IP-TFS: Aggregation and Fragmentation Mode for ESP and its Use for IP Traffic Flow SecurityLabN Consulting, L.L.C.chopps@chopps.orgThis document describes a mechanism for aggregation and
fragmentation of IP packets when they are being encapsulated in ESP
payloads. This new payload type can be used for various purposes such
as decreasing encapsulation overhead for small IP packets; however,
the focus in this document is to enhance IPsec traffic flow security
(IP-TFS) by adding Traffic Flow Confidentiality (TFC) to encrypted IP
encapsulated traffic. TFC is provided by obscuring the size and
frequency of IP traffic using a fixed-sized, constant-send-rate IPsec
tunnel. The solution allows for congestion control as well as
non-constant send-rate usage.Traffic Analysis (, ) is the act of extracting
information about data being sent through a network. While directly
obscuring the data with encryption , fully, the patterns in
the message traffic may expose information due to variations in its
shape and timing (, ). Hiding the size and frequency
of traffic is referred to as Traffic Flow Confidentiality (TFC) per
. provides for TFC by allowing padding to be added to encrypted
IP packets and allowing for transmission of all-pad packets
(indicated using protocol 59). This method has the major limitation
that it can significantly under-utilize the available bandwidth.This document defines an aggregation and fragmentation (AGGFRAG) mode
for ESP, and its use for IP Traffic Flow Security (IP-TFS). This
solution provides for full TFC without the aforementioned bandwidth
limitation. This is accomplished by using a constant-send-rate IPsec
tunnel with fixed-sized encapsulating packets; however, these
fixed-sized packets can contain partial, whole or multiple IP packets
to maximize the bandwidth of the tunnel. A non-constant send-rate is
allowed, but the confidentiality properties of its use are outside
the scope of this document.For a comparison of the overhead of IP-TFS with the RFC4303
prescribed TFC solution see .Additionally, IP-TFS provides for operating fairly within congested
networks . This is important for when the IP-TFS user is not
in full control of the domain through which the IP-TFS tunnel path
flows.The mechanisms, such as the AGGFRAG mode, defined in this document
are generic with the intent of allowing for non-TFS uses, but such
uses are outside the scope of this document.The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in BCP
14 when, and only when, they appear in all capitals,
as shown here.This document assumes familiarity with IP security concepts including
TFC as described in .As mentioned in , AGGFRAG mode utilizes an IPsec tunnel
as its transport. For the purpose of IP-TFS, fixed-sized encapsulating
packets are sent at a constant rate on the AGGFRAG tunnel.The primary input to the tunnel algorithm is the requested bandwidth
to be used by the tunnel. Two values are then required to provide for
this bandwidth use, the fixed size of the encapsulating packets, and
rate at which to send them.The fixed packet size MAY either be specified manually or be
determined through other methods such as the Packetization Layer MTU
Discovery (PLMTUD) (, ) or Path MTU discovery (PMTUD)
(, ). PMTUD is known to have issues so PLMTUD is
considered the more robust option. For PLMTUD, congestion control
payloads can be used as in-band probes (see and ).Given the encapsulating packet size and the requested bandwidth to be
used, the corresponding packet send rate can be calculated. The
packet send rate is the requested bandwidth to be used divided by the
size of the encapsulating packet.The egress (receiving) side of the AGGFRAG tunnel MUST allow for and
expect the ingress (sending) side of the AGGFRAG tunnel to vary the
size and rate of sent encapsulating packets, unless constrained by
other policy.As previously mentioned, one issue with the TFC padding solution in
is the large amount of wasted bandwidth as only one IP
packet can be sent per encapsulating packet. In order to maximize
bandwidth, IP-TFS breaks this one-to-one association by introducing
an AGGFRAG mode for ESP.AGGFRAG mode aggregates as well as fragments the inner IP traffic
flow into encapsulating IPsec tunnel packets. For IP-TFS, the IPsec
encapsulating tunnel packets are a fixed size. Padding is only added
to the tunnel packets if there is no data available to be sent at
the time of tunnel packet transmission, or if fragmentation has been
disabled by the receiver.This is accomplished using a new Encapsulating Security Payload (ESP,
) Next Header field value AGGFRAG_PAYLOAD
().Other non-IP-TFS uses of this AGGFRAG mode have been suggested, such
as increased performance through packet aggregation, as well as
handling MTU issues using fragmentation. These uses are not defined
here, but are also not restricted by this document.The AGGFRAG_PAYLOAD payload content defined in this document
consists of a 4 or 24 octet header followed by either a partial
datablock, a full datablock, or multiple partial or full datablocks.
The following diagram illustrates this payload within the ESP packet.
