length | Data Sequence Number ... :
+---------------+---------------+------------------------------+
: ... ( (length-8) octets ) |
+-------------------------------+
Figure 9: Connection-level Acknowledgement (DATA_ACK)
4.4.3. 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 (e.g.
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
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receive window. With multipath, such a check is done using only the
connection level window. A sanity check could be performed at
subflow level to ensure that: SSN-SUBFLOW_ACK <= DSN - DATA_ACK.
When should segments be processed at connection level? The default
is to wait until they arrive in order at subflow level, and only then
do connection level processing. However, one can optimize for speed
by processing at connection level segments that have not yet been
acked at subflow level; the only requirement for this optimization is
to have a valid data sequence mapping for the segment. Note that the
segment can be dropped at subflow level afterwards (e.g. because it
is out of order and there is more pressure); the DATA_ACK ensure the
connection can make progress without having to wait for the subflow
retransmission.
An issue will arise regarding how large a receive buffer to
implement. The lower bound would be the maximum bandwidth/delay
product of all paths, however this could easily fill when a packet is
lost on a slower subflow and needs to be retransmitted (see
Section 4.4.4). The upper bound would be the maximum RTT multiplied
by the maximum total bandwidth available. This will cover most
eventualities, but could easily become very large. It is FFS what
the best approach is.
4.4.4. Sender Considerations
The sender should only consider receive window advertisements where
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.
The data sequence mapping allows senders to re-send data with the
same data sequence number on a different subflow. When doing this,
an endpoint 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 4.4.6). 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
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retransmits are only used after a few subflow level retransmission
timeouts occur.
Whichever the retransmission strategy, the sender MUST keep data in
its send buffer as long as the data has not been acked at connection
level and on all subflows it 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, 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, and only then delete the
outstanding data segments.
A sender will maintain connection level timers for unacknowledged
segments. These timers will be based on the subflow timers, and will
guard against pro-active acking by middleboxes.
The send buffer must be, at the minimum, as big as the receive
buffer, to enable the sender to reach maximum throughput.
4.4.5. Congestion Control Considerations
Different subflows in an MPTCP connection have different congestion
windows. To achieve 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.
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. Much research
is expected in this area in the near future.
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 where lost and when.
4.4.6. 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 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.
There is no mechanism in MPTCP for a receiver to signal their own
particular preferences for paths, but this is a necessary feature
since receivers will often be the multihomed party, and may have to
pay for metered incoming bandwidth. Instead of incorporating complex
signalling, it is proposed to use existing TCP features to signal
priority implicitly. If a receiver wishes to keep a path active as a
backup but wishes to prevent data being sent on that path, it could
stop sending ACKs for any data it receives on that path. The sender
would interpret this as severe congestion or a broken path and stop
using it. We do not advocate this method, however, since this will
result in unnecessary retransmissions.
Therefore, a proposal is to use ECN [7] to to provide fake congestion
signals on paths that a receiver wishes to stop being used for data.
This has the benefit of causing the sender to back off without the
need to retransmit data unnecessarily, as in the case of a lost ACK.
This should be sufficient to allow a receiver to express their
policy, although does not permit a rapid increase in throughput when
switching to such a path.
TBD: This is clearly an overload of the ECN signal, and as such other
solutions, such as explicitly signalling path operation preferences
(such as in the reserved bits of certain TCP options, or through
entirely new options) may be a preferred solution.
4.5. Closing a Connection
Under single path TCP, a FIN signifies that the sender has no more
data to send. In order to allow subflows to operate independently,
however, and with as little change from regular TCP as possible, 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
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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 the DATA_FIN. This option, shown in Figure 10, is
attached to a regular FIN option on a subflow.
A DATA_FIN is an indication that the sender has no more data to send,
and as such can be used as a rapid indication of the end of data from
a sender. A DATA_FIN, as with the FIN on a regular TCP connection,
is a unidirectional signal.
A DATA_FIN occupies one octet (the final octet) of Data Sequence
Number space. This number is included in the option, and will be
ACKed at data level to ensure reliable delivery.
