<feed xmlns='http://www.w3.org/2005/Atom'>
<title>linux-stable.git/net/ipv4/Makefile, branch linux-4.6.y</title>
<subtitle>Linux kernel stable tree</subtitle>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/'/>
<entry>
<title>ipv4: Remove inet_lro library</title>
<updated>2016-02-17T21:15:46+00:00</updated>
<author>
<name>Ben Hutchings</name>
<email>ben@decadent.org.uk</email>
</author>
<published>2016-02-15T21:25:57+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=7bbf3cae65b6e438bf52033b63fdce4a86e89e17'/>
<id>7bbf3cae65b6e438bf52033b63fdce4a86e89e17</id>
<content type='text'>
There are no longer any in-tree drivers that use it.

Signed-off-by: Ben Hutchings &lt;ben@decadent.org.uk&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
There are no longer any in-tree drivers that use it.

Signed-off-by: Ben Hutchings &lt;ben@decadent.org.uk&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>net: drop tcp_memcontrol.c</title>
<updated>2016-01-21T01:09:18+00:00</updated>
<author>
<name>Vladimir Davydov</name>
<email>vdavydov@virtuozzo.com</email>
</author>
<published>2016-01-20T23:02:44+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=d55f90bfab40e3b5db323711d28186ff09461692'/>
<id>d55f90bfab40e3b5db323711d28186ff09461692</id>
<content type='text'>
tcp_memcontrol.c only contains legacy memory.tcp.kmem.* file definitions
and mem_cgroup-&gt;tcp_mem init/destroy stuff.  This doesn't belong to
network subsys.  Let's move it to memcontrol.c.  This also allows us to
reuse generic code for handling legacy memcg files.

Signed-off-by: Vladimir Davydov &lt;vdavydov@virtuozzo.com&gt;
Acked-by: Johannes Weiner &lt;hannes@cmpxchg.org&gt;
Cc: "David S. Miller" &lt;davem@davemloft.net&gt;
Acked-by: Michal Hocko &lt;mhocko@suse.com&gt;
Signed-off-by: Andrew Morton &lt;akpm@linux-foundation.org&gt;
Signed-off-by: Linus Torvalds &lt;torvalds@linux-foundation.org&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
tcp_memcontrol.c only contains legacy memory.tcp.kmem.* file definitions
and mem_cgroup-&gt;tcp_mem init/destroy stuff.  This doesn't belong to
network subsys.  Let's move it to memcontrol.c.  This also allows us to
reuse generic code for handling legacy memcg files.

Signed-off-by: Vladimir Davydov &lt;vdavydov@virtuozzo.com&gt;
Acked-by: Johannes Weiner &lt;hannes@cmpxchg.org&gt;
Cc: "David S. Miller" &lt;davem@davemloft.net&gt;
Acked-by: Michal Hocko &lt;mhocko@suse.com&gt;
Signed-off-by: Andrew Morton &lt;akpm@linux-foundation.org&gt;
Signed-off-by: Linus Torvalds &lt;torvalds@linux-foundation.org&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>mm: memcontrol: introduce CONFIG_MEMCG_LEGACY_KMEM</title>
<updated>2016-01-21T01:09:18+00:00</updated>
<author>
<name>Johannes Weiner</name>
<email>hannes@cmpxchg.org</email>
</author>
<published>2016-01-20T23:02:41+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=489c2a20a414351fe0813a727c34600c0f7292ae'/>
<id>489c2a20a414351fe0813a727c34600c0f7292ae</id>
<content type='text'>
Let the user know that CONFIG_MEMCG_KMEM does not apply to the cgroup2
interface. This also makes legacy-only code sections stand out better.

[arnd@arndb.de: mm: memcontrol: only manage socket pressure for CONFIG_INET]
Signed-off-by: Johannes Weiner &lt;hannes@cmpxchg.org&gt;
Cc: Michal Hocko &lt;mhocko@suse.cz&gt;
Cc: Tejun Heo &lt;tj@kernel.org&gt;
Acked-by: Vladimir Davydov &lt;vdavydov@virtuozzo.com&gt;
Signed-off-by: Arnd Bergmann &lt;arnd@arndb.de&gt;
Signed-off-by: Andrew Morton &lt;akpm@linux-foundation.org&gt;
Signed-off-by: Linus Torvalds &lt;torvalds@linux-foundation.org&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
Let the user know that CONFIG_MEMCG_KMEM does not apply to the cgroup2
interface. This also makes legacy-only code sections stand out better.

