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  1. 15 5月, 2020 1 次提交
    • X
      alinux: add tcprt framework to kernel · f84e8fa0
      由 xuanzhuo 提交于 
      to #26353046

TcpRT: Instrument and Diagnostic Analysis System for Service Quality
of Cloud Databases at Massive Scale in Real-time.

It can also provide information for all request/response services. Such as
HTTP request.

This is the kernel framework for tcprt, more work needs tcprt module
support.

TcpRt module should call tcp_unregitsert_rt before rmmod.

TcpRt hooks will be called when sock init, recv data, send data,
packet acked and socket been destroy. The private data save to
icsk->icsk_tcp_rt_priv.
Reviewed-by: NCambda Zhu <cambda@linux.alibaba.com>
Acked-by: NDust Li <dust.li@linux.alibaba.com>
Signed-off-by: Nxuanzhuo <xuanzhuo@linux.alibaba.com>
      f84e8fa0
  3. 25 7月, 2018 1 次提交
  5. 02 3月, 2018 1 次提交
  7. 13 10月, 2017 1 次提交
  9. 15 2月, 2017 1 次提交
    • S
      esp: Add a software GRO codepath · 7785bba2
      由 Steffen Klassert 提交于 
      This patch adds GRO ifrastructure and callbacks for ESP on
ipv4 and ipv6.

In case the GRO layer detects an ESP packet, the
esp{4,6}_gro_receive() function does a xfrm state lookup
and calls the xfrm input layer if it finds a matching state.
The packet will be decapsulated and reinjected it into layer 2.
Signed-off-by: NSteffen Klassert <steffen.klassert@secunet.com>
      7785bba2
  11. 09 2月, 2017 1 次提交
  13. 29 11月, 2016 1 次提交
  15. 24 10月, 2016 1 次提交
    • C
      net: ip, diag -- Add diag interface for raw sockets · 432490f9
      由 Cyrill Gorcunov 提交于 
      In criu we are actively using diag interface to collect sockets
present in the system when dumping applications. And while for
unix, tcp, udp[lite], packet, netlink it works as expected,
the raw sockets do not have. Thus add it.

v2:
 - add missing sock_put calls in raw_diag_dump_one (by eric.dumazet@)
 - implement @destroy for diag requests (by dsa@)

v3:
 - add export of raw_abort for IPv6 (by dsa@)
 - pass net-admin flag into inet_sk_diag_fill due to
   changes in net-next branch (by dsa@)

v4:
 - use @pad in struct inet_diag_req_v2 for raw socket
   protocol specification: raw module carries sockets
   which may have custom protocol passed from socket()
   syscall and sole @sdiag_protocol is not enough to
   match underlied ones
 - start reporting protocol specifed in socket() call
   when sockets are raw ones for the same reason: user
   space tools like ss may parse this attribute and use
   it for socket matching

v5 (by eric.dumazet@):
 - use sock_hold in raw_sock_get instead of atomic_inc,
   we're holding (raw_v4_hashinfo|raw_v6_hashinfo)->lock
   when looking up so counter won't be zero here.

v6:
 - use sdiag_raw_protocol() helper which will access @pad
   structure used for raw sockets protocol specification:
   we can't simply rename this member without breaking uapi

v7:
 - sine sdiag_raw_protocol() helper is not suitable for
   uapi lets rather make an alias structure with proper
   names. __check_inet_diag_req_raw helper will catch
   if any of structure unintentionally changed.

CC: David S. Miller <davem@davemloft.net>
CC: Eric Dumazet <eric.dumazet@gmail.com>
CC: David Ahern <dsa@cumulusnetworks.com>
CC: Alexey Kuznetsov <kuznet@ms2.inr.ac.ru>
CC: James Morris <jmorris@namei.org>
CC: Hideaki YOSHIFUJI <yoshfuji@linux-ipv6.org>
CC: Patrick McHardy <kaber@trash.net>
CC: Andrey Vagin <avagin@openvz.org>
CC: Stephen Hemminger <stephen@networkplumber.org>
Signed-off-by: NCyrill Gorcunov <gorcunov@openvz.org>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      432490f9
  17. 21 9月, 2016 1 次提交
    • N
      tcp_bbr: add BBR congestion control · 0f8782ea
      由 Neal Cardwell 提交于 
      This commit implements a new TCP congestion control algorithm: BBR
(Bottleneck Bandwidth and RTT). A detailed description of BBR will be
published in ACM Queue, Vol. 14 No. 5, September-October 2016, as
"BBR: Congestion-Based Congestion Control".

