<feed xmlns='http://www.w3.org/2005/Atom'>
<title>linux-stable.git/net/ipv4/tcp_input.c, branch linux-3.11.y</title>
<subtitle>Linux kernel stable tree</subtitle>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/'/>
<entry>
<title>tcp: fix incorrect ca_state in tail loss probe</title>
<updated>2013-11-04T12:35:11+00:00</updated>
<author>
<name>Yuchung Cheng</name>
<email>ycheng@google.com</email>
</author>
<published>2013-10-12T17:16:27+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=39fc1ed53e50b5a8b6f8a9da7867d214760be19d'/>
<id>39fc1ed53e50b5a8b6f8a9da7867d214760be19d</id>
<content type='text'>
[ Upstream commit 031afe4990a7c9dbff41a3a742c44d3e740ea0a1 ]

On receiving an ACK that covers the loss probe sequence, TLP
immediately sets the congestion state to Open, even though some packets
are not recovered and retransmisssion are on the way.  The later ACks
may trigger a WARN_ON check in step D of tcp_fastretrans_alert(), e.g.,
https://bugzilla.redhat.com/show_bug.cgi?id=989251

The fix is to follow the similar procedure in recovery by calling
tcp_try_keep_open(). The sender switches to Open state if no packets
are retransmissted. Otherwise it goes to Disorder and let subsequent
ACKs move the state to Recovery or Open.

Reported-By: Michael Sterrett &lt;michael@sterretts.net&gt;
Tested-By: Dormando &lt;dormando@rydia.net&gt;
Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
[ Upstream commit 031afe4990a7c9dbff41a3a742c44d3e740ea0a1 ]

On receiving an ACK that covers the loss probe sequence, TLP
immediately sets the congestion state to Open, even though some packets
are not recovered and retransmisssion are on the way.  The later ACks
may trigger a WARN_ON check in step D of tcp_fastretrans_alert(), e.g.,
https://bugzilla.redhat.com/show_bug.cgi?id=989251

The fix is to follow the similar procedure in recovery by calling
tcp_try_keep_open(). The sender switches to Open state if no packets
are retransmissted. Otherwise it goes to Disorder and let subsequent
ACKs move the state to Recovery or Open.

Reported-By: Michael Sterrett &lt;michael@sterretts.net&gt;
Tested-By: Dormando &lt;dormando@rydia.net&gt;
Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: do not forget FIN in tcp_shifted_skb()</title>
<updated>2013-11-04T12:35:11+00:00</updated>
<author>
<name>Eric Dumazet</name>
<email>edumazet@google.com</email>
</author>
<published>2013-10-04T17:31:41+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=dc0791aee672ca2fb8bd61e32a66f77d01bcafc8'/>
<id>dc0791aee672ca2fb8bd61e32a66f77d01bcafc8</id>
<content type='text'>
[ Upstream commit 5e8a402f831dbe7ee831340a91439e46f0d38acd ]

Yuchung found following problem :

 There are bugs in the SACK processing code, merging part in
 tcp_shift_skb_data(), that incorrectly resets or ignores the sacked
 skbs FIN flag. When a receiver first SACK the FIN sequence, and later
 throw away ofo queue (e.g., sack-reneging), the sender will stop
 retransmitting the FIN flag, and hangs forever.

Following packetdrill test can be used to reproduce the bug.

$ cat sack-merge-bug.pkt
`sysctl -q net.ipv4.tcp_fack=0`

// Establish a connection and send 10 MSS.
0.000 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3
+.000 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0
+.000 bind(3, ..., ...) = 0
+.000 listen(3, 1) = 0

+.050 &lt; S 0:0(0) win 32792 &lt;mss 1000,sackOK,nop,nop,nop,wscale 7&gt;
+.000 &gt; S. 0:0(0) ack 1 &lt;mss 1460,nop,nop,sackOK,nop,wscale 6&gt;
+.001 &lt; . 1:1(0) ack 1 win 1024
+.000 accept(3, ..., ...) = 4

+.100 write(4, ..., 12000) = 12000
+.000 shutdown(4, SHUT_WR) = 0
+.000 &gt; . 1:10001(10000) ack 1
+.050 &lt; . 1:1(0) ack 2001 win 257
+.000 &gt; FP. 10001:12001(2000) ack 1
+.050 &lt; . 1:1(0) ack 2001 win 257 &lt;sack 10001:11001,nop,nop&gt;
+.050 &lt; . 1:1(0) ack 2001 win 257 &lt;sack 10001:12002,nop,nop&gt;
// SACK reneg
+.050 &lt; . 1:1(0) ack 12001 win 257
+0 %{ print "unacked: ",tcpi_unacked }%
+5 %{ print "" }%

First, a typo inverted left/right of one OR operation, then
code forgot to advance end_seq if the merged skb carried FIN.