[arnd@arndb.de: mm: memcontrol: only manage socket pressure for CONFIG_INET]
Signed-off-by: Johannes Weiner &lt;hannes@cmpxchg.org&gt;
Cc: Michal Hocko &lt;mhocko@suse.cz&gt;
Cc: Tejun Heo &lt;tj@kernel.org&gt;
Acked-by: Vladimir Davydov &lt;vdavydov@virtuozzo.com&gt;
Signed-off-by: Arnd Bergmann &lt;arnd@arndb.de&gt;
Signed-off-by: Andrew Morton &lt;akpm@linux-foundation.org&gt;
Signed-off-by: Linus Torvalds &lt;torvalds@linux-foundation.org&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: track the packet timings in RACK</title>
<updated>2015-10-21T14:00:48+00:00</updated>
<author>
<name>Yuchung Cheng</name>
<email>ycheng@google.com</email>
</author>
<published>2015-10-17T04:57:46+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=659a8ad56f490279f0efee43a62ffa1ac914a4e0'/>
<id>659a8ad56f490279f0efee43a62ffa1ac914a4e0</id>
<content type='text'>
This patch is the first half of the RACK loss recovery.

RACK loss recovery uses the notion of time instead
of packet sequence (FACK) or counts (dupthresh). It's inspired by the
previous FACK heuristic in tcp_mark_lost_retrans(): when a limited
transmit (new data packet) is sacked, then current retransmitted
sequence below the newly sacked sequence must been lost,
since at least one round trip time has elapsed.

But it has several limitations:
1) can't detect tail drops since it depends on limited transmit
2) is disabled upon reordering (assumes no reordering)
3) only enabled in fast recovery ut not timeout recovery

RACK (Recently ACK) addresses these limitations with the notion
of time instead: a packet P1 is lost if a later packet P2 is s/acked,
as at least one round trip has passed.

Since RACK cares about the time sequence instead of the data sequence
of packets, it can detect tail drops when later retransmission is
s/acked while FACK or dupthresh can't. For reordering RACK uses a
dynamically adjusted reordering window ("reo_wnd") to reduce false
positives on ever (small) degree of reordering.

This patch implements tcp_advanced_rack() which tracks the
most recent transmission time among the packets that have been
delivered (ACKed or SACKed) in tp-&gt;rack.mstamp. This timestamp
is the key to determine which packet has been lost.

Consider an example that the sender sends six packets:
T1: P1 (lost)
T2: P2
T3: P3
T4: P4
T100: sack of P2. rack.mstamp = T2
T101: retransmit P1
T102: sack of P2,P3,P4. rack.mstamp = T4
T205: ACK of P4 since the hole is repaired. rack.mstamp = T101

We need to be careful about spurious retransmission because it may
falsely advance tp-&gt;rack.mstamp by an RTT or an RTO, causing RACK
to falsely mark all packets lost, just like a spurious timeout.

We identify spurious retransmission by the ACK's TS echo value.
If TS option is not applicable but the retransmission is acknowledged
less than min-RTT ago, it is likely to be spurious. We refrain from
using the transmission time of these spurious retransmissions.

The second half is implemented in the next patch that marks packet
lost using RACK timestamp.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
This patch is the first half of the RACK loss recovery.

RACK loss recovery uses the notion of time instead
of packet sequence (FACK) or counts (dupthresh). It's inspired by the
previous FACK heuristic in tcp_mark_lost_retrans(): when a limited
transmit (new data packet) is sacked, then current retransmitted
sequence below the newly sacked sequence must been lost,
since at least one round trip time has elapsed.

But it has several limitations:
1) can't detect tail drops since it depends on limited transmit
2) is disabled upon reordering (assumes no reordering)
3) only enabled in fast recovery ut not timeout recovery

RACK (Recently ACK) addresses these limitations with the notion
of time instead: a packet P1 is lost if a later packet P2 is s/acked,
as at least one round trip has passed.

Since RACK cares about the time sequence instead of the data sequence
of packets, it can detect tail drops when later retransmission is
s/acked while FACK or dupthresh can't. For reordering RACK uses a
dynamically adjusted reordering window ("reo_wnd") to reduce false
positives on ever (small) degree of reordering.