BBR has significantly increased throughput and reduced latency for
connections on Google's internal backbone networks and google.com and
YouTube Web servers.

BBR requires only changes on the sender side, not in the network or
the receiver side. Thus it can be incrementally deployed on today's
Internet, or in datacenters.

The Internet has predominantly used loss-based congestion control
(largely Reno or CUBIC) since the 1980s, relying on packet loss as the
signal to slow down. While this worked well for many years, loss-based
congestion control is unfortunately out-dated in today's networks. On
today's Internet, loss-based congestion control causes the infamous
bufferbloat problem, often causing seconds of needless queuing delay,
since it fills the bloated buffers in many last-mile links. On today's
high-speed long-haul links using commodity switches with shallow
buffers, loss-based congestion control has abysmal throughput because
it over-reacts to losses caused by transient traffic bursts.

In 1981 Kleinrock and Gale showed that the optimal operating point for
a network maximizes delivered bandwidth while minimizing delay and
loss, not only for single connections but for the network as a
whole. Finding that optimal operating point has been elusive, since
any single network measurement is ambiguous: network measurements are
the result of both bandwidth and propagation delay, and those two
cannot be measured simultaneously.

While it is impossible to disambiguate any single bandwidth or RTT
measurement, a connection's behavior over time tells a clearer
story. BBR uses a measurement strategy designed to resolve this
ambiguity. It combines these measurements with a robust servo loop
using recent control systems advances to implement a distributed
congestion control algorithm that reacts to actual congestion, not
packet loss or transient queue delay, and is designed to converge with
high probability to a point near the optimal operating point.

In a nutshell, BBR creates an explicit model of the network pipe by
sequentially probing the bottleneck bandwidth and RTT. On the arrival
of each ACK, BBR derives the current delivery rate of the last round
trip, and feeds it through a windowed max-filter to estimate the
bottleneck bandwidth. Conversely it uses a windowed min-filter to
estimate the round trip propagation delay. The max-filtered bandwidth
and min-filtered RTT estimates form BBR's model of the network pipe.

Using its model, BBR sets control parameters to govern sending
behavior. The primary control is the pacing rate: BBR applies a gain
multiplier to transmit faster or slower than the observed bottleneck
bandwidth. The conventional congestion window (cwnd) is now the
secondary control; the cwnd is set to a small multiple of the
estimated BDP (bandwidth-delay product) in order to allow full
utilization and bandwidth probing while bounding the potential amount
of queue at the bottleneck.

When a BBR connection starts, it enters STARTUP mode and applies a
high gain to perform an exponential search to quickly probe the
bottleneck bandwidth (doubling its sending rate each round trip, like
slow start). However, instead of continuing until it fills up the
buffer (i.e. a loss), or until delay or ACK spacing reaches some
threshold (like Hystart), it uses its model of the pipe to estimate
when that pipe is full: it estimates the pipe is full when it notices
the estimated bandwidth has stopped growing. At that point it exits
STARTUP and enters DRAIN mode, where it reduces its pacing rate to
drain the queue it estimates it has created.

Then BBR enters steady state. In steady state, PROBE_BW mode cycles
between first pacing faster to probe for more bandwidth, then pacing
slower to drain any queue that created if no more bandwidth was
available, and then cruising at the estimated bandwidth to utilize the
pipe without creating excess queue. Occasionally, on an as-needed
basis, it sends significantly slower to probe for RTT (PROBE_RTT
mode).

BBR has been fully deployed on Google's wide-area backbone networks
and we're experimenting with BBR on Google.com and YouTube on a global
scale.  Replacing CUBIC with BBR has resulted in significant
improvements in network latency and application (RPC, browser, and
video) metrics. For more details please refer to our upcoming ACM
Queue publication.

Example performance results, to illustrate the difference between BBR
and CUBIC:

Resilience to random loss (e.g. from shallow buffers):
  Consider a netperf TCP_STREAM test lasting 30 secs on an emulated
  path with a 10Gbps bottleneck, 100ms RTT, and 1% packet loss
  rate. CUBIC gets 3.27 Mbps, and BBR gets 9150 Mbps (2798x higher).

