Internet Engineering Task Force A. Ford
Internet-Draft Roke Manor Research
Intended status: Experimental C. Raiciu
Expires: January 11, 2010 M. Handley
University College London
S. Barre
Universite catholique de
Louvain
July 10, 2009
TCP Extensions for Multipath Operation with Multiple Addresses
draft-ford-mptcp-multiaddressed-01
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Abstract
Often endpoints are connected by multiple paths, but the nature of
TCP/IP restricts communications to a single path per socket.
Resource usage within the network would be more efficient were these
multiple paths able to be used concurrently. This should enhance
user experience through higher throughput and improved resilience to
network failure. This document presents extensions to TCP in order
to transparently provide this multi-path functionality at the
transport layer, if at least one endpoint is multi-addressed.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. Motivation . . . . . . . . . . . . . . . . . . . . . . . . 4
1.2. Design Assumptions . . . . . . . . . . . . . . . . . . . . 4
1.3. Layered Representation . . . . . . . . . . . . . . . . . . 5
1.4. Operation Summary . . . . . . . . . . . . . . . . . . . . 6
1.5. Open Issues . . . . . . . . . . . . . . . . . . . . . . . 7
1.6. Requirements Language . . . . . . . . . . . . . . . . . . 8
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 8
3. Semantic Issues . . . . . . . . . . . . . . . . . . . . . . . 8
4. MPTCP Protocol . . . . . . . . . . . . . . . . . . . . . . . . 9
4.1. Session Initiation . . . . . . . . . . . . . . . . . . . . 9
4.2. Starting a New Subflow . . . . . . . . . . . . . . . . . . 11
4.3. Address Knowledge Exchange (Path Management) . . . . . . . 12
4.3.1. Adding Addresses . . . . . . . . . . . . . . . . . . . 13
4.3.2. Remove Address . . . . . . . . . . . . . . . . . . . . 14
4.4. General MPTCP Operation . . . . . . . . . . . . . . . . . 15
4.4.1. Receive Window Considerations . . . . . . . . . . . . 16
4.4.2. Congestion Control Considerations . . . . . . . . . . 17
4.4.3. Subflow Policy . . . . . . . . . . . . . . . . . . . . 17
4.4.4. Retransmissions . . . . . . . . . . . . . . . . . . . 18
4.5. Closing a Connection . . . . . . . . . . . . . . . . . . . 19
4.6. Error Handling . . . . . . . . . . . . . . . . . . . . . . 20
5. Congestion Control Coupling for MPTCP . . . . . . . . . . . . 20
6. Security Considerations . . . . . . . . . . . . . . . . . . . 21
7. Interactions with Middleboxes . . . . . . . . . . . . . . . . 22
8. Interfaces . . . . . . . . . . . . . . . . . . . . . . . . . . 22
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 23
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 23
11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 23
11.1. Normative References . . . . . . . . . . . . . . . . . . . 23
11.2. Informative References . . . . . . . . . . . . . . . . . . 24
Appendix A. Functional Separation . . . . . . . . . . . . . . . . 24
A.1. Motivations . . . . . . . . . . . . . . . . . . . . . . . 24
A.2. TCP Performance . . . . . . . . . . . . . . . . . . . . . 25
A.3. Architecture overview . . . . . . . . . . . . . . . . . . 25
A.4. PM/MPS interface . . . . . . . . . . . . . . . . . . . . . 27
Appendix B. Notes on use of TCP Options . . . . . . . . . . . . . 28
Appendix C. Resync Packet . . . . . . . . . . . . . . . . . . . . 28
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 30
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1. Introduction
Multipath TCP is set of extensions for regular TCP [RFC0793] to allow
one TCP connection to be spread across multiple paths. This section
describes the motivation behind the design of Multipath TCP
(henceforth referred to as MPTCP), and gives a summary of its
operation. The following sections describe in greater detail the
proposed extensions and the operation of the resulting protocol.
1.1. Motivation
As the Internet evolves, demands on Internet resources are ever-
increasing, but often these resources (in particular, bandwidth)
cannot be fully utilised due to protocol constrains on both the end-
systems and within the network. By the application of resource
pooling [WISCHIK], these resources can be 'pooled' such that they
appear as a single logical resource to the user. Multipath TCP
achieves resource pooling by combining multiple TCP sessions running
over multiple paths, and presenting them as a single TCP connection
to the application.
