tp->lsndtime may not always be the SYNACK timestamp if a passive
Fast Open socket sends data before handshake completes. And if the
remote acknowledges both the data and the SYNACK, the RTT sample
is already taken in tcp_ack(), so no need to call
tcp_update_ack_rtt() in tcp_synack_rtt_meas() aagain.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

For passive TCP connections, upon receiving the ACK that completes the
3WHS, make sure we set our pacing rate after we get our first RTT
sample.

On passive TCP connections, when we receive the ACK completing the
3WHS we do not take an RTT sample in tcp_ack(), but rather in
tcp_synack_rtt_meas(). So upon receiving the ACK that completes the
3WHS, tcp_ack() leaves sk_pacing_rate at its initial value.

Originally the initial sk_pacing_rate value was 0, so passive-side
connections defaulted to sysctl_tcp_min_tso_segs (2 segs) in skbuffs
made in the first RTT. With a default initial cwnd of 10 packets, this
happened to be correct for RTTs 5ms or bigger, so it was hard to
see problems in WAN or emulated WAN testing.

Since 7eec4174 ("pkt_sched: fq: fix non TCP flows pacing"), the
initial sk_pacing_rate is 0xffffffff. So after that change, passive
TCP connections were keeping this value (and using large numbers of
segments per skbuff) until receiving an ACK for data.
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Cc: Eric Dumazet <edumazet@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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: NMichael Sterrett <michael@sterretts.net>
Tested-By: NDormando <dormando@rydia.net>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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 < S 0:0(0) win 32792 <mss 1000,sackOK,nop,nop,nop,wscale 7>
+.000 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 6>
+.001 < . 1:1(0) ack 1 win 1024
+.000 accept(3, ..., ...) = 4

+.100 write(4, ..., 12000) = 12000
+.000 shutdown(4, SHUT_WR) = 0
+.000 > . 1:10001(10000) ack 1
+.050 < . 1:1(0) ack 2001 win 257
+.000 > FP. 10001:12001(2000) ack 1
+.050 < . 1:1(0) ack 2001 win 257 <sack 10001:11001,nop,nop>
+.050 < . 1:1(0) ack 2001 win 257 <sack 10001:12002,nop,nop>
// SACK reneg
+.050 < . 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 832d11c5
("tcp: Try to restore large SKBs while SACK processing")
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Cc: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi>
Acked-by: NIlpo Järvinen <ilpo.jarvinen@helsinki.fi>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

TCP receive window handling is multi staged.

A socket has a memory budget, static or dynamic, in sk_rcvbuf.

Because we do not really know how this memory budget translates to
a TCP window (payload), TCP announces a small initial window
(about 20 MSS).

When a packet is received, we increase TCP rcv_win depending
on the payload/truesize ratio of this packet. Good citizen
packets give a hint that it's reasonable to have rcv_win = sk_rcvbuf/2

This heuristic takes place in tcp_grow_window()

Problem is : We currently call tcp_grow_window() only for in-order
packets.

This means that reorders or packet losses stop proper grow of
rcv_win, and senders are unable to benefit from fast recovery,
or proper reordering level detection.

Really, a packet being stored in OFO queue is not a bad citizen.
It should be part of the game as in-order packets.

In our traces, we very often see sender is limited by linux small
receive windows, even if linux hosts use autotuning (DRS) and should
allow rcv_win to grow to ~3MB.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

In commit 0f7cc9a3 "tcp: increase throughput when reordering is high",
it only allows cwnd to increase in Open state. This mistakenly disables
slow start after timeout (CA_Loss). Moreover cwnd won't grow if the
state moves from Disorder to Open later in tcp_fastretrans_alert().

Therefore the correct logic should be to allow cwnd to grow as long
as the data is received in order in Open, Loss, or even Disorder state.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

tcp_rcv_established() returns only one value namely 0. We change the return
value to void (as suggested by David Miller).