Low latency with the bloated buffers common in today's last-mile links:
  Consider a netperf TCP_STREAM test lasting 120 secs on an emulated
  path with a 10Mbps bottleneck, 40ms RTT, and 1000-packet bottleneck
  buffer. Both fully utilize the bottleneck bandwidth, but BBR
  achieves this with a median RTT 25x lower (43 ms instead of 1.09
  secs).

Our long-term goal is to improve the congestion control algorithms
used on the Internet. We are hopeful that BBR can help advance the
efforts toward this goal, and motivate the community to do further
research.

Test results, performance evaluations, feedback, and BBR-related
discussions are very welcome in the public e-mail list for BBR:

  https://groups.google.com/forum/#!forum/bbr-dev

NOTE: BBR *must* be used with the fq qdisc ("man tc-fq") with pacing
enabled, since pacing is integral to the BBR design and
implementation. BBR without pacing would not function properly, and
may incur unnecessary high packet loss rates.
Signed-off-by: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      0f8782ea
  19. 11 6月, 2016 1 次提交
  21. 18 2月, 2016 1 次提交
  23. 17 2月, 2016 1 次提交
  25. 26 1月, 2016 1 次提交
  27. 16 12月, 2015 1 次提交
  29. 28 8月, 2015 1 次提交
  31. 11 6月, 2015 1 次提交
    • K
      tcp: add CDG congestion control · 2b0a8c9e
      由 Kenneth Klette Jonassen 提交于 
      CAIA Delay-Gradient (CDG) is a TCP congestion control that modifies
the TCP sender in order to [1]:

  o Use the delay gradient as a congestion signal.
  o Back off with an average probability that is independent of the RTT.
  o Coexist with flows that use loss-based congestion control, i.e.,
    flows that are unresponsive to the delay signal.
  o Tolerate packet loss unrelated to congestion. (Disabled by default.)

Its FreeBSD implementation was presented for the ICCRG in July 2012;
slides are available at http://www.ietf.org/proceedings/84/iccrg.html

Running the experiment scenarios in [1] suggests that our implementation
achieves more goodput compared with FreeBSD 10.0 senders, although it also
causes more queueing delay for a given backoff factor.

The loss tolerance heuristic is disabled by default due to safety concerns
for its use in the Internet [2, p. 45-46].

We use a variant of the Hybrid Slow start algorithm in tcp_cubic to reduce
the probability of slow start overshoot.

[1] D.A. Hayes and G. Armitage. "Revisiting TCP congestion control using
    delay gradients." In Networking 2011, pages 328-341. Springer, 2011.
[2] K.K. Jonassen. "Implementing CAIA Delay-Gradient in Linux."
    MSc thesis. Department of Informatics, University of Oslo, 2015.

Cc: Eric Dumazet <edumazet@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Stephen Hemminger <stephen@networkplumber.org>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: David Hayes <davihay@ifi.uio.no>
Cc: Andreas Petlund <apetlund@simula.no>
Cc: Dave Taht <dave.taht@bufferbloat.net>
Cc: Nicolas Kuhn <nicolas.kuhn@telecom-bretagne.eu>
Signed-off-by: NKenneth Klette Jonassen <kennetkl@ifi.uio.no>
Acked-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      2b0a8c9e
  33. 14 5月, 2015 1 次提交
  35. 06 11月, 2014 1 次提交
    • T
      net: Move fou_build_header into fou.c and refactor · 63487bab
      由 Tom Herbert 提交于 
      Move fou_build_header out of ip_tunnel.c and into fou.c splitting
it up into fou_build_header, gue_build_header, and fou_build_udp.
This allows for other users for TX of FOU or GUE. Change ip_tunnel_encap
to call fou_build_header or gue_build_header based on the tunnel
encapsulation type. Similarly, added fou_encap_hlen and gue_encap_hlen
functions which are called by ip_encap_hlen. New net/fou.h has
prototypes and defines for this.