This form of resource pooling bring two key benefits:
o To increase the resilience of the connectivity by providing
multiple paths, protecting end hosts from the failure of one.
o To increase the efficiency of the resource usage, and thus
increase the network capacity available to end hosts.
The protocol presented in this document follows the same service
model as TCP [RFC0793]: byte oriented, in order reliable delivery.
To have a deployable protocol, we impose the following "do no harm"
philosophy: multipath TCP should behave no worse (throughput wise)
than running a single TCP connection over any of its paths, and using
multiple paths should not harm users using single path TCP at shared
bottlenecks.
1.2. 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
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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.
There are three aspects to the backwards-compatibility listed above:
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 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 packet marking or
partial multipath.
o Some features of the design presented here could be combined with
mechanisms such as shim6 [I-D.ietf-shim6-proto].
This draft also suggests a safe way to couple congestion controllers
in a way that achieves the "do no harm philosophy". This is for
completeness or our arguments: we expect this description to evolve
into a companion new internet draft.
1.3. 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
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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 architecture.
+-------------------------------+
| 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
Appendix A.
1.4. Operation Summary
This section provides a high-level summary of normal operation in
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. 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.
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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
initiate new subflows by using its additional addresses, or by
signalling to the other endpoint its available addresses.
o The exact properties of these TCP sessions that are logically
bonded are dependent upon the congestion and flow control
characteristics of the endpoints' MPTCP implementation.
o MPTCP adds connection-level sequence numbers in order to
reassemble the data stream in-order from multiple subflows.
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
1.5. 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.
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o Best handshake mechanisms. 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 security
that could be included (potentially optionally) within this
protocol.
o Issues around simulataneous opens, where both ends attempt to
create a new subflow simultaneously, need to be investigated and
behaviour specified.
o Appropriate mechanisms for controlling policy of subflow usage.
The ECN signal is currently proposed but other alternatives,
including path property options, could be employed instead.
1.6. 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 [RFC2119].
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.
Token: A 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.
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Sequence Number: The TCP sequence number is subflow-specific, with a
data sequence number used for reassembly at connection-level.
FIN: The FIN only applies to a subflow, not to a connection. For a
connection-level FIN, use the DATA FIN option.
ACK: The ACK acknowledges the subflow sequence number only, and the
mapping to the data sequence number is handled out-of-band.
RST: The RST only applies to a subflow. There is no connection-
level RST, since it would be impossible to distinguish the two, as
the link between a subflow and a connection is established at the
SYN handshake. A connection is considered reset if every subflow
sends a RST in response.
Length: There is additionally an explicit length for each MPTCP
segment in order to separate potential TCP/IP-layer segmentation
from the MPTCP data flow.
Address List: The address management is handled per-connection to
permit the application of per-connection local policy.
5-tuple: The 5-tuple (protocol,local IP, local port, remote IP,
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.
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
discussed in Section 10, and are defined throughout the following
subsections.
4.1. Session Initiation
Session Initiation begins with a SYN, SYN/ACK exchange on a single
path. Each of these packets will additionally feature the Multipath
Capable TCP option (Figure 3, which declares the sender's locally
unique 32-bit token for this connection, and a version field.
The "Multipath Capable" option declares an endpoint to be capable of
operating Multipath TCP (or rather, more accurately, a desire to
operate Multipath TCP on this particular connection). As well as
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this declaration, this field presents a token, which is used when
adding additional subflows to this connection.
This token is generated by the sender and has local meaning only, but
therefore 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 6, including the possibility of a 64-bit
token and an initial data sequence number.)
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 techniques 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=OPT_MPC | Length = 7 |(resvd)|Version| Sender Token :
+---------------+---------------+-------------------------------+
: Sender Token (continued - 4 octets total) |
+-----------------------------------------------+
Figure 3: Multipath Capable option
The version field represents the version of MPTCP in use. The
version provided in this specification is 0. The reserved bits may
be used for connection-specific flags in later versions.
If a SYN contains a "multipath capable" option but the SYN/ACK does
not, it is assumed that the recipient is not multipath capable and
thus the MPTCP session will operate as regular, single-path TCP. If
a SYN does not contain a "multipath capable" option, the SYN/ACK MUST
NOT contain one in response.