After commit 0c24604b (tcp: implement RFC 5961 4.2), we no longer send RSTs in
response to SYNs. We can remove the check and processing on the return value of
tcp_rcv_established().

We also fix jtcp_rcv_established() in tcp_probe.c to match that of
tcp_rcv_established().
Signed-off-by: NVijay Subramanian <subramanian.vijay@gmail.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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->xmit_size_goal_segs
Signed-off-by: NEric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Van Jacobson <vanj@google.com>
Cc: Tom Herbert <therbert@google.com>
Acked-by: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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 <xemul@parallels.com>
Cc: Eric Dumazet <eric.dumazet@gmail.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>
Reported-by: NCyrill Gorcunov <gorcunov@openvz.org>
Signed-off-by: NAndrey Vagin <avagin@openvz.org>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

The stack currently detects reordering and avoid spurious
retransmission very well. However the throughput is sub-optimal under
high reordering because cwnd is increased only if the data is deliverd
in order. I.e., FLAG_DATA_ACKED check in tcp_ack().  The more packet
are reordered the worse the throughput is.

Therefore when reordering is proven high, cwnd should advance whenever
the data is delivered regardless of its ordering. If reordering is low,
conservatively advance cwnd only on ordered deliveries in Open state,
and retain cwnd in Disordered state (RFC5681).

Using netperf on a qdisc setup of 20Mbps BW and random RTT from 45ms
to 55ms (for reordering effect). This change increases TCP throughput
by 20 - 25% to near bottleneck BW.

A special case is the stretched ACK with new SACK and/or ECE mark.
For example, a receiver may receive an out of order or ECN packet with
unacked data buffered because of LRO or delayed ACK. The principle on
such an ACK is to advance cwnd on the cummulative acked part first,
then reduce cwnd in tcp_fastretrans_alert().
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

On timeout the TCP sender unconditionally resets the estimated degree
of network reordering (tp->reordering). The idea behind this is that
the estimate is too large to trigger fast recovery (e.g., due to a IP
path change).

But for example if the sender only had 2 packets outstanding, then a
timeout doesn't tell much about reordering. A sender that learns about
reordering on big writes and loses packets on small writes will end up
falsely retransmitting again and again, especially when reordering is
more likely on big writes.

Therefore the sender should only suspect that tp->reordering is too
high if it could have gone into fast recovery with the (lower) default
estimate.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

If RTT is not available because Karn's check has failed or no
new packet is acked, use the RTT measured from SACK to estimate
the RTO. The sender can continue to estimate the RTO during loss
recovery or reordering event upon receiving non-partial ACKs.

This also changes when the RTO is re-armed. Previously it is
only re-armed when some data is cummulatively acknowledged (i.e.,
SND.UNA advances), but now it is re-armed whenever RTT estimator
is updated. This feature is particularly useful to reduce spurious
timeout for buffer bloat including cellular carriers [1], and
RTT estimation on reordering events.

[1] "An In-depth Study of LTE: Effect of Network Protocol and
 Application Behavior on Performance", In Proc. of SIGCOMM 2013
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Take RTT sample if an ACK selectively acks some sequences that
have never been retransmitted. The Karn's algorithm does not apply
even if that ACK (s)acks other retransmitted sequences, because it
must been generated by an original but perhaps out-of-order packet.
There is no ambiguity. In case when multiple blocks are newly
sacked because of ACK losses the earliest block is used to
measure RTT, similar to cummulative ACKs.

Such RTT samples allow the sender to estimate the RTO during loss
recovery and packet reordering events. It is still useful even with
TCP timestamps. That's because during these events the SND.UNA may
not advance preventing RTT samples from TS ECR (thus the FLAG_ACKED
check before calling tcp_ack_update_rtt()).  Therefore this new
RTT source is complementary to existing ACK and TS RTT mechanisms.