Added NET_FOU_IP_TUNNELS configuration. When this is set, IP tunnels
can use FOU/GUE and fou module is also selected.
Signed-off-by: NTom Herbert <therbert@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      63487bab
  37. 07 10月, 2014 1 次提交
    • A
      openvswitch: fix a compilation error when CONFIG_INET is not setW! · 7c5df8fa
      由 Andy Zhou 提交于 
      Fix a openvswitch compilation error when CONFIG_INET is not set:

=====================================================
   In file included from include/net/geneve.h:4:0,
                       from net/openvswitch/flow_netlink.c:45:
		          include/net/udp_tunnel.h: In function 'udp_tunnel_handle_offloads':
			  >> include/net/udp_tunnel.h:100:2: error: implicit declaration of function 'iptunnel_handle_offloads' [-Werror=implicit-function-declaration]
			  >>      return iptunnel_handle_offloads(skb, udp_csum, type);
			  >>           ^
			  >>           >> include/net/udp_tunnel.h:100:2: warning: return makes pointer from integer without a cast
			  >>           >>    cc1: some warnings being treated as errors

=====================================================
Reported-by: Nkbuild test robot <fengguang.wu@intel.com>
Signed-off-by: NAndy Zhou <azhou@nicira.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      7c5df8fa
  39. 06 10月, 2014 1 次提交
  41. 29 9月, 2014 1 次提交
    • D
      net: tcp: add DCTCP congestion control algorithm · e3118e83
      由 Daniel Borkmann 提交于 
      This work adds the DataCenter TCP (DCTCP) congestion control
algorithm [1], which has been first published at SIGCOMM 2010 [2],
resp. follow-up analysis at SIGMETRICS 2011 [3] (and also, more
recently as an informational IETF draft available at [4]).

DCTCP is an enhancement to the TCP congestion control algorithm for
data center networks. Typical data center workloads are i.e.
i) partition/aggregate (queries; bursty, delay sensitive), ii) short
messages e.g. 50KB-1MB (for coordination and control state; delay
sensitive), and iii) large flows e.g. 1MB-100MB (data update;
throughput sensitive). DCTCP has therefore been designed for such
environments to provide/achieve the following three requirements:

  * High burst tolerance (incast due to partition/aggregate)
  * Low latency (short flows, queries)
  * High throughput (continuous data updates, large file
    transfers) with commodity, shallow buffered switches

The basic idea of its design consists of two fundamentals: i) on the
switch side, packets are being marked when its internal queue
length > threshold K (K is chosen so that a large enough headroom
for marked traffic is still available in the switch queue); ii) the
sender/host side maintains a moving average of the fraction of marked
packets, so each RTT, F is being updated as follows:

 F := X / Y, where X is # of marked ACKs, Y is total # of ACKs
 alpha := (1 - g) * alpha + g * F, where g is a smoothing constant

The resulting alpha (iow: probability that switch queue is congested)
is then being used in order to adaptively decrease the congestion
window W:

 W := (1 - (alpha / 2)) * W

The means for receiving marked packets resp. marking them on switch
side in DCTCP is the use of ECN.

RFC3168 describes a mechanism for using Explicit Congestion Notification
from the switch for early detection of congestion, rather than waiting
for segment loss to occur.

However, this method only detects the presence of congestion, not
the *extent*. In the presence of mild congestion, it reduces the TCP
congestion window too aggressively and unnecessarily affects the
throughput of long flows [4].

DCTCP, as mentioned, enhances Explicit Congestion Notification (ECN)
processing to estimate the fraction of bytes that encounter congestion,
rather than simply detecting that some congestion has occurred. DCTCP
then scales the TCP congestion window based on this estimate [4],
thus it can derive multibit feedback from the information present in
the single-bit sequence of marks in its control law. And thus act in
*proportion* to the extent of congestion, not its *presence*.

Switches therefore set the Congestion Experienced (CE) codepoint in
packets when internal queue lengths exceed threshold K. Resulting,
DCTCP delivers the same or better throughput than normal TCP, while
using 90% less buffer space.

It was found in [2] that DCTCP enables the applications to handle 10x
the current background traffic, without impacting foreground traffic.
Moreover, a 10x increase in foreground traffic did not cause any
timeouts, and thus largely eliminates TCP incast collapse problems.

The algorithm itself has already seen deployments in large production
data centers since then.

We did a long-term stress-test and analysis in a data center, short
summary of our TCP incast tests with iperf compared to cubic:

This test measured DCTCP throughput and latency and compared it with
CUBIC throughput and latency for an incast scenario. In this test, 19
senders sent at maximum rate to a single receiver. The receiver simply
ran iperf -s.