If these 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 Multipath 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. In the case
of out-of-order packets, i.e. if a multipath-capable SYN/ACK is
received in response to a multipath-capable SYN, after a standard SYN
has been sent, then once again it is up to the sender to choose how
to behave. For example, the sender could respond to new connections
using the previously declared token, or it could simply drop any new
multipath options within the flow.
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If an endpoint is known to be multiaddressed (e.g. through multiple
addresses returned in a DNS lookup), alternative destination
addresses should be tried first, before falling back to regular TCP.
In addition to this option, a Data Sequence Number option (discussed
in Section 4.4) is included to provide an initial data-level sequence
number (and this initial SYN counts as one octet in this space, as
for a regular SYN in single-path TCP).
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 a path management
technique as described in Section 4.3. Using this knowledge, an
endpoint will initiate a new subflow over a currently unused pair of
addresses.
A new subflow is started as a normal TCP SYN/ACK exchange. The
following TCP option is used to identify 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 Multipath
Capable option received in the first SYN/ACK exchange.
It should be noted that, in theory, additional subflows can exist
between any pair of ports, and as such it is this token that is used
for demuxing at the receiver. A receiver must store some mapping
state, of (source_addr, dest_addr, source_port, dest_port) to its
token, using information from the initial SYN exchange, in order to
enable this. In practice, however, it is envisaged that most new
subflows will connect to a port that is already in use as the source
or destination port of an existing subflow, in order to have a
greater chance of getting through firewalls and other middleboxes,
and to support traffic engineering of the flows.
This option includes an "Address ID". This is an identifier, locally
unique to the sender of this option, that identifies the source
address of this packet. This serves two purposes. Firstly, 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 NATs). Secondly, it allows correlation between
new connection attempts and address signalling (Section 4.3.1), to
prevent duplicate subflow initiation.
This option can only be present when the SYN flag is set.
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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_JOIN | Length = 6 |Receiver Token (4 octets total):
+---------------+---------------+----------------+--------------+
: Receiver Token (continued) | Address ID |
+-------------------------------+----------------+
Figure 4: Join Connection option
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 architectrual thinking behind this design, see the
discussion of functional separation in Appendix A.
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. The second is described in the
following subsections, whereby addresses are signalled explicitly to
the endpoint to allow it to initiate new connections. This approach
has been chosen so as to allow addresses to change in flight, and
thus the use of NATs, whilst also allowing the signalling of
previously unknown addresses, such as those belonging to other
address families.
In more detail, an example of the typical operation is as follows,
where an existing address is used at one endpoint:
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 the
source address and port, or a multihomed source may open a new
subflow from its new address another connection to the existing
destination and port).
o To expand upon this, say a connection is intiated from host "A" on
(address, port) combination A1 to desintation (address, port) B1
on host "B". If host A is multihomed, it starts an additional
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.
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o Simultaneously (or near-simultaneously), 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 a
Add Address packet sent from A1 to B1, informing B of address A2.
o If host B successfully receives the first SYN, starting a new
subflow, it can use the Address ID to correlate this with the Add
Address option that will also arrive on an existing subflow, and
it will respond to the SYN with a SYN/ACK. Otherwise, if it does
not receive such a SYN, it tries to initiate a new subflow from
one or more of its addresses to address A2. This is intended to
permit new sessions to be opened if one endpoint is behind a NAT.
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. Adding Addresses
Announcing additional addresses that an endpoint can be reached on
will be undertaken by the Add Address TCP Option (Figure 5), where an
(ID, address) pair can be announced to the other endpoint. Several
addresses can be added if there is sufficient TCP option space,
otherwise multiple TCP messages containing this option must be sent.
This option can be used at any time during a connection.
The Add Address option announces a list of alternative IP addresses,
beyond the current one in use, that the sender can be contacted on.
This option can be used multiple times until all available addresses
have been announced, in order to get around TCP option space limits.
It should be noted that 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 use. The ID must be unique to the
sender, and although it may be a sequential counter, this is not
mandated.
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).
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
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could be used in later versions of this protocol, 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 5: Add Address option (for IPv4)
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 6), 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.
If there is no address at the requested ID, the receiver will
silently ignore the request.
Address removal is undertaken by ID, so as to permit the use of NATs
and other middleboxes, in the cases where new connections have been
initiated but now want to be removed.
The closure of a single subflow, rather than all using a particular
address, is undertaken as normal with a FIN exchange on the subflow -
for more information, see Section 4.5.
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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_REMADR| Length = 2+n | Address ID | ...