The DATA_FIN is an optimisation to rapidly indicate the end of a data
stream and clean up state associated with a MPTCP connection,
especially when some subflows may have failed. Specifically, when a
DATA_FIN has been received, IF all data has been successfully
received, timeouts on all subflows MAY be reduced. Similarly, when
sending a DATA_FIN, once all data (including the DATA_FIN, since it
occupies one octet of data sequence space) has been acknowledged,
FINs must be sent on every subflow. This applies to both endpoints,
and is required in order to clean up state in middleboxes.
There are complex interactions, however, between a DATA_FIN and
subflow properties:
o A DATA_FIN MUST only be sent on a packet which also has the FIN
flag set.
o When DATA_FIN is sent, it should be sent on all subflows.
o There is a one-to-one mapping between the DATA_FIN and the
subflow's FIN flag (and its associated sequence space and thus its
acknowlegement). In other words, when a subflow's FIN flag has
been acknowledged, the associated DATA_FIN is also acknowledged.
o The DATA_ACK (Section 4.4.2), which will be included with a
DATA_FIN, is used to verify that all data has been successfully
received.
It should be noted that an endpoint 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,
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that is equivalent to having closed the connection with a DATA_FIN.
The full eight-byte data sequence number is always included in a
DATA_FIN.
1
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=OPT_DFIN | Length=10 | Data Sequence Number (8B) :
+---------------+---------------+---------------+--------------+
: Data Sequence Number (contd.) :
+---------------+---------------+---------------+--------------+
: Data Sequence Number (contd.)|
+---------------+---------------+
Figure 10: DATA_FIN option
4.6. Error Handling
TBD
Unknown token in MPTCP SYN should equate to an unknown port, e.g. a
TCP reset? We should make this as silent and tolerant as possible.
Where possible, we should keep this close to the semantics of TCP.
However, some MPTCP-specific issues such as where a data sequence
number is missing from a subflow, will definitely need MPTCP-specific
errors handling in those cases.
5. Security Considerations
TBD
(Token generation, handshake mechanisms, new subflow authentication,
etc...)
A generic threat analysis for the addition of multipath capabilities
to TCP is presented in [8]. The protocol presented here has been
designed to minimise or eliminate these identified threats. (A
future version of this document will explicitly address the presented
threats).
The development of a TCP extension such as this will bring with it
many additional security concerns. We have set out here to produce a
solution that is "no worse" than current TCP, with the possibility
that more secure extensions could be proposed later.
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The primary area of concern will be around the handshake to start new
subflows which join existing connections. The proposal set out in
Section 4.1 and Section 4.2 is for the initiator of the new subflow
to include the token of the other endpoint in the handshake. The
purpose of this is to indicate that the sender of this token was the
same entity that received this token at the initial handshake.
One area of concern is that the token could be simply brute-forced.
The token must be hard to guess, and as such could be randomly
generated. This may still not be strong enough, however, and so the
use of 64 bits for the token would alleviate this somewhat.
The two tokens don't need to be the same length. Token B could be 64
bits and token A 32 bits. If JOIN always contains Token B, this
would provide adequate security while saving scarce space in the
initial SYN, where it is most at a premium.
Use of these tokens only provide an indication that the token is the
same as at the initial handshake, and does not say anything about the
current sender of the token. Therefore, another approach would be to
bring a new measure of freshness in to the handshake, so instead of
using the initial token a sender could request a new token from the
receiver to use in the next handshake. Hash chains could also be
used for this purpose.
Yet another alternative would be for all SYN packets to include a
data sequence number. This could either be used as a passive
identifier to indicate an awareness of the current data sequence
number (although a reasonable window would have to be allowed for
delays). Or, the SYN could form part of the data sequence space -
but this would cause issues in the event of lost SYNs (if a new
subflow is never established), thus causing unnecessary delays for
retransmissions.
6. Interactions with Middleboxes
Multipath TCP will be deployed in a network that no longer provides
just basic datagram delivery. A miriad of middleboxes are deployed
to optimize various perceived problems with the Internet protocols:
NATs primarily address space shortage [9], Performance Enhancing
Proxies (PEPs) optimize TCP for different link characteristics [10],
firewalls [11] and intrusion detection systems try to block malicious
content from reaching a host, and traffic normalizers [12] ensure a
consistent view of the traffic stream to IDSes and hosts.
