Some TCP peers announce a very small MSS option in their SYN and/or
SYN/ACK messages.

This forces the stack to send packets with a very high network/cpu
overhead.

Linux has enforced a minimal value of 48. Since this value includes
the size of TCP options, and that the options can consume up to 40
bytes, this means that each segment can include only 8 bytes of payload.

In some cases, it can be useful to increase the minimal value
to a saner value.

We still let the default to 48 (TCP_MIN_SND_MSS), for compatibility
reasons.

Note that TCP_MAXSEG socket option enforces a minimal value
of (TCP_MIN_MSS). David Miller increased this minimal value
in commit c39508d6 ("tcp: Make TCP_MAXSEG minimum more correct.")
from 64 to 88.

We might in the future merge TCP_MIN_SND_MSS and TCP_MIN_MSS.

CVE-2019-11479 -- tcp mss hardcoded to 48
Signed-off-by: NEric Dumazet <edumazet@google.com>
Suggested-by: NJonathan Looney <jtl@netflix.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Tyler Hicks <tyhicks@canonical.com>
Cc: Bruce Curtis <brucec@netflix.com>
Cc: Jonathan Lemon <jonathan.lemon@gmail.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Jonathan Looney reported that a malicious peer can force a sender
to fragment its retransmit queue into tiny skbs, inflating memory
usage and/or overflow 32bit counters.

TCP allows an application to queue up to sk_sndbuf bytes,
so we need to give some allowance for non malicious splitting
of retransmit queue.

A new SNMP counter is added to monitor how many times TCP
did not allow to split an skb if the allowance was exceeded.

Note that this counter might increase in the case applications
use SO_SNDBUF socket option to lower sk_sndbuf.

CVE-2019-11478 : tcp_fragment, prevent fragmenting a packet when the
	socket is already using more than half the allowed space
Signed-off-by: NEric Dumazet <edumazet@google.com>
Reported-by: NJonathan Looney <jtl@netflix.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Acked-by: NYuchung Cheng <ycheng@google.com>
Reviewed-by: NTyler Hicks <tyhicks@canonical.com>
Cc: Bruce Curtis <brucec@netflix.com>
Cc: Jonathan Lemon <jonathan.lemon@gmail.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Jonathan Looney reported that TCP can trigger the following crash
in tcp_shifted_skb() :

	BUG_ON(tcp_skb_pcount(skb) < pcount);

This can happen if the remote peer has advertized the smallest
MSS that linux TCP accepts : 48

An skb can hold 17 fragments, and each fragment can hold 32KB
on x86, or 64KB on PowerPC.

This means that the 16bit witdh of TCP_SKB_CB(skb)->tcp_gso_segs
can overflow.

Note that tcp_sendmsg() builds skbs with less than 64KB
of payload, so this problem needs SACK to be enabled.
SACK blocks allow TCP to coalesce multiple skbs in the retransmit
queue, thus filling the 17 fragments to maximal capacity.

CVE-2019-11477 -- u16 overflow of TCP_SKB_CB(skb)->tcp_gso_segs

Fixes: 832d11c5 ("tcp: Try to restore large SKBs while SACK processing")
Signed-off-by: NEric Dumazet <edumazet@google.com>
Reported-by: NJonathan Looney <jtl@netflix.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Reviewed-by: NTyler Hicks <tyhicks@canonical.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Bruce Curtis <brucec@netflix.com>
Cc: Jonathan Lemon <jonathan.lemon@gmail.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Add SPDX license identifiers to all files which:

 - Have no license information of any form

 - Have EXPORT_.*_SYMBOL_GPL inside which was used in the
   initial scan/conversion to ignore the file

These files fall under the project license, GPL v2 only. The resulting SPDX
license identifier is:

  GPL-2.0-only
Signed-off-by: NThomas Gleixner <tglx@linutronix.de>
Signed-off-by: NGreg Kroah-Hartman <gregkh@linuxfoundation.org>