This patch does not update the RTO. It is done in the next patch.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Prefer packet timings to TS-ecr for RTT measurements when both
sources are available. That's because broken middle-boxes and remote
peer can return packets with corrupted TS ECR fields. Similarly most
congestion controls that require RTT signals favor timing-based
sources as well. Also check for bad TS ECR values to avoid RTT
blow-ups. It has happened on production Web servers.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

The first patch consolidates SYNACK and other RTT measurement to use a
central function tcp_ack_update_rtt(). A (small) bonus is now SYNACK
RTT measurement happens after PAWS check, potentially reducing the
impact of RTO seeding on bad TCP timestamps values.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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 <eric.dumazet@gmail.com>
Cc: Rick Jones <rick.jones2@hp.com>
Cc: Stephen Hemminger <stephen@networkplumber.org>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: Thomas Graf <tgraf@suug.ch>
CC: David Laight <David.Laight@ACULAB.COM>
Signed-off-by: NCong Wang <amwang@redhat.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Linux sends new unset data during disorder and recovery state if all
(suspected) lost packets have been retransmitted ( RFC5681, section
3.2 step 1 & 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: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

case TCP_FIN_WAIT1 can also be simplified by reversing tests
and adding breaks;

Add braces after case and move automatic definitions.
Signed-off-by: NJoe Perches <joe@perches.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

case TCP_SYN_RECV: can have another indentation level removed
by converting

	if (acceptable) {
		...;
	} else {
		return 1;
	}

to
	if (!acceptable)
		return 1;
	...;

Reflow code and comments to fit 80 columns.

Another pure cleanup patch.
Signed-off-by: NJoe Perches <joe@perches.com>
Improved-by: NEric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Remove one level of indentation 'introduced' in commit
c3ae62af (tcp: should drop incoming frames without ACK flag set)

if (true) {
        ...
}

@acceptable variable is a boolean.

This patch is a pure cleanup.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This patch is a fix for a bug triggering newly_acked_sacked < 0
in tcp_ack(.).

The bug is triggered by sacked_out decreasing relative to prior_sacked,
but packets_out remaining the same as pior_packets. This is because the
snapshot of prior_packets is taken after tcp_sacktag_write_queue() while
prior_sacked is captured before tcp_sacktag_write_queue(). The problem
is: tcp_sacktag_write_queue (tcp_match_skb_to_sack() -> tcp_fragment)
adjusts the pcount for packets_out and sacked_out (MSS change or other
reason). As a result, this delta in pcount is reflected in
(prior_sacked - sacked_out) but not in (prior_packets - packets_out).

This patch does the following:
1) initializes prior_packets at the start of tcp_ack() so as to
capture the delta in packets_out created by tcp_fragment.
2) introduces a new "previous_packets_out" variable that snapshots
packets_out right before tcp_clean_rtx_queue, so pkts_acked can be
correctly computed as before.
3) Computes pkts_acked using previous_packets_out, and computes
newly_acked_sacked using prior_packets.
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Acked-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

tcp_timeout_skb() was intended to trigger fast recovery on timeout,
unfortunately in reality it often causes spurious retransmission
storms during fast recovery. The particular sign is a fast retransmit
over the highest sacked sequence (SND.FACK).

Currently the RTO timer re-arming (as in RFC6298) offers a nice cushion
to avoid spurious timeout: when SND.UNA advances the sender re-arms
RTO and extends the timeout by icsk_rto. The sender does not offset
the time elapsed since the packet at SND.UNA was sent.

But if the next (DUP)ACK arrives later than ~RTTVAR and triggers
tcp_fastretrans_alert(), then tcp_timeout_skb() will mark any packet
sent before the icsk_rto interval lost, including one that's above the
highest sacked sequence. Most likely a large part of scorebard will be
marked.

If most packets are not lost then the subsequent DUPACKs with new SACK
blocks will cause the sender to continue to retransmit packets beyond
SND.FACK spuriously. Even if only one packet is lost the sender may
falsely retransmit almost the entire window.