All these middleboxes optimize current applications at the expense of
future applications. In effect, future applications must mimic
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existing ones if they want to be deployed. Further, the precise
behaviour of all these middleboxes is not clearly specified, and
implementation errors make matters worse, raising the bar for the
deployment of new technologies.
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 new TCP options. 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 SYN packets contain the MPC option to indicate the use of
MPTCP. When the middlebox drops the packet containing the MPC option
either on the outgoing or the return path, the connection will fail.
Host A SHOULD fall back to TCP in such cases (studies suggest that
few middleboxes drop packets with unknown options). The same applies
for subflow setup.
The second case is when the middleboxes strip options. Let's first
discuss behaviour for initial connection SYNs (see Figure 11). If
the option is stripped from the packet on the outgoing path, the
connection falls back to regular TCP. If the option is stripped on
the return path, host B will wait for a DATA_ACK of its connection
SYN, retransmitting the SYN/ACK until it declares the connection
failed. Host A thinks it is talking to a regular host, and may send
data segments, but these will not be acked by host B as they do not
have the proper mapping. Hence the connection fails. Host A SHOULD
fall back to regular TCP after the connection times out.
Subflow SYNs contain the OPT_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
connection fails). When host A receives the SYN/ACK it sends a RST
because the SYN/ACK does not contain the OPT_JOIN option and its
token. Either way, the connection fails.
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Host A Host B
| Middlebox M |
| | |
| SYN(OPT_MPC) | SYN |
|-------------------|---------------->|
| SYN/ACK |
|<------------------------------------|
a) OPT_MPC option stripped on outgoing path
Host A Host B
| SYN(OPT_MPC) |
|------------------------------------>|
| Middlebox M |
| | |
| SYN/ACK | SYN/ACK(OPT_MPC) |
|<----------------|-------------------|
b) OPT_MPC option stripped on return path
Figure 11: 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.
If options are stripped in either direction by middleboxes (this is
unlikely, as the SYN options did get through), the particular subflow
will timeout repeatedly while waiting for a DATA_ACK or subflow-level
ACK, and will be closed. If the subflow is the initial one, host A
SHOULD fall back to regular TCP.
We can further analyze what happens when a fraction of options is
stripped. The multipath subflow should survive losing a fraction of
DATA_ACKs and data sequence mappings, 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
or stripping options were discussed above, and are not included here:
o NAT: will prevent flow/subflow setup when the server does not have
a public address. MPTCP assumes the server has at least one
public address (or the client uses standard NAT traversal to reach
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it) that is used to setup the connection. If uses Add Address
messages to signal the existence of other addresses.
o Performance Enhancing Proxies: might pro-actively ACK data and
then fail. MPTCP 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: do not allow holes in sequence numbers, 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 Segmentation/Coalescing (e.g. tcp segmentation offloading, etc):
might copy options between packets and might strip some options.
MPTCP's data sequence mapping includes the 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 Firewalls: might perform sequence number randomization on outgoing
connections. MPTCP uses relative sequence numbers in data
sequence mapping to cope with this.
7. Interfaces
TBD
Interface with applications, interface with TCP, interface with lower
layers...
Discussion of interaction with applications (both in terms of how
MPTCP will affect an application's assumptions of the transport
layer, and what API extensions an application may wish to use with
MPTCP) are discussed in [5].
8. Open Issues
This specification is a work-in-progress, and as such there are many
issues that are still to be resolved. This section lists many of the
key open issues within this specification; these are discussed in
more detail in the appropriate sections throughout this document.
o Best handshake mechanisms (Section 4.1). This document contains a
proposed scheme by which connections and subflows can be set up.