The senders ran iperf -c <receiver> -t 30. All senders started
simultaneously (using local clocks synchronized by ntp).

This test was repeated multiple times. Below shows the results from a
single test. Other tests are similar. (DCTCP results were extremely
consistent, CUBIC results show some variance induced by the TCP timeouts
that CUBIC encountered.)

For this test, we report statistics on the number of TCP timeouts,
flow throughput, and traffic latency.

1) Timeouts (total over all flows, and per flow summaries):

            CUBIC            DCTCP
  Total     3227             25
  Mean       169.842          1.316
  Median     183              1
  Max        207              5
  Min        123              0
  Stddev      28.991          1.600

Timeout data is taken by measuring the net change in netstat -s
"other TCP timeouts" reported. As a result, the timeout measurements
above are not restricted to the test traffic, and we believe that it
is likely that all of the "DCTCP timeouts" are actually timeouts for
non-test traffic. We report them nevertheless. CUBIC will also include
some non-test timeouts, but they are drawfed by bona fide test traffic
timeouts for CUBIC. Clearly DCTCP does an excellent job of preventing
TCP timeouts. DCTCP reduces timeouts by at least two orders of
magnitude and may well have eliminated them in this scenario.

2) Throughput (per flow in Mbps):

            CUBIC            DCTCP
  Mean      521.684          521.895
  Median    464              523
  Max       776              527
  Min       403              519
  Stddev    105.891            2.601
  Fairness    0.962            0.999

Throughput data was simply the average throughput for each flow
reported by iperf. By avoiding TCP timeouts, DCTCP is able to
achieve much better per-flow results. In CUBIC, many flows
experience TCP timeouts which makes flow throughput unpredictable and
unfair. DCTCP, on the other hand, provides very clean predictable
throughput without incurring TCP timeouts. Thus, the standard deviation
of CUBIC throughput is dramatically higher than the standard deviation
of DCTCP throughput.

Mean throughput is nearly identical because even though cubic flows
suffer TCP timeouts, other flows will step in and fill the unused
bandwidth. Note that this test is something of a best case scenario
for incast under CUBIC: it allows other flows to fill in for flows
experiencing a timeout. Under situations where the receiver is issuing
requests and then waiting for all flows to complete, flows cannot fill
in for timed out flows and throughput will drop dramatically.

3) Latency (in ms):

            CUBIC            DCTCP
  Mean      4.0088           0.04219
  Median    4.055            0.0395
  Max       4.2              0.085
  Min       3.32             0.028
  Stddev    0.1666           0.01064

Latency for each protocol was computed by running "ping -i 0.2
<receiver>" from a single sender to the receiver during the incast
test. For DCTCP, "ping -Q 0x6 -i 0.2 <receiver>" was used to ensure
that traffic traversed the DCTCP queue and was not dropped when the
queue size was greater than the marking threshold. The summary
statistics above are over all ping metrics measured between the single
sender, receiver pair.

The latency results for this test show a dramatic difference between
CUBIC and DCTCP. CUBIC intentionally overflows the switch buffer
which incurs the maximum queue latency (more buffer memory will lead
to high latency.) DCTCP, on the other hand, deliberately attempts to
keep queue occupancy low. The result is a two orders of magnitude
reduction of latency with DCTCP - even with a switch with relatively
little RAM. Switches with larger amounts of RAM will incur increasing
amounts of latency for CUBIC, but not for DCTCP.

4) Convergence and stability test:

This test measured the time that DCTCP took to fairly redistribute
bandwidth when a new flow commences. It also measured DCTCP's ability
to remain stable at a fair bandwidth distribution. DCTCP is compared
with CUBIC for this test.

At the commencement of this test, a single flow is sending at maximum
rate (near 10 Gbps) to a single receiver. One second after that first
flow commences, a new flow from a distinct server begins sending to
the same receiver as the first flow. After the second flow has sent
data for 10 seconds, the second flow is terminated. The first flow
sends for an additional second. Ideally, the bandwidth would be evenly
shared as soon as the second flow starts, and recover as soon as it
stops.

The results of this test are shown below. Note that the flow bandwidth
for the two flows was measured near the same time, but not
simultaneously.