Bug was added in 2.6.29 by commit 832d11c5cd076ab
("tcp: Try to restore large SKBs while SACK processing")

Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Cc: Ilpo Järvinen &lt;ilpo.jarvinen@helsinki.fi&gt;
Acked-by: Ilpo Järvinen &lt;ilpo.jarvinen@helsinki.fi&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
[ Upstream commit 5e8a402f831dbe7ee831340a91439e46f0d38acd ]

Yuchung found following problem :

 There are bugs in the SACK processing code, merging part in
 tcp_shift_skb_data(), that incorrectly resets or ignores the sacked
 skbs FIN flag. When a receiver first SACK the FIN sequence, and later
 throw away ofo queue (e.g., sack-reneging), the sender will stop
 retransmitting the FIN flag, and hangs forever.

Following packetdrill test can be used to reproduce the bug.

$ cat sack-merge-bug.pkt
`sysctl -q net.ipv4.tcp_fack=0`

// Establish a connection and send 10 MSS.
0.000 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3
+.000 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0
+.000 bind(3, ..., ...) = 0
+.000 listen(3, 1) = 0

+.050 &lt; S 0:0(0) win 32792 &lt;mss 1000,sackOK,nop,nop,nop,wscale 7&gt;
+.000 &gt; S. 0:0(0) ack 1 &lt;mss 1460,nop,nop,sackOK,nop,wscale 6&gt;
+.001 &lt; . 1:1(0) ack 1 win 1024
+.000 accept(3, ..., ...) = 4

+.100 write(4, ..., 12000) = 12000
+.000 shutdown(4, SHUT_WR) = 0
+.000 &gt; . 1:10001(10000) ack 1
+.050 &lt; . 1:1(0) ack 2001 win 257
+.000 &gt; FP. 10001:12001(2000) ack 1
+.050 &lt; . 1:1(0) ack 2001 win 257 &lt;sack 10001:11001,nop,nop&gt;
+.050 &lt; . 1:1(0) ack 2001 win 257 &lt;sack 10001:12002,nop,nop&gt;
// SACK reneg
+.050 &lt; . 1:1(0) ack 12001 win 257
+0 %{ print "unacked: ",tcpi_unacked }%
+5 %{ print "" }%

First, a typo inverted left/right of one OR operation, then
code forgot to advance end_seq if the merged skb carried FIN.

Bug was added in 2.6.29 by commit 832d11c5cd076ab
("tcp: Try to restore large SKBs while SACK processing")

Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Cc: Ilpo Järvinen &lt;ilpo.jarvinen@helsinki.fi&gt;
Acked-by: Ilpo Järvinen &lt;ilpo.jarvinen@helsinki.fi&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: TSO packets automatic sizing</title>
<updated>2013-11-04T12:35:11+00:00</updated>
<author>
<name>Eric Dumazet</name>
<email>edumazet@google.com</email>
</author>
<published>2013-08-27T12:46:32+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=dbeb18b221975144a8bd9258c7dba4b0d7aa3a31'/>
<id>dbeb18b221975144a8bd9258c7dba4b0d7aa3a31</id>
<content type='text'>
[ Upstream commits 6d36824e730f247b602c90e8715a792003e3c5a7,
  02cf4ebd82ff0ac7254b88e466820a290ed8289a, and parts of
  7eec4174ff29cd42f2acfae8112f51c228545d40 ]

After hearing many people over past years complaining against TSO being
bursty or even buggy, we are proud to present automatic sizing of TSO
packets.