This patch implements tcp_advanced_rack() which tracks the
most recent transmission time among the packets that have been
delivered (ACKed or SACKed) in tp-&gt;rack.mstamp. This timestamp
is the key to determine which packet has been lost.

Consider an example that the sender sends six packets:
T1: P1 (lost)
T2: P2
T3: P3
T4: P4
T100: sack of P2. rack.mstamp = T2
T101: retransmit P1
T102: sack of P2,P3,P4. rack.mstamp = T4
T205: ACK of P4 since the hole is repaired. rack.mstamp = T101

We need to be careful about spurious retransmission because it may
falsely advance tp-&gt;rack.mstamp by an RTT or an RTO, causing RACK
to falsely mark all packets lost, just like a spurious timeout.

We identify spurious retransmission by the ACK's TS echo value.
If TS option is not applicable but the retransmission is acknowledged
less than min-RTT ago, it is likely to be spurious. We refrain from
using the transmission time of these spurious retransmissions.

The second half is implemented in the next patch that marks packet
lost using RACK timestamp.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>geneve: Consolidate Geneve functionality in single module.</title>
<updated>2015-08-27T22:42:48+00:00</updated>
<author>
<name>Pravin B Shelar</name>
<email>pshelar@nicira.com</email>
</author>
<published>2015-08-27T06:46:54+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=371bd1061d29562e6423435073623add8c475ee2'/>
<id>371bd1061d29562e6423435073623add8c475ee2</id>
<content type='text'>
geneve_core module handles send and receive functionality.
This way OVS could use the Geneve API. Now with use of
tunnel meatadata mode OVS can directly use Geneve netdevice.
So there is no need for separate module for Geneve. Following
patch consolidates Geneve protocol processing in single module.

Signed-off-by: Pravin B Shelar &lt;pshelar@nicira.com&gt;
Reviewed-by: Jesse Gross &lt;jesse@nicira.com&gt;
Acked-by: John W. Linville &lt;linville@tuxdriver.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
geneve_core module handles send and receive functionality.
This way OVS could use the Geneve API. Now with use of
tunnel meatadata mode OVS can directly use Geneve netdevice.
So there is no need for separate module for Geneve. Following
patch consolidates Geneve protocol processing in single module.

Signed-off-by: Pravin B Shelar &lt;pshelar@nicira.com&gt;
Reviewed-by: Jesse Gross &lt;jesse@nicira.com&gt;
Acked-by: John W. Linville &lt;linville@tuxdriver.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: add CDG congestion control</title>
<updated>2015-06-11T07:09:12+00:00</updated>
<author>
<name>Kenneth Klette Jonassen</name>
<email>kennetkl@ifi.uio.no</email>
</author>
<published>2015-06-10T17:08:17+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=2b0a8c9eee81882fc0001ccf6d9af62cdc682f9e'/>
<id>2b0a8c9eee81882fc0001ccf6d9af62cdc682f9e</id>
<content type='text'>
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 &lt;edumazet@google.com&gt;
Cc: Yuchung Cheng &lt;ycheng@google.com&gt;
Cc: Stephen Hemminger &lt;stephen@networkplumber.org&gt;
Cc: Neal Cardwell &lt;ncardwell@google.com&gt;
Cc: David Hayes &lt;davihay@ifi.uio.no&gt;
Cc: Andreas Petlund &lt;apetlund@simula.no&gt;
Cc: Dave Taht &lt;dave.taht@bufferbloat.net&gt;
Cc: Nicolas Kuhn &lt;nicolas.kuhn@telecom-bretagne.eu&gt;
Signed-off-by: Kenneth Klette Jonassen &lt;kennetkl@ifi.uio.no&gt;
Acked-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
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 &lt;edumazet@google.com&gt;
Cc: Yuchung Cheng &lt;ycheng@google.com&gt;
Cc: Stephen Hemminger &lt;stephen@networkplumber.org&gt;
Cc: Neal Cardwell &lt;ncardwell@google.com&gt;
Cc: David Hayes &lt;davihay@ifi.uio.no&gt;
Cc: Andreas Petlund &lt;apetlund@simula.no&gt;
Cc: Dave Taht &lt;dave.taht@bufferbloat.net&gt;
Cc: Nicolas Kuhn &lt;nicolas.kuhn@telecom-bretagne.eu&gt;
Signed-off-by: Kenneth Klette Jonassen &lt;kennetkl@ifi.uio.no&gt;
Acked-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>geneve: Rename support library as geneve_core</title>
<updated>2015-05-13T19:59:13+00:00</updated>
<author>
<name>John W. Linville</name>
<email>linville@tuxdriver.com</email>
</author>
<published>2015-05-13T16:57:28+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=11e1fa46b43216458e0f67f1f0b257586c5d8e5c'/>
<id>11e1fa46b43216458e0f67f1f0b257586c5d8e5c</id>
<content type='text'>
net/ipv4/geneve.c -&gt; net/ipv4/geneve_core.c

This name better reflects the purpose of the module.