DCTCP performs nearly perfectly within the measurement limitations
of this test: bandwidth is quickly distributed fairly between the two
flows, remains stable throughout the duration of the test, and
recovers quickly. CUBIC, in contrast, is slow to divide the bandwidth
fairly, and has trouble remaining stable.

  CUBIC                      DCTCP

  Seconds  Flow 1  Flow 2    Seconds  Flow 1  Flow 2
   0       9.93    0          0       9.92    0
   0.5     9.87    0          0.5     9.86    0
   1       8.73    2.25       1       6.46    4.88
   1.5     7.29    2.8        1.5     4.9     4.99
   2       6.96    3.1        2       4.92    4.94
   2.5     6.67    3.34       2.5     4.93    5
   3       6.39    3.57       3       4.92    4.99
   3.5     6.24    3.75       3.5     4.94    4.74
   4       6       3.94       4       5.34    4.71
   4.5     5.88    4.09       4.5     4.99    4.97
   5       5.27    4.98       5       4.83    5.01
   5.5     4.93    5.04       5.5     4.89    4.99
   6       4.9     4.99       6       4.92    5.04
   6.5     4.93    5.1        6.5     4.91    4.97
   7       4.28    5.8        7       4.97    4.97
   7.5     4.62    4.91       7.5     4.99    4.82
   8       5.05    4.45       8       5.16    4.76
   8.5     5.93    4.09       8.5     4.94    4.98
   9       5.73    4.2        9       4.92    5.02
   9.5     5.62    4.32       9.5     4.87    5.03
  10       6.12    3.2       10       4.91    5.01
  10.5     6.91    3.11      10.5     4.87    5.04
  11       8.48    0         11       8.49    4.94
  11.5     9.87    0         11.5     9.9     0

SYN/ACK ECT test:

This test demonstrates the importance of ECT on SYN and SYN-ACK packets
by measuring the connection probability in the presence of competing
flows for a DCTCP connection attempt *without* ECT in the SYN packet.
The test was repeated five times for each number of competing flows.

              Competing Flows  1 |    2 |    4 |    8 |   16
                               ------------------------------
Mean Connection Probability    1 | 0.67 | 0.45 | 0.28 |    0
Median Connection Probability  1 | 0.65 | 0.45 | 0.25 |    0

As the number of competing flows moves beyond 1, the connection
probability drops rapidly.

Enabling DCTCP with this patch requires the following steps:

DCTCP must be running both on the sender and receiver side in your
data center, i.e.:

  sysctl -w net.ipv4.tcp_congestion_control=dctcp

Also, ECN functionality must be enabled on all switches in your
data center for DCTCP to work. The default ECN marking threshold (K)
heuristic on the switch for DCTCP is e.g., 20 packets (30KB) at
1Gbps, and 65 packets (~100KB) at 10Gbps (K > 1/7 * C * RTT, [4]).

In above tests, for each switch port, traffic was segregated into two
queues. For any packet with a DSCP of 0x01 - or equivalently a TOS of
0x04 - the packet was placed into the DCTCP queue. All other packets
were placed into the default drop-tail queue. For the DCTCP queue,
RED/ECN marking was enabled, here, with a marking threshold of 75 KB.
More details however, we refer you to the paper [2] under section 3).

There are no code changes required to applications running in user
space. DCTCP has been implemented in full *isolation* of the rest of
the TCP code as its own congestion control module, so that it can run
without a need to expose code to the core of the TCP stack, and thus
nothing changes for non-DCTCP users.

Changes in the CA framework code are minimal, and DCTCP algorithm
operates on mechanisms that are already available in most Silicon.
The gain (dctcp_shift_g) is currently a fixed constant (1/16) from
the paper, but we leave the option that it can be chosen carefully
to a different value by the user.

In case DCTCP is being used and ECN support on peer site is off,
DCTCP falls back after 3WHS to operate in normal TCP Reno mode.

ss {-4,-6} -t -i diag interface:

  ... dctcp wscale:7,7 rto:203 rtt:2.349/0.026 mss:1448 cwnd:2054
  ssthresh:1102 ce_state 0 alpha 15 ab_ecn 0 ab_tot 735584
  send 10129.2Mbps pacing_rate 20254.1Mbps unacked:1822 retrans:0/15
  reordering:101 rcv_space:29200

  ... dctcp-reno wscale:7,7 rto:201 rtt:0.711/1.327 ato:40 mss:1448
  cwnd:10 ssthresh:1102 fallback_mode send 162.9Mbps pacing_rate
  325.5Mbps rcv_rtt:1.5 rcv_space:29200

More information about DCTCP can be found in [1-4].