One part of the problem is that tcp_tso_should_defer() uses an heuristic
relying on upcoming ACKS instead of a timer, but more generally, having
big TSO packets makes little sense for low rates, as it tends to create
micro bursts on the network, and general consensus is to reduce the
buffering amount.

This patch introduces a per socket sk_pacing_rate, that approximates
the current sending rate, and allows us to size the TSO packets so
that we try to send one packet every ms.

This field could be set by other transports.

Patch has no impact for high speed flows, where having large TSO packets
makes sense to reach line rate.

For other flows, this helps better packet scheduling and ACK clocking.

This patch increases performance of TCP flows in lossy environments.

A new sysctl (tcp_min_tso_segs) is added, to specify the
minimal size of a TSO packet (default being 2).

A follow-up patch will provide a new packet scheduler (FQ), using
sk_pacing_rate as an input to perform optional per flow pacing.

This explains why we chose to set sk_pacing_rate to twice the current
rate, allowing 'slow start' ramp up.

sk_pacing_rate = 2 * cwnd * mss / srtt

v2: Neal Cardwell reported a suspect deferring of last two segments on
initial write of 10 MSS, I had to change tcp_tso_should_defer() to take
into account tp-&gt;xmit_size_goal_segs

Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Cc: Neal Cardwell &lt;ncardwell@google.com&gt;
Cc: Yuchung Cheng &lt;ycheng@google.com&gt;
Cc: Van Jacobson &lt;vanj@google.com&gt;
Cc: Tom Herbert &lt;therbert@google.com&gt;
Acked-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
[ Upstream commits 6d36824e730f247b602c90e8715a792003e3c5a7,
  02cf4ebd82ff0ac7254b88e466820a290ed8289a, and parts of
  7eec4174ff29cd42f2acfae8112f51c228545d40 ]

After hearing many people over past years complaining against TSO being
bursty or even buggy, we are proud to present automatic sizing of TSO
packets.

One part of the problem is that tcp_tso_should_defer() uses an heuristic
relying on upcoming ACKS instead of a timer, but more generally, having
big TSO packets makes little sense for low rates, as it tends to create
micro bursts on the network, and general consensus is to reduce the
buffering amount.

This patch introduces a per socket sk_pacing_rate, that approximates
the current sending rate, and allows us to size the TSO packets so
that we try to send one packet every ms.

This field could be set by other transports.

Patch has no impact for high speed flows, where having large TSO packets
makes sense to reach line rate.

For other flows, this helps better packet scheduling and ACK clocking.

This patch increases performance of TCP flows in lossy environments.

A new sysctl (tcp_min_tso_segs) is added, to specify the
minimal size of a TSO packet (default being 2).

A follow-up patch will provide a new packet scheduler (FQ), using
sk_pacing_rate as an input to perform optional per flow pacing.

This explains why we chose to set sk_pacing_rate to twice the current
rate, allowing 'slow start' ramp up.

sk_pacing_rate = 2 * cwnd * mss / srtt

v2: Neal Cardwell reported a suspect deferring of last two segments on
initial write of 10 MSS, I had to change tcp_tso_should_defer() to take
into account tp-&gt;xmit_size_goal_segs

Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Cc: Neal Cardwell &lt;ncardwell@google.com&gt;
Cc: Yuchung Cheng &lt;ycheng@google.com&gt;
Cc: Van Jacobson &lt;vanj@google.com&gt;
Cc: Tom Herbert &lt;therbert@google.com&gt;
Acked-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: don't apply tsoffset if rcv_tsecr is zero</title>
<updated>2013-08-29T19:11:12+00:00</updated>
<author>
<name>Andrew Vagin</name>
<email>avagin@openvz.org</email>
</author>
<published>2013-08-27T08:21:55+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=e3e12028315749b7fa2edbc37328e5847be9ede9'/>
<id>e3e12028315749b7fa2edbc37328e5847be9ede9</id>
<content type='text'>
The zero value means that tsecr is not valid, so it's a special case.

tsoffset is used to customize tcp_time_stamp for one socket.
tsoffset is usually zero, it's used when a socket was moved from one
host to another host.

Currently this issue affects logic of tcp_rcv_rtt_measure_ts. Due to
incorrect value of rcv_tsecr, tcp_rcv_rtt_measure_ts sets rto to
TCP_RTO_MAX.