The situation becomes common in the world of bufferbloat: the RTT
continues to grow as the queue builds up but RTTVAR remains small and
close to the minimum 200ms. If a data packet is lost and the DUPACK
triggered by the next data packet is slightly delayed, then a spurious
retransmission storm forms.

As the original comment on tcp_timeout_skb() suggests: the usefulness
of this feature is questionable. It also wastes cycles walking the
sack scoreboard and is actually harmful because of false recovery.

It's time to remove this.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Acked-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

tcp_fixup_rcvbuf() contains a loop to estimate initial socket
rcv space needed for a given mss. With large MTU (like 64K on lo),
we can loop ~500 times and consume a lot of cpu cycles.

perf top of 200 concurrent netperf -t TCP_CRR

5.62%  netperf  [kernel.kallsyms]  [k] tcp_init_buffer_space
1.71%  netperf  [kernel.kallsyms]  [k] _raw_spin_lock
1.55%  netperf  [kernel.kallsyms]  [k] kmem_cache_free
1.51%  netperf  [kernel.kallsyms]  [k] tcp_transmit_skb
1.50%  netperf  [kernel.kallsyms]  [k] tcp_ack

Lets use a 100% factor, and remove the loop.

100% is needed anyway for tcp_adv_win_scale=1
default value, and is also the maximum factor.

Refs: commit b49960a0
      ("tcp: change tcp_adv_win_scale and tcp_rmem[2]")
Signed-off-by: NEric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Add MIB counters for checksum errors in IP layer,
and TCP/UDP/ICMP layers, to help diagnose problems.

$ nstat -a | grep  Csum
IcmpInCsumErrors                72                 0.0
TcpInCsumErrors                 382                0.0
UdpInCsumErrors                 463221             0.0
Icmp6InCsumErrors               75                 0.0
Udp6InCsumErrors                173442             0.0
IpExtInCsumErrors               10884              0.0
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

commit bd090dfc (tcp: tcp_replace_ts_recent() should not be called
from tcp_validate_incoming()) introduced a TS ecr bug in slow path
processing.

1 A > B P. 1:10001(10000) ack 1 <nop,nop,TS val 1001 ecr 200>
2 B < A . 1:1(0) ack 1 win 257 <sack 9001:10001,TS val 300 ecr 1001>
3 A > B . 1:1001(1000) ack 1 win 227 <nop,nop,TS val 1002 ecr 200>
4 A > B . 1001:2001(1000) ack 1 win 227 <nop,nop,TS val 1002 ecr 200>

(ecr 200 should be ecr 300 in packets 3 & 4)

Problem is tcp_ack() can trigger send of new packets (retransmits),
reflecting the prior TSval, instead of the TSval contained in the
currently processed incoming packet.

Fix this by calling tcp_replace_ts_recent() from tcp_ack() after the
checks, but before the actions.
Reported-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

On SACK reneging the sender immediately retransmits and forces a
timeout but disables Eifel (undo). If the (buggy) receiver does not
drop any packet this can trigger a false slow-start retransmit storm
driven by the ACKs of the original packets. This can be detected with
undo and TCP timestamps.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This patch implements F-RTO (foward RTO recovery):

When the first retransmission after timeout is acknowledged, F-RTO
sends new data instead of old data. If the next ACK acknowledges
some never-retransmitted data, then the timeout was spurious and the
congestion state is reverted.  Otherwise if the next ACK selectively
acknowledges the new data, then the timeout was genuine and the
loss recovery continues. This idea applies to recurring timeouts
as well. While F-RTO sends different data during timeout recovery,
it does not (and should not) change the congestion control.

The implementaion follows the three steps of SACK enhanced algorithm
(section 3) in RFC5682. Step 1 is in tcp_enter_loss(). Step 2 and
3 are in tcp_process_loss().  The basic version is not supported
because SACK enhanced version also works for non-SACK connections.