See for the exact formats of the
AGGFRAG_PAYLOAD payload.The BlockOffset value is either zero or some offset into or past
the end of the DataBlocks data.If the BlockOffset value is zero it means that the DataBlocks
data begins with a new data block.Conversely, if the BlockOffset value is non-zero it points to the
start of the new data block, and the initial DataBlocks data
belongs to the data block that is still being re-assembled.If the BlockOffset points past the end of the DataBlocks data
then the next data block occurs in a subsequent encapsulating packet.Having the BlockOffset always point at the next available data
block allows for recovering the next inner packet in the
presence of outer encapsulating packet loss.An example AGGFRAG mode packet flow can be found in .A data block is defined by a 4-bit type code followed by the data
block data. The type values have been carefully chosen to coincide
with the IPv4/IPv6 version field values so that no per-data block
type overhead is required to encapsulate an IP packet. Likewise, the
length of the data block is extracted from the encapsulated IPv4's
Total Length or IPv6's Payload Length fields.Since a data block's type is identified in its first 4-bits, the only
time padding is required is when there is no data to encapsulate. For
this end padding a Pad Data Block is used.In order for a receiver to reassemble fragmented inner packets, the
sender MUST send the inner packet fragments back-to-back in the
logical outer packet stream (i.e., using consecutive ESP sequence
numbers). However, the sender is allowed to insert "all-pad" payloads
(i.e., payloads with a BlockOffset of zero and a single pad
DataBlock) in between the packets carrying the inner packet
fragment payloads. This interleaving of all-pad payloads allows the
sender to always send a tunnel packet, regardless of the
encapsulation computational requirements.When a receiver is reassembling an inner packet, and it receives an
"all-pad" payload, it increments the expected sequence number that
the next inner packet fragment is expected to arrive in.Given the above, the receiver will need to handle out-of-order
arrival of outer ESP packets prior to reassembly processing. ESP
already provides for optionally detecting replay attacks. Detecting
replay attacks normally utilizes a window method. A similar sequence
number based sliding window can be used to correct re-ordering of the
outer packet stream. Receiving a larger (newer) sequence number
packet advances the window, and received older ESP packets whose
sequence numbers the window has passed by are dropped. A good choice
for the size of this window depends on the amount of misordering the
user may normally experience.As the amount of misordering that may be present is hard to predict,
the window size SHOULD be configurable by the user. Implementations
MAY also dynamically adjust the reordering window based on actual
misordering seen in arriving packets.Please note, when IP-TFS sends a continuous stream of packets, there
is no requirement for an explicit lost packet timer; however, using a
lost packet timer is RECOMMENDED. If an implementation does not use a
lost packet timer and only considers an outer packet lost when the
reorder window moves by it, the inner traffic can be delayed by up to
the reorder window size times the per packet send rate. This
delay could be significant for slower send rates or when larger
reorder window sizes are in use. As the lost packet timer affects
delay of inner packet delivery, an implementation or user could choose to set it
proportionate to the tunnel rate.While ESP guarantees an increasing sequence number with subsequently
sent packets, it does not actually require the sequence numbers to be
generated consecutively (e.g., sending only even numbered sequence
numbers would be allowed as long as they are always increasing). Gaps
in the sequence numbers will not work for this document so the
sequence number stream MUST increase monotonically by 1 for each
subsequent packet.When using the AGGFRAG_PAYLOAD in conjunction with replay detection,
the window size for both MAY be reduced to the smaller of the two
window sizes. This is because packets outside of the smaller window
but inside the larger would still be dropped by the mechanism with
the smaller window size. However, there is also no requirement to
make these values the same. Indeed, in some cases, such as slow
tunnels where a very small or zero reorder window size is
appropriate, the user may still want a large replay detection window
to log replayed packets. Additionally, large replay windows can be
implemented with very little overhead compared to large reorder
windows.Finally, as sequence numbers are reset when switching SAs (e.g., when
re-keying a child SA), senders MUST NOT send initial fragments of an
inner packet using one SA and subsequent fragments in a different SA.When the tunnel bandwidth is not being fully utilized, a
sender MAY pad-out the current encapsulating packet in order
to deliver an inner packet un-fragmented in the following outer
packet. The benefit would be to avoid inner packet fragmentation in
the presence of a bursty offered load (non-bursty traffic will
naturally not fragment). Senders MAY also choose to allow
for a minimum fragment size to be configured (e.g., as a percentage
of the AGGFRAG_PAYLOAD payload size) to avoid fragmentation at the
cost of tunnel bandwidth. The cost with these methods is complexity
and added delay of inner traffic. The main advantage to avoiding
fragmentation is to minimize inner packet loss in the presence of
outer packet loss. When this is worthwhile (e.g., how much loss and
what type of loss is required, given different inner traffic shapes
and utilization, for this to make sense), and what values to use for
the allowable/added delay may be worth researching but is outside
the scope of this document.While use of padding to avoid fragmentation does not impact
interoperability, used inappropriately it can reduce the effective
throughput of a tunnel. Senders implementing either of the
above approaches will need to take care to not reduce the effective