Cc: Pavel Emelyanov &lt;xemul@parallels.com&gt;
Cc: Eric Dumazet &lt;eric.dumazet@gmail.com&gt;
Cc: "David S. Miller" &lt;davem@davemloft.net&gt;
Cc: Alexey Kuznetsov &lt;kuznet@ms2.inr.ac.ru&gt;
Cc: James Morris &lt;jmorris@namei.org&gt;
Cc: Hideaki YOSHIFUJI &lt;yoshfuji@linux-ipv6.org&gt;
Cc: Patrick McHardy &lt;kaber@trash.net&gt;
Reported-by: Cyrill Gorcunov &lt;gorcunov@openvz.org&gt;
Signed-off-by: Andrey Vagin &lt;avagin@openvz.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>
The zero value means that tsecr is not valid, so it's a special case.

tsoffset is used to customize tcp_time_stamp for one socket.
tsoffset is usually zero, it's used when a socket was moved from one
host to another host.

Currently this issue affects logic of tcp_rcv_rtt_measure_ts. Due to
incorrect value of rcv_tsecr, tcp_rcv_rtt_measure_ts sets rto to
TCP_RTO_MAX.

Cc: Pavel Emelyanov &lt;xemul@parallels.com&gt;
Cc: Eric Dumazet &lt;eric.dumazet@gmail.com&gt;
Cc: "David S. Miller" &lt;davem@davemloft.net&gt;
Cc: Alexey Kuznetsov &lt;kuznet@ms2.inr.ac.ru&gt;
Cc: James Morris &lt;jmorris@namei.org&gt;
Cc: Hideaki YOSHIFUJI &lt;yoshfuji@linux-ipv6.org&gt;
Cc: Patrick McHardy &lt;kaber@trash.net&gt;
Reported-by: Cyrill Gorcunov &lt;gorcunov@openvz.org&gt;
Signed-off-by: Andrey Vagin &lt;avagin@openvz.org&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: introduce a per-route knob for quick ack</title>
<updated>2013-06-20T06:06:51+00:00</updated>
<author>
<name>Cong Wang</name>
<email>amwang@redhat.com</email>
</author>
<published>2013-06-15T01:39:18+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=bcefe17cffd06efdda3e7ad679ea743236e6271a'/>
<id>bcefe17cffd06efdda3e7ad679ea743236e6271a</id>
<content type='text'>
In previous discussions, I tried to find some reasonable heuristics
for delayed ACK, however this seems not possible, according to Eric:

	"ACKS might also be delayed because of bidirectional
	traffic, and is more controlled by the application
	response time. TCP stack can not easily estimate it."

	"ACK can be incredibly useful to recover from losses in
	a short time.

	The vast majority of TCP sessions are small lived, and we
	send one ACK per received segment anyway at beginning or
	retransmits to let the sender smoothly increase its cwnd,
	so an auto-tuning facility wont help them that much."

and according to David:

	"ACKs are the only information we have to detect loss.

	And, for the same reasons that TCP VEGAS is fundamentally
	broken, we cannot measure the pipe or some other
	receiver-side-visible piece of information to determine
	when it's "safe" to stretch ACK.

	And even if it's "safe", we should not do it so that losses are
	accurately detected and we don't spuriously retransmit.

	The only way to know when the bandwidth increases is to
	"test" it, by sending more and more packets until drops happen.
	That's why all successful congestion control algorithms must
	operate on explicited tested pieces of information.

	Similarly, it's not really possible to universally know if
	it's safe to stretch ACK or not."

It still makes sense to enable or disable quick ack mode like
what TCP_QUICK_ACK does.

Similar to TCP_QUICK_ACK option, but for people who can't
modify the source code and still wants to control
TCP delayed ACK behavior. As David suggested, this should belong
to per-path scope, since different pathes may want different
behaviors.