  [1] http://simula.stanford.edu/~alizade/Site/DCTCP.html
  [2] http://simula.stanford.edu/~alizade/Site/DCTCP_files/dctcp-final.pdf
  [3] http://simula.stanford.edu/~alizade/Site/DCTCP_files/dctcp_analysis-full.pdf
  [4] http://tools.ietf.org/html/draft-bensley-tcpm-dctcp-00

Joint work with Florian Westphal and Glenn Judd.
Signed-off-by: NDaniel Borkmann <dborkman@redhat.com>
Signed-off-by: NFlorian Westphal <fw@strlen.de>
Signed-off-by: NGlenn Judd <glenn.judd@morganstanley.com>
Acked-by: NStephen Hemminger <stephen@networkplumber.org>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      e3118e83
  43. 20 9月, 2014 1 次提交
    • T
      fou: Support for foo-over-udp RX path · 23461551
      由 Tom Herbert 提交于 
      This patch provides a receive path for foo-over-udp. This allows
direct encapsulation of IP protocols over UDP. The bound destination
port is used to map to an IP protocol, and the XFRM framework
(udp_encap_rcv) is used to receive encapsulated packets. Upon
reception, the encapsulation header is logically removed (pointer
to transport header is advanced) and the packet is reinjected into
the receive path with the IP protocol indicated by the mapping.

Netlink is used to configure FOU ports. The configuration information
includes the port number to bind to and the IP protocol corresponding
to that port.

This should support GRE/UDP
(http://tools.ietf.org/html/draft-yong-tsvwg-gre-in-udp-encap-02),
as will as the other IP tunneling protocols (IPIP, SIT).
Signed-off-by: NTom Herbert <therbert@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      23461551
  45. 15 7月, 2014 1 次提交
  47. 04 9月, 2013 1 次提交
    • T
      net: neighbour: Remove CONFIG_ARPD · 3e25c65e
      由 Tim Gardner 提交于 
      This config option is superfluous in that it only guards a call
to neigh_app_ns(). Enabling CONFIG_ARPD by default has no
change in behavior. There will now be call to __neigh_notify()
for each ARP resolution, which has no impact unless there is a
user space daemon waiting to receive the notification, i.e.,
the case for which CONFIG_ARPD was designed anyways.
Suggested-by: NEric W. Biederman <ebiederm@xmission.com>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: Alexey Kuznetsov <kuznet@ms2.inr.ac.ru>
Cc: James Morris <jmorris@namei.org>
Cc: Hideaki YOSHIFUJI <yoshfuji@linux-ipv6.org>
Cc: Patrick McHardy <kaber@trash.net>
Cc: "Eric W. Biederman" <ebiederm@xmission.com>
Cc: Gao feng <gaofeng@cn.fujitsu.com>
Cc: Joe Perches <joe@perches.com>
Cc: Veaceslav Falico <vfalico@redhat.com>
Signed-off-by: NTim Gardner <tim.gardner@canonical.com>
Reviewed-by: N"Eric W. Biederman" <ebiederm@xmission.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      3e25c65e
  49. 05 6月, 2013 1 次提交
  51. 27 3月, 2013 3 次提交
  53. 12 1月, 2013 1 次提交
    • K
      net/ipv4: remove depends on CONFIG_EXPERIMENTAL · 44fbe920
      由 Kees Cook 提交于 
      The CONFIG_EXPERIMENTAL config item has not carried much meaning for a
while now and is almost always enabled by default. As agreed during the
Linux kernel summit, remove it from any "depends on" lines in Kconfigs.