Signed-off-by: John W. Linville &lt;linville@tuxdriver.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
net/ipv4/geneve.c -&gt; net/ipv4/geneve_core.c

This name better reflects the purpose of the module.

Signed-off-by: John W. Linville &lt;linville@tuxdriver.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>net: Add Geneve tunneling protocol driver</title>
<updated>2014-10-06T04:32:20+00:00</updated>
<author>
<name>Andy Zhou</name>
<email>azhou@nicira.com</email>
</author>
<published>2014-10-03T22:35:28+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=0b5e8b8eeae40bae6ad7c7e91c97c3c0d0e57882'/>
<id>0b5e8b8eeae40bae6ad7c7e91c97c3c0d0e57882</id>
<content type='text'>
This adds a device level support for Geneve -- Generic Network
Virtualization Encapsulation. The protocol is documented at
http://tools.ietf.org/html/draft-gross-geneve-01

Only protocol layer Geneve support is provided by this driver.
Openvswitch can be used for configuring, set up and tear down
functional Geneve tunnels.

Signed-off-by: Jesse Gross &lt;jesse@nicira.com&gt;
Signed-off-by: Andy Zhou &lt;azhou@nicira.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
This adds a device level support for Geneve -- Generic Network
Virtualization Encapsulation. The protocol is documented at
http://tools.ietf.org/html/draft-gross-geneve-01

Only protocol layer Geneve support is provided by this driver.
Openvswitch can be used for configuring, set up and tear down
functional Geneve tunnels.

Signed-off-by: Jesse Gross &lt;jesse@nicira.com&gt;
Signed-off-by: Andy Zhou &lt;azhou@nicira.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>net: tcp: add DCTCP congestion control algorithm</title>
<updated>2014-09-29T04:13:10+00:00</updated>
<author>
<name>Daniel Borkmann</name>
<email>dborkman@redhat.com</email>
</author>
<published>2014-09-26T20:37:36+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=e3118e8359bb7c59555aca60c725106e6d78c5ce'/>
<id>e3118e8359bb7c59555aca60c725106e6d78c5ce</id>
<content type='text'>
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 &gt; 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 &lt;receiver&gt; -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
&lt;receiver&gt;" from a single sender to the receiver during the incast
test. For DCTCP, "ping -Q 0x6 -i 0.2 &lt;receiver&gt;" 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 &gt; 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: Daniel Borkmann &lt;dborkman@redhat.com&gt;
Signed-off-by: Florian Westphal &lt;fw@strlen.de&gt;
Signed-off-by: Glenn Judd &lt;glenn.judd@morganstanley.com&gt;
Acked-by: Stephen Hemminger &lt;stephen@networkplumber.org&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
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 &gt; 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 &lt;receiver&gt; -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
&lt;receiver&gt;" from a single sender to the receiver during the incast
test. For DCTCP, "ping -Q 0x6 -i 0.2 &lt;receiver&gt;" 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 &gt; 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: Daniel Borkmann &lt;dborkman@redhat.com&gt;
Signed-off-by: Florian Westphal &lt;fw@strlen.de&gt;
Signed-off-by: Glenn Judd &lt;glenn.judd@morganstanley.com&gt;
Acked-by: Stephen Hemminger &lt;stephen@networkplumber.org&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>fou: Support for foo-over-udp RX path</title>
<updated>2014-09-19T21:15:31+00:00</updated>
<author>
<name>Tom Herbert</name>
<email>therbert@google.com</email>
</author>
<published>2014-09-17T19:25:56+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=23461551c00628c3f3fe9cf837bf53cf8f212b63'/>
<id>23461551c00628c3f3fe9cf837bf53cf8f212b63</id>
<content type='text'>
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: Tom Herbert &lt;therbert@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
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: Tom Herbert &lt;therbert@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
</feed>