Cc: Eric Dumazet &lt;eric.dumazet@gmail.com&gt;
Cc: Rick Jones &lt;rick.jones2@hp.com&gt;
Cc: Stephen Hemminger &lt;stephen@networkplumber.org&gt;
Cc: "David S. Miller" &lt;davem@davemloft.net&gt;
Cc: Thomas Graf &lt;tgraf@suug.ch&gt;
CC: David Laight &lt;David.Laight@ACULAB.COM&gt;
Signed-off-by: Cong Wang &lt;amwang@redhat.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>
In previous discussions, I tried to find some reasonable heuristics
for delayed ACK, however this seems not possible, according to Eric:

	"ACKS might also be delayed because of bidirectional
	traffic, and is more controlled by the application
	response time. TCP stack can not easily estimate it."

	"ACK can be incredibly useful to recover from losses in
	a short time.

	The vast majority of TCP sessions are small lived, and we
	send one ACK per received segment anyway at beginning or
	retransmits to let the sender smoothly increase its cwnd,
	so an auto-tuning facility wont help them that much."

and according to David:

	"ACKs are the only information we have to detect loss.

	And, for the same reasons that TCP VEGAS is fundamentally
	broken, we cannot measure the pipe or some other
	receiver-side-visible piece of information to determine
	when it's "safe" to stretch ACK.

	And even if it's "safe", we should not do it so that losses are
	accurately detected and we don't spuriously retransmit.

	The only way to know when the bandwidth increases is to
	"test" it, by sending more and more packets until drops happen.
	That's why all successful congestion control algorithms must
	operate on explicited tested pieces of information.

	Similarly, it's not really possible to universally know if
	it's safe to stretch ACK or not."

It still makes sense to enable or disable quick ack mode like
what TCP_QUICK_ACK does.

Similar to TCP_QUICK_ACK option, but for people who can't
modify the source code and still wants to control
TCP delayed ACK behavior. As David suggested, this should belong
to per-path scope, since different pathes may want different
behaviors.

Cc: Eric Dumazet &lt;eric.dumazet@gmail.com&gt;
Cc: Rick Jones &lt;rick.jones2@hp.com&gt;
Cc: Stephen Hemminger &lt;stephen@networkplumber.org&gt;
Cc: "David S. Miller" &lt;davem@davemloft.net&gt;
Cc: Thomas Graf &lt;tgraf@suug.ch&gt;
CC: David Laight &lt;David.Laight@ACULAB.COM&gt;
Signed-off-by: Cong Wang &lt;amwang@redhat.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: properly send new data in fast recovery in first RTT</title>
<updated>2013-06-13T09:46:29+00:00</updated>
<author>
<name>Yuchung Cheng</name>
<email>ycheng@google.com</email>
</author>
<published>2013-06-11T22:35:32+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=85f16525a2eb66e6092cbd8dcf42371df8334ed0'/>
<id>85f16525a2eb66e6092cbd8dcf42371df8334ed0</id>
<content type='text'>
Linux sends new unset data during disorder and recovery state if all
(suspected) lost packets have been retransmitted ( RFC5681, section
3.2 step 1 &amp; 2, RFC3517 section 4, NexSeg() Rule 2).  One requirement
is to keep the receive window about twice the estimated sender's
congestion window (tcp_rcv_space_adjust()), assuming the fast
retransmits repair the losses in the next round trip.

But currently it's not the case on the first round trip in either
normal or Fast Open connection, beucase the initial receive window
is identical to (expected) sender's initial congestion window. The
fix is to double it.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Acked-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>
Linux sends new unset data during disorder and recovery state if all
(suspected) lost packets have been retransmitted ( RFC5681, section
3.2 step 1 &amp; 2, RFC3517 section 4, NexSeg() Rule 2).  One requirement
is to keep the receive window about twice the estimated sender's
congestion window (tcp_rcv_space_adjust()), assuming the fast
retransmits repair the losses in the next round trip.