+---------------+---------------+---------------+
Figure 6: Remove Address option
4.4. General MPTCP Operation
This section discusses operation of MPTCP for data transfer,
independent of the path management mechanism used.
At a high level, the an MPTCP implementation will take one input data
stream from an application, and split it into one or more subflows.
The data stream as a whole can be reassembled through the use of the
Data Sequence Number (Figure 7) option, which defines the sequence in
the data stream of the first octet of the packet's payload, and this
is used by the receiver to ensure in-order delivery to th application
layers. Meanwhile, the subflow-level sequence numbers (i.e. the
regular sequence numbers in the TCP header) have subflow-only
relevance.
The only acknowledgements are those at the subflow-level, so the
sender must be able to map these acknowledgements to the data
sequence numbers that were contained in the relevant packets. The
sender thus knows, if subflow data goes unackowledged, which part of
the original data stream this equates to, and thus what data must be
retransmitted. It is expected (but not mandated) that SACK [RFC2018]
is used as an efficiency at the subflow level. Each subflow will
maintain its own congestion widow.
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-4) octets ) | Data-level Length (2 octets) |
+-------------------------------+------------------------------+
Figure 7: Data Sequence Number option
In addition to the Data Sequence Number, this option also includes a
Data-level Length field. The purpose of this field is to assist with
compatibility with situations where TCP/IP segmentation is undertaken
separately from the stack that is generating the data flow (e.g.
through the use of TCP segementation offloading on network interface
cards, or by middleboxes). This field declares what length of data
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this data sequence number is valid for, allowing a receiver to infer
when it has received sufficient segments. The primary motivation for
this behaviour is the understanding that devices involved in re-
segmentation typically repeat additional TCP options into every re-
segmented packet. The use of this length field will make it clear
when all relevant segments have been received. (It is FFS whether
this is the optimal solution to this issue.)
As a TCP option contains a length field, the length of the Data
Sequence Number can be declared implicitly. Although it is expected
that initial implementations will use 32-bit sequence numbers (i.e. 4
octets, so a length field of 8), setting the length field to 12 and
including a 64-bit sequence number (of four octets) MUST be
considered valid and processed appropriately. This may have also
have useful security implications, discussed in Section 6.
As wth 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 Number
option along with the Multipath Capable option in the initial SYN
(which occupies one octet of data sequence space; see Section 4.1).
The Data Sequence Number is included in every MPTCP packet that
contains data (or a DATA FIN, see Section 4.5), even if only one path
is in use, so long as the MPTCP handshake has been completed and the
endpoints have therefore agreed to use MPTCP.
The MPTCP data and subflow level sequence numbering could be seen to
be analogous to that used in SACK, however there are subtle
differences. The key similarity is that it is possible to have
temporary "holes" in the received data sequence space - later data
may have arrived earlier (most likely on a different subflow), but
does not need to be retransmitted. The "holes" are later filled in.
The key difference, however, is that while SACK can rely on the
regular TCP cumulative acknowledgements to indicate how much data has
been successfully received (with no holes), there is no similar
method in MPTCP. Instead, the sender must keep track of the
acknowledgements to derive what data has been successfully received.
This leads to some oddities especially with session termination (see
Section 4.5).
4.4.1. Receive Window Considerations
Normal 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.
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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.
4.4.2. Congestion Control Considerations
Different subflows in an MPTCP connection have different congestion
windows. To achieve resource pooling [WISCHIK], 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 Section 5; 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.
4.4.3. 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,
all available paths will be used simultaneously for data transfer.
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 delay or bandwidth, in cases
where the additional paths are significantly worse and not worth
including in the base operation. Other metrics such as this could be
wrapped into an overall "cost" metric for a link.
The ability to make effective choices at the sender requires full
knowledge of the path cost, which is unlikely to be the case. There
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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, such as in the
case of laptop computers with wired and wireless connectivity.
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, this could be achieved by the receiver not
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 is brutal,
naive, and will result in unnecessary retransmissions.
Therefore, it is proposed to use ECN [RFC3168] 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.
4.4.4. Retransmissions
This protocol specification does not mandate any mechanisms for
handling retransmissions in the event of path failures, and much will
be dependent upon local policy (as discussed in Section 4.4.3). The
data sequence number, as given in a TCP option, is used to reassemble
the incoming streams before presentation to the application layers,
so a sender is free 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 we feel this is the best behaviour.