It is felt that, although this is "no worse than regular TCP",
there could be opportunities for significant improvements in
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security that could be included (potentially optionally) within
this protocol.
o Issues around simultaneous opens, where both ends attempt to
create a new subflow simultaneously, need to be investigated and
behaviour specified.
o Appropriate mechanisms for controlling policy/priority of subflow
usage (specifically regarding controlling incoming traffic,
Section 4.4.6). The ECN signal is currently proposed but other
alternatives, including per subflow receive windows or options
indicating path properties, could be employed instead.
o How much control do we want over subflows from other subflows
(e.g. closing when interface has failed)? Do we want to
differentiate between subflows and addresses (Section 4.2)?
o Do we want a connection identifier in every packet? E.g. would it
make the implementation of an IDS easier?
o Should we do signaling in the TCP payload, rather than options as
proposed in this draft? We discuss this alternative in the
appendix.
o Should we explicitly support SYN cookies? With the current
design, MPTCP would be downgraded to basic TCP if SYN cookies were
used. Is it worth designing the protocol to allow stateless
server handshake?
o Instead of an Address ID in JOIN, are there any cases where a
Subflow ID (i.e. unique to the subflow) would be useful instead?
For example, two addresses which become NATted to the same
address?
9. 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 many members of the Trilogy project, notably Iljitsch
van Beijnum, Lars Eggert, Marcelo Bagnulo Braun, Robert Hancock, Pasi
Sarolahti, Olivier Bonaventure, Toby Moncaster, Philip Eardley,
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Andrew McDonald and Sergio Lembo.
10. IANA Considerations
This document will make a request to IANA to allocate new values for
TCP Option identifiers, as follows:
+------------+-----------------------+---------------+-------+
| Symbol | Name | Ref | Value |
+------------+-----------------------+---------------+-------+
| OPT_MPCAP | Multipath Capable | Section 4.1 | (tbc) |
| OPT_ADDR | Add Address | Section 4.3.1 | (tbc) |
| OPT_REMADR | Remove Address | Section 4.3.2 | (tbc) |
| OPT_JOIN | Join Connection | Section 4.2 | (tbc) |
| OPT_DSN | Data Sequence Mapping | Section 4.4 | (tbc) |
| OPT_DACK | DATA_ACK | Section 4.4 | (tbc) |
| OPT_DFIN | DATA_FIN | Section 4.5 | (tbc) |
+------------+-----------------------+---------------+-------+
Table 1: TCP Options for MPTCP
11. References
11.1. Normative References
[1] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
11.2. Informative References
[2] Postel, J., "Transmission Control Protocol", STD 7, RFC 793,
September 1981.
[3] Ford, A., Raiciu, C., Barre, S., and J. Iyengar, "Architectural
Guidelines for Multipath TCP Development",
draft-ietf-mptcp-architecture-00 (work in progress),
March 2010.
[4] Raiciu, C., Handley, M., and D. Wischik, "Coupled Multipath-
Aware Congestion Control", draft-raiciu-mptcp-congestion-00
(work in progress), October 2009.
[5] Scharf, M. and A. Ford, "MPTCP Application Interface
Considerations", draft-scharf-mptcp-api-00 (work in progress),
October 2009.
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[6] Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP
Selective Acknowledgment Options", RFC 2018, October 1996.
[7] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition of
Explicit Congestion Notification (ECN) to IP", RFC 3168,
September 2001.
[8] Bagnulo, M., "Threat Analysis for Multi-addressed/Multi-path
TCP", draft-ietf-mptcp-threat-00 (work in progress),
February 2010.
[9] Srisuresh, P. and K. Egevang, "Traditional IP Network Address
Translator (Traditional NAT)", RFC 3022, January 2001.
[10] Border, J., Kojo, M., Griner, J., Montenegro, G., and Z.
Shelby, "Performance Enhancing Proxies Intended to Mitigate
Link-Related Degradations", RFC 3135, June 2001.
[11] Freed, N., "Behavior of and Requirements for Internet
Firewalls", RFC 2979, October 2000.
[12] 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>.
[13] Eddy, W. and A. Langley, "Extending the Space Available for TCP
Options", draft-eddy-tcp-loo-04 (work in progress), July 2008.
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
both SYN, data packets 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, window scale, sack permitted and
timestamp options. Together these sum to 19B. The multipath capable
(MPC) option requires a max of 16B, and the Join option requires 8B,
so they both fit the existing space.
TCP data packets typically carry timestamp options in every packet,
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taking 10B. That leaves 30B which are enough to encode the data
sequence mapping (max 16B) and the DATA_ACK if the flow is
bidirectional (max 10B).
Pure ACKs in TCP typically contain only timestamps (10B). Here,
multipath TCP typically needs to encode the DATA_ACK (max 10B).