capacity, and overall utility, of the tunnel through the overuse of
padding.To support reporting of congestion control information (described
later) using a non-AGGFRAG_PAYLOAD-enabled SA, it is allowed to send
an AGGFRAG_PAYLOAD payload with no data blocks (i.e., the ESP payload
length is equal to the AGGFRAG_PAYLOAD header length). This special
payload is called an empty payload.Currently this situation is only applicable in non-IKEv2 use cases. provides some direction on when and how to map various values
from an inner IP header to the outer encapsulating header, namely the
Don't-Fragment (DF) bit ( and ), the Differentiated
Services (DS) field and the Explicit Congestion Notification
(ECN) field . Unlike , AGGFRAG mode may and often will be
encapsulating more than one IP packet per ESP packet. To deal with
this, these mappings are restricted further.AGGFRAG mode never maps the inner DF bit as it is unrelated to the
AGGFRAG tunnel functionality; AGGFRAG mode never needs to IP fragment
the inner packets and the inner packets will not affect the
fragmentation of the outer encapsulation packets.The ECN value need not be mapped as any congestion related to the
constant-send-rate IP-TFS tunnel is unrelated (by design) to the
inner traffic flow. The sender MAY still set the ECN value of inner
packets based on the normal ECN specification .By default, the DS field SHOULD NOT be copied, although a sender MAY
choose to allow for configuration to override this behavior. A sender
SHOULD also allow the DS value to be set by configuration. specifies how to modify the inner packet TTL .Any errors (e.g., ICMP errors arriving back at the tunnel ingress due
to tunnel traffic) are handled the same as with non-AGGFRAG
IPsec tunnels.Unlike , there is normally no effective MTU (EMTU) on an
AGGFRAG tunnel as all IP packet sizes are properly transmitted without
requiring IP fragmentation prior to tunnel ingress. That said, a
sender MAY allow for explicitly configuring an MTU for the
tunnel.If fragmentation has been disabled on the AGGFRAG tunnel, then the
tunnel's EMTU and behaviors are the same as normal IPsec tunnels
.This document does not specify mixed use of an
AGGFRAG_PAYLOAD-enabled SA. A sender MUST only send AGGFRAG_PAYLOAD
payloads over an SA configured for AGGFRAG mode.Just as with normal IPsec/ESP tunnels, AGGFRAG tunnels are
unidirectional. Bidirectional IP-TFS functionality is achieved by
setting up 2 AGGFRAG tunnels, one in either direction.An AGGFRAG tunnel used for IP-TFS can operate in 2 modes, a
non-congestion-controlled mode and congestion-controlled mode.In the non-congestion-controlled mode, IP-TFS sends fixed-sized
packets over an AGGFRAG tunnel at a constant rate. The packet send
rate is constant and is not automatically adjusted regardless of any
network congestion (e.g., packet loss).For similar reasons as given in the non-congestion-controlled
mode should only be used where the user has full administrative
control over the path the tunnel will take. This is required so the
user can guarantee the bandwidth and also be sure as to not be
negatively affecting network congestion . In this case, packet
loss should be reported to the administrator (e.g., via syslog, YANG
notification, SNMP traps, etc.) so that any failures due to a lack of
bandwidth can be corrected.Non-congestion-controlled mode is also appropriate if ESP over TCP is in
use .With the congestion-controlled mode, IP-TFS adapts to network
congestion by lowering the packet send rate to accommodate the
congestion, as well as raising the rate when congestion subsides.
Since overhead is per packet, by allowing for maximal fixed-size
packets and varying the send rate, transport overhead is minimized.The output of the congestion control algorithm will adjust the rate
at which the ingress sends packets. While this document does not
require a specific congestion control algorithm, best current
practice RECOMMENDS that the algorithm conform to . Congestion
control principles are documented in as well.
provides an example of the algorithm which matches the
requirements of IP-TFS (i.e., designed for fixed-size packets and send
rate varied based on congestion).The required inputs for the TCP friendly rate control algorithm
described in are the receiver's loss event rate and the
sender's estimated round-trip time (RTT). These values are provided by
IP-TFS using the congestion information header fields described in
. In particular, these values are sufficient to
implement the algorithm described in .At a minimum, the congestion information MUST be sent, from the
receiver and from the sender, at least once per RTT. Prior to
establishing an RTT the information SHOULD be sent constantly from
the sender and the receiver so that an RTT estimate can be
established. Not receiving this information over multiple
consecutive RTT intervals should be considered a congestion event
that causes the sender to adjust its sending rate lower. For
example, calls this the "no feedback timeout" and it is equal
to 4 RTT intervals. When a "no feedback timeout" has occurred
halves the sending rate.An implementation MAY choose to always include the congestion
information in its AGGFRAG payload header if sending on an IP-TFS-enabled
SA. Since IP-TFS normally will operate with a large packet
size, the congestion information should represent a small portion of
the available tunnel bandwidth. An implementation choosing to always
send the data MAY also choose to only update the LossEventRate
and RTT header field values it sends every RTT though.When choosing a congestion control algorithm (or a selection of
algorithms), note that IP-TFS is not providing for reliable delivery
of IP traffic, and so per packet ACKs are not required and are not
provided.It is worth noting that the variable send-rate of a
congestion-controlled AGGFRAG tunnel, is not private; however, this
send-rate is being driven by network congestion, and as long as the
encapsulated (inner) traffic flow shape and timing are not directly
affecting the (outer) network congestion, the variations in the
tunnel rate will not weaken the provided inner traffic flow