The new implementation is functionally in parity with the old F-RTO
implementation except the one case where it increases undo events:
In addition to the RFC algorithm, a spurious timeout may be detected
without sending data in step 2, as long as the SACK confirms not
all the original data are dropped. When this happens, the sender
will undo the cwnd and perhaps enter fast recovery instead. This
additional check increases the F-RTO undo events by 5x compared
to the prior implementation on Google Web servers, since the sender
often does not have new data to send for HTTP.

Note F-RTO may detect spurious timeout before Eifel with timestamps
does so.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Consolidate all of TCP CA_Loss state processing in
tcp_fastretrans_alert() into a new function called tcp_process_loss().
This is to prepare the new F-RTO implementation in the next patch.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

The patch series refactor the F-RTO feature (RFC4138/5682).

This is to simplify the loss recovery processing. Existing F-RTO
was developed during the experimental stage (RFC4138) and has
many experimental features.  It takes a separate code path from
the traditional timeout processing by overloading CA_Disorder
instead of using CA_Loss state. This complicates CA_Disorder state
handling because it's also used for handling dubious ACKs and undos.
While the algorithm in the RFC does not change the congestion control,
the implementation intercepts congestion control in various places
(e.g., frto_cwnd in tcp_ack()).

The new code implements newer F-RTO RFC5682 using CA_Loss processing
path.  F-RTO becomes a small extension in the timeout processing
and interfaces with congestion control and Eifel undo modules.
It lets congestion control (module) determines how many to send
independently.  F-RTO only chooses what to send in order to detect
spurious retranmission. If timeout is found spurious it invokes
existing Eifel undo algorithms like DSACK or TCP timestamp based
detection.

The first patch removes all F-RTO code except the sysctl_tcp_frto is
left for the new implementation.  Since CA_EVENT_FRTO is removed, TCP
westwood now computes ssthresh on regular timeout CA_EVENT_LOSS event.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

TCPCT uses option-number 253, reserved for experimental use and should
not be used in production environments.
Further, TCPCT does not fully implement RFC 6013.

As a nice side-effect, removing TCPCT increases TCP's performance for
very short flows:

Doing an apache-benchmark with -c 100 -n 100000, sending HTTP-requests
for files of 1KB size.

before this patch:
	average (among 7 runs) of 20845.5 Requests/Second
after:
	average (among 7 runs) of 21403.6 Requests/Second
Signed-off-by: NChristoph Paasch <christoph.paasch@uclouvain.be>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This is the second of the TLP patch series; it augments the basic TLP
algorithm with a loss detection scheme.

This patch implements a mechanism for loss detection when a Tail
loss probe retransmission plugs a hole thereby masking packet loss
from the sender. The loss detection algorithm relies on counting
TLP dupacks as outlined in Sec. 3 of:
http://tools.ietf.org/html/draft-dukkipati-tcpm-tcp-loss-probe-01

The basic idea is: Sender keeps track of TLP "episode" upon
retransmission of a TLP packet. An episode ends when the sender receives
an ACK above the SND.NXT (tracked by tlp_high_seq) at the time of the
episode. We want to make sure that before the episode ends the sender
receives a "TLP dupack", indicating that the TLP retransmission was
unnecessary, so there was no loss/hole that needed plugging. If the
sender gets no TLP dupack before the end of the episode, then it reduces
ssthresh and the congestion window, because the TLP packet arriving at
the receiver probably plugged a hole.
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This patch series implement the Tail loss probe (TLP) algorithm described
in http://tools.ietf.org/html/draft-dukkipati-tcpm-tcp-loss-probe-01. The
first patch implements the basic algorithm.

TLP's goal is to reduce tail latency of short transactions. It achieves
this by converting retransmission timeouts (RTOs) occuring due
to tail losses (losses at end of transactions) into fast recovery.
TLP transmits one packet in two round-trips when a connection is in
Open state and isn't receiving any ACKs. The transmitted packet, aka
loss probe, can be either new or a retransmission. When there is tail
loss, the ACK from a loss probe triggers FACK/early-retransmit based
fast recovery, thus avoiding a costly RTO. In the absence of loss,
there is no change in the connection state.