Ocasionally acks will contain SACK information. Depending on the
number of lost packets, SACK may utilize the entire option space. We
propose reducing the number of SACK blocks by one to accomodate the
DATA_ACK.
Encoding Add/Remove address options uses at most 10B (for IPv6
addresses). These will fit in data packets if the DATA_ACK is not
present. Otherwise, the endpoint can insert pure ACKs that contain
the add address option. Finally, if SACK information is included in
the data packets, one further block can be removed to accomodate the
add address option.
All the new options fit in the space available yet there is little
room for adding new options. We note that if 8B data sequence
numbers are used, PAWS is no longer needed. Hence the use for
timestamps is limited to providing RTT measurements for retransmitted
packets. As loss rates are typically low, we believe we can just
stop using timestamps, claiming 10B of options space on all packets.
Alternatively, we could use a TCP option to increase the option
space, such as that proposed in [13]. The proposal extends the 4 bit
header to 16 bits. Such an option could only be used between nodes
that support it, however, and so long options could not be used until
a handshake is complete.
Finally, there are issues with options reliability. 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/remove address options. Here we favour redundant
transmissions at the sender (whether on multiple paths, or on the
same path on a number of ACKs). The cases where options are stripped
by middleboxes are discussed in Section 6.
Appendix B. Signaling Control Information in the Payload
Appendix C. Resync Packet
In earlier versions of this draft, we proposed the use of a "re-sync"
option that would be used in certain circumstances when a sender
needs to instruct the receiver to skip over certain subflow sequence
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numbers (i.e. to treat the specified sequence space as having been
received and acknowledged).
The typical use of this option will be when packets are retransmitted
on different subflows, after failing to be acknowledged on the
original subflow. In such a case, it becomes necessary to move
forward the original subflow's sequence numbering so as not to later
transmit different data with a previously used sequence number (i.e.
when more data comes to be transmitted on the original subflow, it
would be different data, and so must not be sent with previously-used
(but unacknowledged) sequence numbering).
The rationale for needing to do this is two-fold: firstly, when ACKs
are received they are for the subflow only, and the sender infers
from this the data that was sent - if the same sequence space could
be occupied by different data, the sender won't know whether the
intended data was received. Secondly, certain classes of middleboxes
may cache data and not send the new data on a previously-seen
sequence number.
This option was dropped, however, since some middleboxes may get
confused when they meet a hole in the sequence space, and do not
understand the resync option. It is therefore felt that the same
data must continue to be retransmitted on a subflow even if it is
already received after being retransmitted on another. There should
not be a significant performance hit from this since the amount of
data involved and needing to be retransmitted multiple times will be
relatively small.
Therefore, it is necessary to 're-sync' the expected sequence
numbering at the receiving end of a subflow, using the following TCP
option. This packet declares a sequence number space (inclusive)
which the receiving node should skip over, i.e. if the receiver's
next expected sequence number was previously within the range
start_seq_num to end_seq_num, move it forward to end_seq_num + 1.
This option will be used on the first new packet on the subflow that
needs its sequence numbering re-synchronised. It will be continue to
be included on every packet sent on this subflow until a packet
containing this option has been acknowledged (i.e. if subflow
acknowledgements exist for packets beyond the end sequence number).
If the end sequence number is earlier than the current expected
sequence number (i.e. if a resync packet has already been received),
this option should be ignored.
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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=OPT_RESYNC| Length = 10 | Start Sequence Number :
+---------------+---------------+------------------------------+
: (4 octets) | End Sequence Number :
+---------------+---------------+------------------------------+
: (4 octets) |
+-------------------------------+
Figure 12: Resync option
Appendix D. Changelog
This section maintains logs of significant changes made to this
document between versions.
D.1. Changes since draft-ford-mptcp-multiaddressed-02
o Remote Version and Address ID in MP_CAPABLE in Section 4.1, 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 OPT_JOIN in Section 4.2.
o Added DATA_ACK to Section 4.4.
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
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Costin Raiciu
University College London
Gower Street
London WC1E 6BT
UK
Email: c.raiciu@cs.ucl.ac.uk
Mark Handley
University College London
Gower Street
London WC1E 6BT
UK
Email: m.handley@cs.ucl.ac.uk
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