But currently it's not the case on the first round trip in either
normal or Fast Open connection, beucase the initial receive window
is identical to (expected) sender's initial congestion window. The
fix is to double it.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Acked-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>tcp: undo on DSACK during recovery</title>
<updated>2013-05-31T01:06:11+00:00</updated>
<author>
<name>Yuchung Cheng</name>
<email>ycheng@google.com</email>
</author>
<published>2013-05-29T14:20:14+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=c7d9d6a185a7ea383b719b79c428d34ec1470275'/>
<id>c7d9d6a185a7ea383b719b79c428d34ec1470275</id>
<content type='text'>
If the receiver supports DSACK, sender can detect false recoveries and
revert cwnd reductions triggered by either severe network reordering or
concurrent reordering and loss event.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@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>
If the receiver supports DSACK, sender can detect false recoveries and
revert cwnd reductions triggered by either severe network reordering or
concurrent reordering and loss event.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: fix undo on partial ack in recovery</title>
<updated>2013-05-31T01:06:11+00:00</updated>
<author>
<name>Yuchung Cheng</name>
<email>ycheng@google.com</email>
</author>
<published>2013-05-29T14:20:13+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=7026b912f97d912476dff5465ed9a127be094208'/>
<id>7026b912f97d912476dff5465ed9a127be094208</id>
<content type='text'>
Upon detecting spurious fast retransmit via timestamps during recovery,
use PRR to clock out new data packet instead of retransmission. Once
all retransmission are proven spurious, the sender then reverts the
cwnd reduction and congestion state to open or disorder.

The current code does the opposite: it undoes cwnd as soon as any
retransmission is spurious and continues to retransmit until all
data are acked. This nullifies the point to undo the cwnd because
the sender is still retransmistting spuriously. This patch fixes
it. The undo_ssthresh argument of tcp_undo_cwnd_reductiuon() is no
longer needed and is removed.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@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>
Upon detecting spurious fast retransmit via timestamps during recovery,
use PRR to clock out new data packet instead of retransmission. Once
all retransmission are proven spurious, the sender then reverts the
cwnd reduction and congestion state to open or disorder.

The current code does the opposite: it undoes cwnd as soon as any
retransmission is spurious and continues to retransmit until all
data are acked. This nullifies the point to undo the cwnd because
the sender is still retransmistting spuriously. This patch fixes
it. The undo_ssthresh argument of tcp_undo_cwnd_reductiuon() is no
longer needed and is removed.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: refactor undo functions</title>
<updated>2013-05-31T01:06:11+00:00</updated>
<author>
<name>Yuchung Cheng</name>
<email>ycheng@google.com</email>
</author>
<published>2013-05-29T14:20:12+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=6a63df46a7363833a0dc0c431027f522b3487972'/>
<id>6a63df46a7363833a0dc0c431027f522b3487972</id>
<content type='text'>
Refactor and relocate various functions or variables to prepare the
undo fix.  Remove some unused function arguments. Rename tcp_undo_cwr
to tcp_undo_cwnd_reduction to be consistent with the rest of
CWR related function names.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@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>
Refactor and relocate various functions or variables to prepare the
undo fix.  Remove some unused function arguments. Rename tcp_undo_cwr
to tcp_undo_cwnd_reduction to be consistent with the rest of
CWR related function names.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: consolidate PRR packet accounting</title>
<updated>2013-05-31T01:06:11+00:00</updated>
<author>
<name>Yuchung Cheng</name>
<email>ycheng@google.com</email>
</author>
<published>2013-05-29T14:20:11+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=6804973ffb4288bba14d53223e2fbb2bbd1d2e1b'/>
<id>6804973ffb4288bba14d53223e2fbb2bbd1d2e1b</id>
<content type='text'>
This patch series fixes an undo bug in fast recovery: the sender
mistakenly undos the cwnd too early but continues fast retransmits
until all pending data are acked. This also multiplies the SNMP
stat PARTIALUNDO events by the degree of the network reordering.

The first patch prepares the fix by consolidating the accounting
of newly_acked_sacked in tcp_cwnd_reduction(), instead of updating
newly_acked_sacked everytime sacked_out is adjusted.  Also pass
acked and prior_unsacked as const type because they are readonly
in the rest of recovery processing.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@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 series fixes an undo bug in fast recovery: the sender
mistakenly undos the cwnd too early but continues fast retransmits
until all pending data are acked. This also multiplies the SNMP
stat PARTIALUNDO events by the degree of the network reordering.

The first patch prepares the fix by consolidating the accounting
of newly_acked_sacked in tcp_cwnd_reduction(), instead of updating
newly_acked_sacked everytime sacked_out is adjusted.  Also pass
acked and prior_unsacked as const type because they are readonly
in the rest of recovery processing.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
</feed>