Detecting spurious SYNACK timeout using timestamp option requires
recording the exact SYNACK skb timestamp. Previously the SYNACK
sent timestamp was stamped slightly earlier before the skb
was transmitted. This patch uses the SYNACK skb transmission
timestamp directly.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

The non-null check on tskb is always false because it is in an else
path of a check on tskb and hence tskb is null in this code block.
This is check is therefore redundant and can be removed as well
as the label coalesc.

if (tsbk) {
        ...
} else {
        ...
        if (unlikely(!skb)) {
                if (tskb)       /* can never be true, redundant code */
                        goto coalesc;
                return;
        }
}

Addresses-Coverity: ("Logically dead code")
Signed-off-by: NColin Ian King <colin.king@canonical.com>
Reviewed-by: NMukesh Ojha <mojha@codeaurora.org>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

tcp_clock_ns() (aka ktime_get_ns()) is using monotonic clock,
so the checks we had in tcp_mstamp_refresh() are no longer
relevant.

This patch removes cpu stall (when the cache line is not hot)
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

tso_fragment() is only called for packets still in write queue.

Remove the tcp_queue parameter to make this more obvious,
even if the comment clearly states this.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

We prefer static_branch_unlikely() over static_key_false() these days.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

syzbot reported a WARN_ON(!tcp_skb_pcount(skb))
in tcp_send_loss_probe() [1]

This was caused by TCP_REPAIR sent skbs that inadvertenly
were missing a call to tcp_init_tso_segs()