confidentiality.In additional to congestion control, implementations MAY choose to
define and implement circuit breakers as a recovery method
of last resort. Enabling circuit breakers is also a reason a user may
wish to enable congestion information reports even when using the
non-congestion-controlled mode of operation.An AGGFRAG-enabled SA receiver has a few tasks to perform.The receiver MAY process incoming AGGFRAG_PAYLOAD payloads as soon as
they arrive as much as it can. I.e., if the incoming AGGFRAG_PAYLOAD
packet contains complete inner packet(s), the receiver should extract
and transmit them immediately. For partial packets, the receiver needs
to keep the partial packets in the memory until they fall out
from the reordering window, or until the missing parts of the packets
are received, in which case it will reassemble and transmit them. If
the AGGFRAG_PAYLOAD payload contains multiple packets they SHOULD be sent
out in the order they are in the AGGFRAG_PAYLOAD (i.e., keep the
original order they were received on the other end). The cost of
using this method is that an amplification of out-of-order delivery
of inner packets can occur due to inner packet aggregation.Instead of the method described in the previous paragraph, the
receiver MAY reorder out-of-order AGGFRAG_PAYLOAD payloads received
into in-sequence-order AGGFRAG_PAYLOAD payloads (), and only after it has an
in-order AGGFRAG_PAYLOAD payload stream would the receiver transmits
the inner packets. Using this method will ensure the inner packets
are sent in order. The cost of this method is that a lost packet will
cause a delay of up to the lost packet timer interval (or the full
reorder window if no lost packet timer is used). Additionally, there
can be extra burstiness in the output stream. This burstiness can
happen when a lost packet is dropped from the re-order window,
and the remaining outer packets in the re-order window are immediately
processed and sent out back to back.Additionally, if congestion control is enabled, the receiver sends
congestion control data () back to the sender as described in
and .In order to support the congestion-controlled mode, the sender needs to
know the loss event rate and to approximate the RTT . In order
to obtain these values, the receiver sends congestion control
information on its SA back to the sender. Thus, to support
congestion control the receiver MUST have a paired SA back to the
sender (this is always the case when the tunnel was created using
IKEv2). If the SA back to the sender is a non-AGGFRAG_PAYLOAD enabled
SA then an AGGFRAG_PAYLOAD empty payload (i.e., header only) is used
to convey the information.In order to calculate a loss event rate compatible with , the
receiver needs to have a round-trip time estimate. Thus the sender
communicates this estimate in the RTT header field. On startup this
value will be zero as no RTT estimate is yet known.In order for the sender to estimate its RTT value, the sender
places a timestamp value in the TVal header field. On first receipt
of this TVal, the receiver records the new TVal value along with
the time it arrived locally. Subsequent receipt of the same TVal
MUST NOT update the recorded time.When the receiver sends its congestion control header it places this latest recorded
TVal in the TEcho header field, along with 2 delay values, Echo
Delay and Transmit Delay. The Echo Delay value is the time delta
from the recorded arrival time of TVal and the current clock in
microseconds. The second value, Transmit Delay, is the receiver's
current transmission delay on the tunnel (i.e., the average time
between sending packets on its half of the AGGFRAG tunnel).When the sender receives back its TVal in the TEcho header field
it calculates 2 RTT estimates. The first is the actual delay found by
subtracting the TEcho value from its current clock and then
subtracting Echo Delay as well. The second RTT estimate is found by
adding the received Transmit Delay header value to the sender's own
transmission delay (i.e., the average time between sending packets on
its half of the AGGFRAG tunnel). The larger of these 2 RTT estimates
SHOULD be used as the RTT value.The two RTT estimates are required to handle different combinations of
faster or slower tunnel packet paths with faster or slower fixed
tunnel rates. Choosing the larger of the two values guarantees that
the RTT is never considered faster than the aggregate transmission
delay based on the IP-TFS send rate (the second estimate), as well
as never being considered faster than the actual RTT along the tunnel
packet path (the first estimate).The receiver also calculates, and communicates in the LossEventRate
header field, the loss event rate for use by the sender. This is
slightly different from which periodically sends all the loss
interval data back to the sender so that it can do the calculation.
See for a suggested way to
calculate the loss event rate value. Initially this value will be
zero (indicating no loss) until enough data has been collected by the
receiver to update it.In additional to normal packet loss information AGGFRAG mode supports use
of the ECN bits in the encapsulating IP header for
identifying congestion. If ECN use is enabled and a packet arrives at
the egress (receiving) side with the Congestion Experienced (CE) value set,
then the receiver considers that packet as being dropped, although it
does not drop it. The receiver MUST set the E bit in any
AGGFRAG_PAYLOAD payload header containing a LossEventRate value
derived from a CE value being considered.As noted in the ECN bits are not protected by IPsec and
thus may constitute a covert channel. For this reason, ECN use SHOULD
NOT be enabled by default.IP-TFS is meant to be deployable with a minimal amount of
configuration. All IP-TFS specific configuration should be
specified at the unidirectional tunnel ingress (sending) side. It
is intended that non-IKEv2 operation is supported, at least, with
local static configuration.YANG and MIB documents have been defined for IP-TFS in
and .Bandwidth is a local configuration option. For
non-congestion-controlled mode, the bandwidth SHOULD be configured.