Optimisations could be negotiated in future versions of this
protocol.
Of course, retransmissions on alternative subflows will only occur if
this is what local policy suggests. Indeed, it may be equally valid
to retransmit on the same subflow if alternative paths have
considerably worse quality of service, or are only kept for backup
purposes. Additionally, it may be possible for some implementations
to signal from lower layers if there are problems with the paths, and
so more appropriate responses could occur.
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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 will only affect 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 the DATA FIN. This option, shown in Figure 8, 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. Therefore, it is an optimisation to 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 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 can only be sent on a packet which also has the FIN
flag set.
o A DATA FIN occupies one octet (the final octet) of Data Sequence
Number space. Therefore, even if there is no user data, a Data
Sequence Number option must be added to a packet containing the
DATA FIN option. This allows the receiver to easily determine the
last data sequence number that should have been received.
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 As such, the acknowledgement of a FIN and DATA FIN DOES NOT
indicate that all data has been successfully received. Because
the data level ack is inferred from subflow acks, the endpoint can
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tell when all data before the DATA FIN has been 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,
that is equivalent to having closed the connection with a DATA FIN.
1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+---------------+---------------+
| Kind=OPT_DFIN | Length = 2 |
+---------------+---------------+
Figure 8: 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.
The amount of error handling required may also have an impact on the
choice of path management schemes. Issues may include odd cases
where a data sequence number is missing from a subflow. Will
definitely need errors in those cases.
5. Congestion Control Coupling for MPTCP
Coupling congestion windows can achieve resource pooling, by pushing
traffic to underutilized areas of the network. Another effect of
coupling is fairness at bottleneck: when two MPTCP flows share a
common bottleneck, their combined throughput should not be more than
that of a single TCP flow.
To achieve perfect resource pooling, one must couple both increase
and decrease of congestion windows across subflows. Yet this tends
to exhibit flappyness: when the two paths have similar levels of
congestion, the controller will tend to allocate all the window to
one or the other subflows, and perform random flips between the two
equilibrium points. This seems not desirable in general, and is
particularly bad when the achieved rates depend on the RTT (as in the
current Internet).
By only coupling increases we remove flappyness but also reduce the
extent of resource pooling the protocol achieves. We now succintly
describe our protocol, assuming there are only two subflows (the
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general case is easy to derive, but is more difficult to understand).
Let v_1 and v_2 be the congestion windows on the two subflows, and
assume there is always data to send. Let w = v_1 + v_2. Let p_i,
rtt_i be the drop probability and round trip time on path i.
Our proposed algorithm is as follows:
o Increase v_i by a/w for each ack received on subflow i.
o Decrease v_i by v_i/2 for each drop on subflow i.
"a" is a parameter of the algorithm, and we'll describe next how to
pick proper values for it.
This algorithm will allocate window to the two subflows such that p1
* v1 = p2 * v2. Thus, when the drop probabilities are equals, each
subflow gets an equal window; when they are different, more and more
window will be allocated to the flow with the lower drop probability.
The total throughput of the algorithm depends on the drop
probabilities and rtts of the two paths. We require that the total
throughput is no worse than the throughput a single TCP would get on
the fastest path. If we kept a constant regardless of path
properties, this requirement would be violated. However, if we
increase a according to the difference in drop probabilities and
rtts, it is always possible to match the throughput of the best path.
The second requirement is that none of the subflows should be, on
their own, more aggressive than a single TCP on the same path.
Increasing "a" indefinitely as required above, may create fairness
issues in some scenarios. In such cases, the "a" parameter is capped
on the paths where the increase is too aggressive, and some traffic
is pushed on the other paths.
It is possible to achieve all this behavior (adjusting and capping a)
by only using estimates of the rtts and the current windows for the
two subflows; explicit estimates of the drop probabilities are not
needed.
A full description of the congestion control algorithm is beyond the
scope of this document. The algorithm will be thoroughly described
in a companion document, soon to be released.
6. Security Considerations
TBD
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(Token generation, handshake mechanisms, new subflow authentication,
etc...)
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.
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 behard 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.
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.
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.
7. Interactions with Middleboxes
TBD
How we get around NATs, firewalls. Problems with TCP proxies. How
to make an MPTCP-aware middlebox, ...
8. Interfaces
TBD
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Interface with applications, interface with TCP, interface with lower
layers...