[1]
WARNING: CPU: 1 PID: 0 at net/ipv4/tcp_output.c:2534 tcp_send_loss_probe+0x771/0x8a0 net/ipv4/tcp_output.c:2534
Kernel panic - not syncing: panic_on_warn set ...
CPU: 1 PID: 0 Comm: swapper/1 Not tainted 5.0.0-rc7+ #77
Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011
Call Trace:
 <IRQ>
 __dump_stack lib/dump_stack.c:77 [inline]
 dump_stack+0x172/0x1f0 lib/dump_stack.c:113
 panic+0x2cb/0x65c kernel/panic.c:214
 __warn.cold+0x20/0x45 kernel/panic.c:571
 report_bug+0x263/0x2b0 lib/bug.c:186
 fixup_bug arch/x86/kernel/traps.c:178 [inline]
 fixup_bug arch/x86/kernel/traps.c:173 [inline]
 do_error_trap+0x11b/0x200 arch/x86/kernel/traps.c:271
 do_invalid_op+0x37/0x50 arch/x86/kernel/traps.c:290
 invalid_op+0x14/0x20 arch/x86/entry/entry_64.S:973
RIP: 0010:tcp_send_loss_probe+0x771/0x8a0 net/ipv4/tcp_output.c:2534
Code: 88 fc ff ff 4c 89 ef e8 ed 75 c8 fb e9 c8 fc ff ff e8 43 76 c8 fb e9 63 fd ff ff e8 d9 75 c8 fb e9 94 f9 ff ff e8 bf 03 91 fb <0f> 0b e9 7d fa ff ff e8 b3 03 91 fb 0f b6 1d 37 43 7a 03 31 ff 89
RSP: 0018:ffff8880ae907c60 EFLAGS: 00010206
RAX: ffff8880a989c340 RBX: 0000000000000000 RCX: ffffffff85dedbdb
RDX: 0000000000000100 RSI: ffffffff85dee0b1 RDI: 0000000000000005
RBP: ffff8880ae907c90 R08: ffff8880a989c340 R09: ffffed10147d1ae1
R10: ffffed10147d1ae0 R11: ffff8880a3e8d703 R12: ffff888091b90040
R13: ffff8880a3e8d540 R14: 0000000000008000 R15: ffff888091b90860
 tcp_write_timer_handler+0x5c0/0x8a0 net/ipv4/tcp_timer.c:583
 tcp_write_timer+0x10e/0x1d0 net/ipv4/tcp_timer.c:607
 call_timer_fn+0x190/0x720 kernel/time/timer.c:1325
 expire_timers kernel/time/timer.c:1362 [inline]
 __run_timers kernel/time/timer.c:1681 [inline]
 __run_timers kernel/time/timer.c:1649 [inline]
 run_timer_softirq+0x652/0x1700 kernel/time/timer.c:1694
 __do_softirq+0x266/0x95a kernel/softirq.c:292
 invoke_softirq kernel/softirq.c:373 [inline]
 irq_exit+0x180/0x1d0 kernel/softirq.c:413
 exiting_irq arch/x86/include/asm/apic.h:536 [inline]
 smp_apic_timer_interrupt+0x14a/0x570 arch/x86/kernel/apic/apic.c:1062
 apic_timer_interrupt+0xf/0x20 arch/x86/entry/entry_64.S:807
 </IRQ>
RIP: 0010:native_safe_halt+0x2/0x10 arch/x86/include/asm/irqflags.h:58
Code: ff ff ff 48 89 c7 48 89 45 d8 e8 59 0c a1 fa 48 8b 45 d8 e9 ce fe ff ff 48 89 df e8 48 0c a1 fa eb 82 90 90 90 90 90 90 fb f4 <c3> 0f 1f 00 66 2e 0f 1f 84 00 00 00 00 00 f4 c3 90 90 90 90 90 90
RSP: 0018:ffff8880a98afd78 EFLAGS: 00000286 ORIG_RAX: ffffffffffffff13
RAX: 1ffffffff1125061 RBX: ffff8880a989c340 RCX: 0000000000000000
RDX: dffffc0000000000 RSI: 0000000000000001 RDI: ffff8880a989cbbc
RBP: ffff8880a98afda8 R08: ffff8880a989c340 R09: 0000000000000000
R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000001
R13: ffffffff889282f8 R14: 0000000000000001 R15: 0000000000000000
 arch_cpu_idle+0x10/0x20 arch/x86/kernel/process.c:555
 default_idle_call+0x36/0x90 kernel/sched/idle.c:93
 cpuidle_idle_call kernel/sched/idle.c:153 [inline]
 do_idle+0x386/0x570 kernel/sched/idle.c:262
 cpu_startup_entry+0x1b/0x20 kernel/sched/idle.c:353
 start_secondary+0x404/0x5c0 arch/x86/kernel/smpboot.c:271
 secondary_startup_64+0xa4/0xb0 arch/x86/kernel/head_64.S:243
Kernel Offset: disabled
Rebooting in 86400 seconds..

Fixes: 79861919 ("tcp: fix TCP_REPAIR xmit queue setup")
Signed-off-by: NEric Dumazet <edumazet@google.com>
Reported-by: Nsyzbot <syzkaller@googlegroups.com>
Cc: Andrey Vagin <avagin@openvz.org>
Cc: Soheil Hassas Yeganeh <soheil@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Acked-by: NSoheil Hassas Yeganeh <soheil@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

In order to be more confident about an on-going interactive session, we
increment pingpong count by 1 for every interactive transaction and we
adjust TCP_PINGPONG_THRESH to 3.
This means, we only consider a session in pingpong mode after we see 3
interactive transactions, and start to activate delayed acks in quick
ack mode.
And in order to not over-count the credits, we only increase pingpong
count for the first packet sent in response for the previous received
packet.
This is mainly to prevent delaying the ack immediately after some
handshake protocol but no real interactive traffic pattern afterwards.
Signed-off-by: NWei Wang <weiwan@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Instead of using pingpong as a single bit information, we refactor the
code to treat it as a counter. When interactive session is detected,
we set pingpong count to TCP_PINGPONG_THRESH. And when pingpong count
is >= TCP_PINGPONG_THRESH, we consider the session in pingpong mode.