For congestion-controlled mode, the bandwidth can be configured or
the congestion control algorithm discovers and uses the maximum
bandwidth available. No standardized configuration method is
required.The fixed packet size to be used for the tunnel encapsulation packets
MAY be configured manually or can be automatically determined using
other methods such as PLMTUD (, ) or PMTUD (,
). As PMTUD is known to have issues, PLMTUD is considered the
more robust option. No standardized configuration method is required.Congestion control is a local configuration option. No standardized
configuration method is required.As mentioned previously AGGFRAG tunnels utilize ESP payloads of type
AGGFRAG_PAYLOAD.When using IKEv2, a new "USE_AGGFRAG" Notification Message enables
the AGGFRAG_PAYLOAD payload on a child SA pair. The
method used is similar to how USE_TRANSPORT_MODE is negotiated, as
described in .To request use of the AGGFRAG_PAYLOAD payload on the Child SA pair,
the initiator includes the USE_AGGFRAG notification in an SA payload
requesting a new Child SA (either during the initial IKE_AUTH or
during CREATE_CHILD_SA exchanges). If the request is
accepted then the response MUST also include a notification of type
USE_AGGFRAG. If the responder declines the request the child SA will
be established without AGGFRAG_PAYLOAD payload use enabled. If
this is unacceptable to the initiator, the initiator MUST delete the
child SA.As the use of the AGGFRAG_PAYLOAD payload is currently only defined
for non-transport mode tunnels, the USE_AGGFRAG notification MUST NOT
be combined with USE_TRANSPORT notification.The USE_AGGFRAG notification contains a 1 octet payload of flags that
specify requirements from the sender of the notification. If any
requirement flags are not understood or cannot be supported by the
receiver then the receiver SHOULD NOT enable use of AGGFRAG_PAYLOAD
(either by not responding with the USE_AGGFRAG notification, or in
the case of the initiator, by deleting the child SA if the now
established non-AGGFRAG_PAYLOAD using SA is unacceptable).The notification type and payload flag values are defined in .The packet and data formats defined below are generic with the intent
of allowing for non-IP-TFS uses, but such uses are outside the scope of
this document.ESP Next Header value: 0x5An AGGFRAG payload is identified by the ESP Next Header value
AGGFRAG_PAYLOAD which has the value 0x5. The value 5 was chosen to not
conflict with other used values. The first octet of this payload
indicates the format of the remaining payload data.An 8-bit value indicating the payload format.This document defines 2 payload sub-types. These payload formats
are defined in the following sections.The non-congestion control AGGFRAG_PAYLOAD payload consists of a
4-octet header followed by a variable amount of DataBlocks data as
shown below.An octet indicating the payload format. For this
non-congestion control format, the value is 0.An octet set to 0 on generation and ignored on
receipt.A 16-bit unsigned integer counting the number of
octets of DataBlocks data before the start of a
new data block. If the start of a new data block
occurs in a subsequent payload the BlockOffset
will point past the end of the DataBlocks data.
In this case all the DataBlocks data belongs to
the current data block being assembled. When the
BlockOffset extends into subsequent payloads it
continues to only count DataBlocks data (i.e.,
it does not count subsequent packets
non-DataBlocks data such as header octets).Variable number of octets that begins with the start
of a data block, or the continuation of a previous
data block, followed by zero or more additional data
blocks.The congestion control AGGFRAG_PAYLOAD payload consists of a 24
octet header followed by a variable amount of DataBlocks data as
shown below.An octet indicating the payload format. For this
congestion control format, the value is 1.A 6-bit field set to 0 on generation and ignored on
receipt.A 1-bit value that if set indicates that PLMTUD probing is in
progress. This information can be used to avoid treating
missing packets as loss events by the CC algorithm when
running the PLMTUD probe algorithm.A 1-bit value that if set indicates that Congestion Experienced
(CE) ECN bits were received and used in deriving the
reported LossEventRate.The same value as the non-congestion-controlled
payload format value.A 32-bit value specifying the inverse of the
current loss event rate as calculated by the
receiver. A value of zero indicates no loss.
Otherwise the loss event rate is
1/LossEventRate.A 22-bit value specifying the sender's current round-trip
time estimate in microseconds. The value MAY be zero prior
to the sender having calculated a round-trip time estimate.
The value SHOULD be set to zero on
non-AGGFRAG_PAYLOAD-enabled SAs. If the RTT is equal to or
larger than 0x3FFFFF the value MUST be set to 0x3FFFFF.A 21-bit value specifying the delay in microseconds
incurred between the receiver first receiving the TVal
value which it is sending back in TEcho. If the delay
is equal to or larger than 0x1FFFFF the value MUST be
set to 0x1FFFFF.A 21-bit value specifying the transmission delay in
microseconds. This is the fixed (or average) delay on the
receiver between it sending packets on the IPTFS tunnel.