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 and
Andrew McDonald.
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_MPC | 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 Number | Section 4.4 | (tbc) |
| OPT_DFIN | DATA FIN | Section 4.5 | (tbc) |
+------------+----------------------+---------------+-------+
Table 1: TCP Options for MPTCP
11. References
11.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
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11.2. Informative References
[I-D.eddy-tcp-loo]
Eddy, W. and A. Langley, "Extending the Space Available
for TCP Options", draft-eddy-tcp-loo-04 (work in
progress), July 2008.
[I-D.ietf-shim6-proto]
Nordmark, E. and M. Bagnulo, "Shim6: Level 3 Multihoming
Shim Protocol for IPv6", draft-ietf-shim6-proto-12 (work
in progress), February 2009.
[I-D.van-beijnum-1e-mp-tcp-00]
van Beijnum, I., "One-ended Multipath TCP",
draft-van-beijnum-1e-mp-tcp-00 (work in progress),
May 2009.
[RFC0793] Postel, J., "Transmission Control Protocol", STD 7,
RFC 793, September 1981.
[RFC2018] Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP
Selective Acknowledgment Options", RFC 2018, October 1996.
[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP",
RFC 3168, September 2001.
[WISCHIK] Wischik, D., Handley, M., and M. Bagnulo Braun, "The
Resource Pooling Principle", ACM SIGCOMM CCR vol. 38 num.
5, pp. 47-52, October 2008,
<http://ccr.sigcomm.org/online/files/p47-handleyA4.pdf>.
Appendix A. Functional Separation
[Potential to move to separate architectural document]
This section describes the functional separation that drives the
design of the MPTCP protocol. Its main goal is to separate MPTCP in
two parts that communicate through a well defined interface. We
first provide the motivations for this functional separation, then we
describe in more details the two main components of the MPTCP
architecture.
A.1. Motivations
The major goal behind MPTCP is to send data over different paths in
the same time. This assumes that an MPTCP implementation must be
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able to discover and use the multiple paths that connect two given
hosts, when they exist. However, different mechanisms can be
envisioned for multipath discovery and use. Examples are as follows:
Use multiple addresses: This is the method currently proposed in
this document - if hosts are multi-addressed, different address
pairs may take different routes.
Use a path selector value: An end-host might be able to tag packets
with a path selector value, or "colour". If some network nodes
are able to read the colour and use it as a path selector, the
host can influence the outgoing path of the packet.
Next-hop selection: In a network configuration where multiple next-
hops can offer to forward packets, a host may decide to send some
of its packets through one next-hop, and some through another.
The above list is not exhaustive, and could grow as new network
technologies are deployed.
A.2. TCP Performance
In addition to purely sending data over multiple paths, MTCP must do
it in a way that will not affect TCP performance. This raises the
need for an efficient multipath congestion control algorithm. While
this specification does not mandate the use of any particular
algorithm for congestion control, it ensures that the protocol is
designed in such a way that any CC algorithm can be designed,
independently of the particular path management mechanism available
to the host. Consequently our architecture for MTCP decouples the
policy from the mechanism. The policy is the decision of what path
to use for each packet to send. It is mainly driven by the
implementation-dependent congestion control algorithm. The mechanism
is the technology used to ensure that a packet will be sent on the
desired path. This separation is intended to be relatively future-
proof by allowing these components to evolve at different speeds.
A.3. Architecture overview
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Control plane <-- | --> Data plane
+---------------------------------------------------------------+
| Multipath Scheduler (MPS) |
+---------------------------------------------------------------+
^ | |
| | |
|Announcing new | +-------------+
|paths. (referred | | Data packet |<--Path idx:3
|to as path indices) | +-------------+ attached
| | | by MPS
| | V
+--------------------------------------------\------------------+
| Path Manager (PM) \__________zzzzz |
+--------------------------------------------------------\------+
/ \ | \
/---------------------\ | /"\ /"\ /"\
| Path key Action | | | | | | | |
| 1 xxxxx | | | | | | | |
| 2 yyyyy | | \./ \./ \./
| 3 zzzzz | | path1 path2 path3
+---------------------+
Figure 9: Overview of MTCP architecture
A general overview of the architecture is provided in Figure 9. The
Multipath Scheduler (MPS) learns about the number of available paths
through notifications received from the Path Manager (PM). From the
point of view of the Multipath Scheduler, a path is just a number,
called a Path Index. Notifications from the PM to the MPS MAY
contain supporting information about the paths, if relevant, so that
the MPS can make more intelligent decisions about where to route
traffic. When the Multipath Scheduler initiates a communication to a
new host, it can only send the packets to the default path. But
since the Path manager is layered below the MPS, it can detect that a
new communication is happening, and tell the MPS about the other
paths it knows about.