CC: "David S. Miller" <davem@davemloft.net>
CC: Alexey Kuznetsov <kuznet@ms2.inr.ac.ru>
CC: James Morris <jmorris@namei.org>
CC: Hideaki YOSHIFUJI <yoshfuji@linux-ipv6.org>
CC: Patrick McHardy <kaber@trash.net>
Signed-off-by: NKees Cook <keescook@chromium.org>
Acked-by: NDavid S. Miller <davem@davemloft.net>
      44fbe920
  55. 19 7月, 2012 1 次提交
  57. 16 5月, 2012 2 次提交
  59. 08 2月, 2012 1 次提交
  61. 08 1月, 2012 1 次提交
  63. 11 12月, 2011 1 次提交
  65. 10 12月, 2011 1 次提交
  67. 31 10月, 2011 1 次提交
    • P
      Kconfig: remove a few puzzling comments · bfc994b5
      由 Paul Bolle 提交于 
      These comments mention CONFIG options that do not exist: not as a symbol
in a Kconfig file (without the CONFIG_ prefix) and neither as a symbol
(with that prefix) in the code.

There's one reference to XSCALE_PMU_TIMER as a negative dependency.
But XSCALE_PMU_TIMER is never defined (CONFIG_XSCALE_PMU_TIMER is
also unused in the code). It shows up with type "unknown" if you search
for it in menuconfig. Apparently a negative dependency on an unknown
symbol is always true. That negative dependency can be removed too.
Signed-off-by: NPaul Bolle <pebolle@tiscali.nl>
Signed-off-by: NJiri Kosina <jkosina@suse.cz>
      bfc994b5
  69. 02 2月, 2011 1 次提交
    • D
      ipv4: Remove fib_hash. · 3630b7c0
      由 David S. Miller 提交于 
      The time has finally come to remove the hash based routing table
implementation in ipv4.

FIB Trie is mature, well tested, and I've done an audit of it's code
to confirm that it implements insert, delete, and lookup with the same
identical semantics as fib_hash did.

If there are any semantic differences found in fib_trie, we should
simply fix them.

I've placed the trie statistic config option under advanced router
configuration.
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
Acked-by: NStephen Hemminger <shemminger@vyatta.com>
      3630b7c0
  71. 14 1月, 2011 1 次提交
    • P
      netfilter: fix Kconfig dependencies · c7066f70
      由 Patrick McHardy 提交于 
      Fix dependencies of netfilter realm match: it depends on NET_CLS_ROUTE,
which itself depends on NET_SCHED; this dependency is missing from netfilter.

Since matching on realms is also useful without having NET_SCHED enabled and
the option really only controls whether the tclassid member is included in
route and dst entries, rename the config option to IP_ROUTE_CLASSID and move
it outside of traffic scheduling context to get rid of the NET_SCHED dependeny.
Reported-by: NVladis Kletnieks <Valdis.Kletnieks@vt.edu>
Signed-off-by: NPatrick McHardy <kaber@trash.net>
      c7066f70
  73. 16 11月, 2010 1 次提交
    • M
      Docs/Kconfig: Update: osdl.org -> linuxfoundation.org · c996d8b9
      由 Michael Witten 提交于 
      Some of the documentation refers to web pages under
the domain `osdl.org'. However, `osdl.org' now
redirects to `linuxfoundation.org'.

Rather than rely on redirections, this patch updates
the addresses appropriately; for the most part, only
documentation that is meant to be current has been
updated.

The patch should be pretty quick to scan and check;
each new web-page url was gotten by trying out the
original URL in a browser and then simply copying the
the redirected URL (formatting as necessary).

There is some conflict as to which one of these domain
names is preferred:

  linuxfoundation.org
  linux-foundation.org

So, I wrote:

  info@linuxfoundation.org

and got this reply:

  Message-ID: <4CE17EE6.9040807@linuxfoundation.org>
  Date: Mon, 15 Nov 2010 10:41:42 -0800
  From: David Ames <david@linuxfoundation.org>

  ...

  linuxfoundation.org is preferred. The canonical name for our web site is
  www.linuxfoundation.org. Our list site is actually
  lists.linux-foundation.org.

  Regarding email linuxfoundation.org is preferred there are a few people
  who choose to use linux-foundation.org for their own reasons.

Consequently, I used `linuxfoundation.org' for web pages and
`lists.linux-foundation.org' for mailing-list web pages and email addresses;
the only personal email address I updated from `@osdl.org' was that of
Andrew Morton, who prefers `linux-foundation.org' according `git log'.
Signed-off-by: NMichael Witten <mfwitten@gmail.com>
Signed-off-by: NJiri Kosina <jkosina@suse.cz>
      c996d8b9