If the delay is equal to or larger than 0x1FFFFF the
value MUST be set to 0x1FFFFF.An opaque 32-bit value that will be echoed back by the
receiver in later packets in the TEcho field, along with
an Echo Delay value of how long that echo took.The opaque 32-bit value from a received packet's TVal
field. The received TVal is placed in TEcho along with
an Echo Delay value indicating how long it has been since
receiving the TVal value.Variable number of octets that begins with the start
of a data block, or the continuation of a previous
data block, followed by zero or more additional data
blocks. For the special case of sending congestion
control information on a non-IP-TFS enabled SA this
field MUST be empty (i.e., be zero octets long).A 4-bit field where 0x0 identifies a pad data block, 0x4
indicates an IPv4 data block, and 0x6 indicates an IPv6
data block.These values are the actual values within the encapsulated IPv4
header. In other words, the start of this data block is the start of
the encapsulated IP packet.A 4-bit value of 0x4 indicating IPv4 (i.e., first nibble of
the IPv4 packet).The 16-bit unsigned integer "Total Length" field of
the IPv4 inner packet.These values are the actual values within the encapsulated IPv6
header. In other words, the start of this data block is the start of
the encapsulated IP packet.A 4-bit value of 0x6 indicating IPv6 (i.e., first nibble of
the IPv6 packet).The 16-bit unsigned integer "Payload Length" field
of the inner IPv6 inner packet.A 4-bit value of 0x0 indicating a padding data block.Extends to end of the encapsulating packet.As discussed in , a notification
message USE_AGGFRAG is used to negotiate use of the ESP AGGFRAG_PAYLOAD
Next Header value.The USE_AGGFRAG Notification Message State Type is (TBD2).The notification payload contains 1 octet of requirement flags. There
are currently 2 requirement flags defined. This may be revised by
later specifications.6 bits - Reserved MUST be zero on send, unless defined by
later specifications.Congestion Control bit. If set, then the sender is requiring
that congestion control information MUST be returned to it
periodically as defined in .Don't Fragment bit. If set, indicates the sender of the notify
message does not support receiving packet fragments (i.e., inner
packets MUST be sent using a single Data Block). This value only
applies to what the sender is capable of receiving; the sender MAY
still send packet fragments unless similarly restricted by the
receiver in its USE_AGGFRAG notification.This document requests IANA create a registry called "AGGFRAG_PAYLOAD
Sub-Type Registry" under a new category named "ESP AGGFRAG_PAYLOAD Parameters".
The registration policy for this registry is "Expert Review"
( and ).AGGFRAG_PAYLOAD Sub-Type RegistryAGGFRAG_PAYLOAD Payload Formats.This documentThis initial content for this registry is as follows:This document requests a status type USE_AGGFRAG be allocated from
the "IKEv2 Notify Message Types - Status Types" registry.TBD2USE_AGGFRAGThis documentThis document describes an aggregation and fragmentation mechanism to
efficiently implement TFC for IP traffic. This approach is expected to reduce
the efficacy of traffic analysis on IPsec communication. Other than
the additional security afforded by using this mechanism, IP-TFS
utilizes the security protocols and and so their
security considerations apply to IP-TFS as well.As noted in , the ECN bits are not protected by IPsec and
thus may constitute a covert channel. For this reason, ECN use SHOULD
NOT be enabled by default.As noted previously in , for TFC to be
maintained, the encapsulated traffic flow should not be
affecting network congestion in a predictable way, and if it would be,
then non-congestion-controlled mode use should be considered instead.Key words for use in RFCs to Indicate Requirement LevelsIn many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.IP Encapsulating Security Payload (ESP)This document describes an updated version of the Encapsulating Security Payload (ESP) protocol, which is designed to provide a mix of security services in IPv4 and IPv6. ESP is used to provide confidentiality, data origin authentication, connectionless integrity, an anti-replay service (a form of partial sequence integrity), and limited traffic flow confidentiality. This document obsoletes RFC 2406 (November 1998). [STANDARDS-TRACK]Internet Key Exchange Protocol Version 2 (IKEv2)This document describes version 2 of the Internet Key Exchange (IKE) protocol. IKE is a component of IPsec used for performing mutual authentication and establishing and maintaining Security Associations (SAs). This document obsoletes RFC 5996, and includes all of the errata for it. It advances IKEv2 to be an Internet Standard.Ambiguity of Uppercase vs Lowercase in RFC 2119 Key WordsRFC 2119 specifies common key words that may be used in protocol specifications. This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the defined special meanings.Applied Cryptography: Protocols, Algorithms, and Source Code in CInternet ProtocolPath MTU discoveryThis memo describes a technique for dynamically discovering the maximum transmission unit (MTU) of an arbitrary internet path. It specifies a small change to the way routers generate one type of ICMP message. For a path that passes through a router that has not been so changed, this technique might not discover the correct Path MTU, but it will always choose a Path MTU as accurate as, and in many cases more accurate than, the Path MTU that would be chosen by current practice. [STANDARDS-TRACK]Definition of the Differentiated Services Field (DS Field) in the IPv4 and IPv6 HeadersThis document defines the IP header field, called the DS (for differentiated services) field. [STANDARDS-TRACK]Congestion Control PrinciplesThe goal of this document is to explain the need for congestion control in the Internet, and to discuss what constitutes correct congestion control. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.The Addition of Explicit Congestion Notification (ECN) to IPThis memo specifies the incorporation of ECN (Explicit Congestion Notification) to TCP and IP, including ECN's use of two bits in the IP header. [STANDARDS-TRACK]Security Architecture for the Internet ProtocolThis document describes an updated version of the "Security Architecture for IP", which is designed to provide security services for traffic at the IP layer. This document obsoletes RFC 2401 (November 1998). [STANDARDS-TRACK]Profile for Datagram Congestion Control Protocol (DCCP) Congestion Control ID 3: TCP-Friendly Rate Control (TFRC)This document contains the profile for Congestion Control Identifier 3, TCP-Friendly Rate Control (TFRC), in the Datagram Congestion Control Protocol (DCCP). CCID 3 should be used by senders that want a TCP-friendly sending rate, possibly with Explicit Congestion Notification (ECN), while minimizing abrupt rate changes. [STANDARDS-TRACK]Packetization Layer Path MTU DiscoveryThis document describes a robust method for Path MTU Discovery (PMTUD) that relies on TCP or some other Packetization Layer to probe an Internet path with progressively larger packets. This method is described as an extension to RFC 1191 and RFC 1981, which specify ICMP-based Path MTU Discovery for IP versions 4 and 6, respectively. [STANDARDS-TRACK]TCP Friendly Rate Control (TFRC): Protocol SpecificationThis document specifies TCP Friendly Rate Control (TFRC). TFRC is a congestion control mechanism for unicast flows operating in a best-effort Internet environment. It is reasonably fair when competing for bandwidth with TCP flows, but has a much lower variation of throughput over time compared with TCP, making it more suitable for applications such as streaming media where a relatively smooth sending rate is of importance.This document obsoletes RFC 3448 and updates RFC 4342. [STANDARDS-TRACK]Early IANA Allocation of Standards Track Code PointsThis memo describes the process for early allocation of code points by IANA from registries for which "Specification Required", "RFC Required", "IETF Review", or "Standards Action" policies apply. This process can be used to alleviate the problem where code point allocation is needed to facilitate desired or required implementation and deployment experience prior to publication of an RFC, which would normally trigger code point allocation. The procedures in this document are intended to apply only to IETF Stream documents.Encapsulating MPLS in UDPThis document specifies an IP-based encapsulation for MPLS, called MPLS-in-UDP for situations where UDP (User Datagram Protocol) encapsulation is preferred to direct use of MPLS, e.g., to enable UDP-based ECMP (Equal-Cost Multipath) or link aggregation. The MPLS- in-UDP encapsulation technology must only be deployed within a single network (with a single network operator) or networks of an adjacent set of cooperating network operators where traffic is managed to avoid congestion, rather than over the Internet where congestion control is required. Usage restrictions apply to MPLS-in-UDP usage for traffic that is not congestion controlled and to UDP zero checksum usage with IPv6.Network Transport Circuit BreakersThis document explains what is meant by the term "network transport Circuit Breaker". It describes the need for Circuit Breakers (CBs) for network tunnels and applications when using non-congestion- controlled traffic and explains where CBs are, and are not, needed. It also defines requirements for building a CB and the expected outcomes of using a CB within the Internet.Guidelines for Writing an IANA Considerations Section in RFCsMany protocols make use of points of extensibility that use constants to identify various protocol parameters. To ensure that the values in these fields do not have conflicting uses and to promote interoperability, their allocations are often coordinated by a central record keeper. For IETF protocols, that role is filled by the Internet Assigned Numbers Authority (IANA).To make assignments in a given registry prudently, guidance describing the conditions under which new values should be assigned, as well as when and how modifications to existing values can be made, is needed. This document defines a framework for the documentation of these guidelines by specification authors, in order to assure that the provided guidance for the IANA Considerations is clear and addresses the various issues that are likely in the operation of a registry.This is the third edition of this document; it obsoletes RFC 5226.Internet Protocol, Version 6 (IPv6) SpecificationThis document specifies version 6 of the Internet Protocol (IPv6). It obsoletes RFC 2460.Path MTU Discovery for IP version 6This document describes Path MTU Discovery (PMTUD) for IP version 6. It is largely derived from RFC 1191, which describes Path MTU Discovery for IP version 4. It obsoletes RFC 1981.TCP Encapsulation of IKE and IPsec PacketsThis document describes a method to transport Internet Key Exchange Protocol (IKE) and IPsec packets over a TCP connection for traversing network middleboxes that may block IKE negotiation over UDP. This method, referred to as "TCP encapsulation", involves sending both IKE packets for Security Association establishment and Encapsulating Security Payload (ESP) packets over a TCP connection. This method is intended to be used as a fallback option when IKE cannot be negotiated over UDP.The Wire Image of a Network ProtocolThis document defines the wire image, an abstraction of the information available to an on-path non-participant in a networking protocol. This abstraction is intended to shed light on the implications that increased encryption has for network functions that use the wire image.Packetization Layer Path MTU Discovery for Datagram TransportsThis document specifies Datagram Packetization Layer Path MTU Discovery (DPLPMTUD). This is a robust method for Path MTU Discovery (PMTUD) for datagram Packetization Layers (PLs). It allows a PL, or a datagram application that uses a PL, to discover whether a network path can support the current size of datagram. This can be used to detect and reduce the message size when a sender encounters a packet black hole. It can also probe a network path to discover whether the maximum packet size can be increased. This provides functionality for datagram transports that is equivalent to the PLPMTUD specification for TCP, specified in RFC 4821, which it updates. It also updates the UDP Usage Guidelines to refer to this method for use with UDP datagrams and updates SCTP.The document provides implementation notes for incorporating Datagram PMTUD into IETF datagram transports or applications that use datagram transports.This specification updates RFC 4960, RFC 4821, RFC 6951, RFC 8085, and RFC 8261.Definitions of Managed Objects for IP Traffic Flow SecurityLabN Consulting, L.L.C.LabN Consulting, L.L.C. This document describes managed objects for the the management of IP
Traffic Flow Security additions to IKEv2 and IPsec. This document
provides a read only version of the objects defined in the YANG
module for the same purpose.