From then on, it is possible for the MPS to attach a Path Index to
the control structure of its packets (internal to the MTCP
implementation), so that the Path Manager can map this Path Index to
the corresponding action. (see table in the lower left part of
Figure 9). The particular action depends on the network mechanism
used to select a path. Examples are address rewriting, tunnelling or
setting a path selector valude inside the packet.
The applicability of the architecture is not limited to the MTCP
protocol. While we define in this document an MTCP MPS (MTCP
Multipath Scheduler), other Multipath Schedulers can be defined. For
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example, if an appropriate socket interface is designed, applications
could behave as a Multipath Scheduler and decide where to send any
particular data. In this document we concentrate on the MTCP case,
however.
In this specification, we define the core protocol for Multipath TCP.
The core protocol is not dependent on the Path Management technique
that is chosen, and MUST be implemented in any MTCP MPS. We also
provide a default Path Manager that is based on declaring IP
addresses, and carries control information in TCP options. An
implementation of Multipath TCP can use any Path Manager, but it MUST
be able to fallback to the default PM in case the other end does not
support the custom PM. Alternative Path Managers may be specified in
separate documents in the future.
A.4. PM/MPS interface
The minimal set of requirement for a Path Manager is as follows:
o Outgoing untagged packets: Any outgoing packet flowing through the
Path Manager is either tagged or untagged (by the MPS) with a path
index. If it is untagged, the packet is sent normally to the
Internet, as if no multi-path support were present. Untagged
packets can be used to trigger a path discovery procedure, that
is, a Path Manager can listen to untagged packets and decide at
some time to find if any other path than the default one is
useable for the corresponding host pair. Note that any other
criteria could be used to decide when to start discovering
available paths. Note also that MPS scheduling will not be
possible until the Path Manager has notified the available paths.
The PM is thus the first entity coming into action.
o Outgoing tagged packets: The Path Manager maintains a table
mapping path indices to actions. The action is the operation that
allows using a particular path. Examples of possible actions are
route selection, interface selection or packet transformation.
When the PM sees a packet tagged with a path index, it looks up
its table to find the appropriate action for that packet. The tag
is purely local. It is removed before the packet is transmitted.
o Incoming packets: A Path Manager MUST ensure that incoming path is
mapped unambiguously to exactly one outgoing path. Note that this
requirement implies that the same number of incoming/outgoing
paths must be established. Moreover, a PM MUST tag any incoming
path with the same Path Index as the one used for the
corresponding outgoing path. This is necessary for MTCP to know
what outgoing path in acknowledged by an incoming packet.
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o Module interface: A PM MUST be able to notify the MPS about the
number of available paths. Such notifications MUST contain the
path indices that are legal for use by the MPS. In case the PM
decides to stop providing service for one path, it MUST notify the
MPS about path deletion. Additionnaly, a PM MAY provide
complementary path information when available, such as link
quality or preference level.
Appendix B. 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.
As such, when doing address list manipulation, not all data may fit.
This can be mitigated in one of two ways:
o Using an option to extend the option space, such as that proposed
in [I-D.eddy-tcp-loo], which proposes an option providing a 16-bit
header length field. 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.
o Alternatively, since at least one IP address option field should
be able to fit per packet, address list manipulation can be
undertaken with one address per packet. One method could be to
wait for data to send, and then append one new address per packet.
This would seem reasonable if the TCP session begins rapidly, but
if it is required that the multipath session is ready before the
first data is to be sent, address list manipulation would be
required on empty data (signalling only) packets. Issues may
arise regarding acknowledged delivery of signalling versus data -
this is discussed in Section 3 below.
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
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
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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 10: Resync option
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
Mark Handley
University College London
Gower Street
London WC1E 6BT
UK
Email: m.handley@cs.ucl.ac.uk
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Sebastien Barre
Universite catholique de Louvain
Pl. Ste Barbe, 2
Louvain-la-Neuve 1348
Belgium
Phone: +32 10 47 91 03
Email: sebastien.barre@uclouvain.be
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