PTO stands for probe timeout. It is a timer event indicating
that an ACK is overdue and triggers a loss probe packet. The PTO value
is set to max(2*SRTT, 10ms) and is adjusted to account for delayed
ACK timer when there is only one oustanding packet.

TLP Algorithm

On transmission of new data in Open state:
  -> packets_out > 1: schedule PTO in max(2*SRTT, 10ms).
  -> packets_out == 1: schedule PTO in max(2*RTT, 1.5*RTT + 200ms)
  -> PTO = min(PTO, RTO)

Conditions for scheduling PTO:
  -> Connection is in Open state.
  -> Connection is either cwnd limited or no new data to send.
  -> Number of probes per tail loss episode is limited to one.
  -> Connection is SACK enabled.

When PTO fires:
  new_segment_exists:
    -> transmit new segment.
    -> packets_out++. cwnd remains same.

  no_new_packet:
    -> retransmit the last segment.
       Its ACK triggers FACK or early retransmit based recovery.

ACK path:
  -> rearm RTO at start of ACK processing.
  -> reschedule PTO if need be.

In addition, the patch includes a small variation to the Early Retransmit
(ER) algorithm, such that ER and TLP together can in principle recover any
N-degree of tail loss through fast recovery. TLP is controlled by the same
sysctl as ER, tcp_early_retrans sysctl.
tcp_early_retrans==0; disables TLP and ER.
		 ==1; enables RFC5827 ER.
		 ==2; delayed ER.
		 ==3; TLP and delayed ER. [DEFAULT]
		 ==4; TLP only.

The TLP patch series have been extensively tested on Google Web servers.
It is most effective for short Web trasactions, where it reduced RTOs by 15%
and improved HTTP response time (average by 6%, 99th percentile by 10%).
The transmitted probes account for <0.5% of the overall transmissions.
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Acked-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

We should not update ts_recent and call tcp_rcv_rtt_measure_ts() both
before and after going to step5. That wastes CPU and double-counts the
receiver-side RTT sample.
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Patch cef401de (net: fix possible wrong checksum
generation) fixed wrong checksum calculation but it broke TSO by
defining new GSO type but not a netdev feature for that type.
net_gso_ok() would not allow hardware checksum/segmentation
offload of such packets without the feature.

Following patch fixes TSO and wrong checksum. This patch uses
same logic that Eric Dumazet used. Patch introduces new flag
SKBTX_SHARED_FRAG if at least one frag can be modified by
the user. but SKBTX_SHARED_FRAG flag is kept in skb shared
info tx_flags rather than gso_type.

tx_flags is better compared to gso_type since we can have skb with
shared frag without gso packet. It does not link SHARED_FRAG to
GSO, So there is no need to define netdev feature for this.
Signed-off-by: NPravin B Shelar <pshelar@nicira.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

A socket timestamp is a sum of the global tcp_time_stamp and
a per-socket offset.

A socket offset is added in places where externally visible
tcp timestamp option is parsed/initialized.

Connections in the SYN_RECV state are not supported, global
tcp_time_stamp is used for them, because repair mode doesn't support
this state. In a future it can be implemented by the similar way
as for TIME_WAIT sockets.

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 Dumazet <edumazet@google.com>
Cc: Pavel Emelyanov <xemul@parallels.com>
Signed-off-by: NAndrey Vagin <avagin@openvz.org>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

There are transients during normal FRTO procedure during which
the packets_in_flight can go to zero between write_queue state
updates and firing the resulting segments out. As FRTO processing
occurs during that window the check must be more precise to
not match "spuriously" :-). More specificly, e.g., when
packets_in_flight is zero but FLAG_DATA_ACKED is true the problematic
branch that set cwnd into zero would not be taken and new segments
might be sent out later.
Signed-off-by: NIlpo Järvinen <ilpo.jarvinen@helsinki.fi>
Tested-by: NEric Dumazet <edumazet@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>