A YANG Data Model for IP Traffic Flow SecurityLabN Consulting, L.L.C.LabN Consulting, L.L.C. This document describes a yang module for the management of IP
Traffic Flow Security additions to IKEv2 and IPsec.
Below, an example inner IP packet flow within the encapsulating tunnel
packet stream is shown. Notice how encapsulated IP packets can start
and end anywhere, and more than one or less than 1 may occur in a
single encapsulating packet.Each outer encapsulating ESPupayload space is a fixed-size of 1404
octets the first 4 octets of which contains the AGGFRAG header.
The encapsulated IP packet flow (lengths include IP header and
payload) is as follows: a 750-octet packet, a 750-octet packet, a
60-octet packet, a 240-octet packet, a 3000-octet packet.The BlockOffset values in the 4 AGGFRAG payload headers for this
packet flow would thus be: 0, 100, 2000, 600 respectively. The first
encapsulating packet (ESP1) has a zero BlockOffset which points at the
IP data block immediately following the AGGFRAG header. The following
packet's (ESP2) BlockOffset points inward 100 octets to the start of the
60-octet data block. The third encapsulating packet (ESP3) contains the
middle portion of the 3000-octet data block so the offset points past
its end and into the fourth encapsulating packet. The fourth packet's
(ESP4) offset is 600, pointing at the padding which follows the
completion of the continued 3000-octet packet.The current best practice indicates that congestion control SHOULD be
done in a TCP-friendly way. A TCP-friendly congestion control algorithm
is described in . For this IP-TFS use case (as with ), the
(fixed) packet size is used as the segment size for the algorithm. The
main formula in the algorithm for the send rate is then as follows:Where X is the send rate in packets per second, R is the
round trip time estimate and p is the loss event rate (the inverse
of which is provided by the receiver).In addition, the algorithm in also uses an X_recv value (the
receiver's receive rate). For IP-TFS one MAY set this value according to
the sender's current tunnel send-rate (X).The IP-TFS receiver, having the RTT estimate from the sender can use the
same method as described in and to collect the loss
intervals and calculate the loss event rate value using the weighted
average as indicated. The receiver communicates the inverse of this
value back to the sender in the AGGFRAG_PAYLOAD payload header field
LossEventRate.The IP-TFS sender now has both the R and p values and can calculate
the correct sending rate. If following , the sender should also
use the slow start mechanism described therein when the IP-TFS SA is
first established.For comparing overhead, the overhead of ESP for both normal and AGGFRAG
tunnel packets must be calculated, and so an algorithm for encryption
and authentication must be chosen. For the data below AES-GCM-256 was
selected. This leads to an IP+ESP overhead of 54.Additionally, for IP-TFS, non-congestion control AGGFRAG_PAYLOAD
headers were chosen which adds 4 octets for a total overhead of 58.For comparison, the overhead of an AGGFRAG payload is 58 octets per outer packet.
Therefore, the octet overhead per inner packet is 58 divided by the
number of outer packets required (fractions allowed). The overhead
as a percentage of inner packet size is a constant based on the Outer
MTU size.The overhead per inner packet for constant-send-rate padded ESP
(i.e., traditional IPsec TFC) is 36 octets plus any padding, unless
fragmentation is required.When fragmentation of the inner packet is required to fit in the
outer IPsec packet, overhead is the number of outer packets required
to carry the fragmented inner packet times both the inner IP overhead
(20) and the outer packet overhead (54) minus the initial inner IP
overhead plus any required tail padding in the last encapsulation
packet. The required tail padding is the number of required packets
times the difference of the Outer Payload Size and the IP Overhead
minus the Inner Payload Size. So:The following tables collect the overhead values for some common L3
MTU sizes in order to compare them. The first table is the number of
octets of overhead for a given L3 MTU sized packet. The second table
is the percentage of overhead in the same MTU sized packet.Another way to compare the two solutions is to look at the amount of
available bandwidth each solution provides. The following sections
consider and compare the percentage of available bandwidth. For the
sake of providing a well-understood baseline normal (unencrypted)
Ethernet as well as normal ESP values are included.In order to calculate the available bandwidth the per packet overhead
is calculated first. The total overhead of Ethernet is 14+4 octets of
header and CRC plus an additional 20 octets of framing (preamble,
start, and inter-packet gap), for a total of 38 octets. Additionally,
the minimum payload is 46 octets.A sometimes unexpected result of using an AGGFRAG tunnel (or any packet
aggregating tunnel) is that, for small- to medium-sized packets, the
available bandwidth is actually greater than native Ethernet. This is
due to the reduction in Ethernet framing overhead. This increased
bandwidth is paid for with an increase in latency. This latency is
the time to send the unrelated octets in the outer tunnel frame. The
following table illustrates the latency for some common values on a
10G Ethernet link. The table also includes latency introduced by
padding if using ESP with padding.Notice that the latency values are very similar between the two
solutions; however, whereas IP-TFS provides for constant high
bandwidth, in some cases even exceeding native Ethernet, ESP with
padding often greatly reduces available bandwidth.We would like to thank Don Fedyk for help in reviewing and editing
this work. We would also like to thank Michael Richardson, Sean
Turner, Valery Smyslov and Tero Kivinen for reviews and many
suggestions for improvements, as well as Joseph Touch for the
transport area review and suggested improvements.The following people made significant contributions to this document.