This patch is a pure refactor and sets foundation for the next patch.
This patch itself does not change any pingpong logic.
Signed-off-by: NWei Wang <weiwan@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Accept MSG_ZEROCOPY in all the TCP states that allow sendmsg. Remove
the explicit check for ESTABLISHED and CLOSE_WAIT states.

This requires correctly handling zerocopy state (uarg, sk_zckey) in
all paths reachable from other TCP states. Such as the EPIPE case
in sk_stream_wait_connect, which a sendmsg() in incorrect state will
now hit. Most paths are already safe.

Only extension needed is for TCP Fastopen active open. This can build
an skb with data in tcp_send_syn_data. Pass the uarg along with other
fastopen state, so that this skb also generates a zerocopy
notification on release.

Tested with active and passive tcp fastopen packetdrill scripts at
https://github.com/wdebruij/packetdrill/commit/1747eef03d25a2404e8132817d0f1244fd6f129dSigned-off-by: NWillem de Bruijn <willemb@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Previously when the sender fails to send (original) data packet or
window probes due to congestion in the local host (e.g. throttling
in qdisc), it'll retry within an RTO or two up to 500ms.

In low-RTT networks such as data-centers, RTO is often far below
the default minimum 200ms. Then local host congestion could trigger
a retry storm pouring gas to the fire. Worse yet, the probe counter
(icsk_probes_out) is not properly updated so the aggressive retry
may exceed the system limit (15 rounds) until the packet finally
slips through.

On such rare events, it's wise to retry more conservatively
(500ms) and update the stats properly to reflect these incidents
and follow the system limit. Note that this is consistent with
the behaviors when a keep-alive probe or RTO retry is dropped
due to local congestion.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Reviewed-by: NNeal Cardwell <ncardwell@google.com>
Reviewed-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Previously TCP socket's retrans_stamp is not set if the
retransmission has failed to send. As a result if a socket is
experiencing local issues to retransmit packets, determining when
to abort a socket is complicated w/o knowning the starting time of
the recovery since retrans_stamp may remain zero.

This complication causes sub-optimal behavior that TCP may use the
latest, instead of the first, retransmission time to compute the
elapsed time of a stalling connection due to local issues. Then TCP
may disrecard TCP retries settings and keep retrying until it finally
succeed: not a good idea when the local host is already strained.

The simple fix is to always timestamp the start of a recovery.
It's worth noting that retrans_stamp is also used to compare echo
timestamp values to detect spurious recovery. This patch does
not break that because retrans_stamp is still later than when the
original packet was sent.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Reviewed-by: NNeal Cardwell <ncardwell@google.com>
Reviewed-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Previously TCP skbs are not always timestamped if the transmission
failed due to memory or other local issues. This makes deciding
when to abort a socket tricky and complicated because the first
unacknowledged skb's timestamp may be 0 on TCP timeout.

The straight-forward fix is to always timestamp skb on every
transmission attempt. Also every skb retransmission needs to be
flagged properly to avoid RTT under-estimation. This can happen
upon receiving an ACK for the original packet and the a previous
(spurious) retransmission has failed.

It's worth noting that this reverts to the old time-stamping
style before commit 8c72c65b ("tcp: update skb->skb_mstamp more
carefully") which addresses a problem in computing the elapsed time
of a stalled window-probing socket. The problem will be addressed
differently in the next patches with a simpler approach.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Reviewed-by: NNeal Cardwell <ncardwell@google.com>
Reviewed-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

In commit f9bfe4e6 ("tcp: lack of available data can also cause
TSO defer") we moved the test in tcp_tso_should_defer() for packets
with a FIN flag, and we mentioned that the same would be done
later for EOR flag.

Both flags should be handled at the same time, after all other
heuristics have been considered. They both mean that no more bytes
can be added to this skb by an application.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Acked-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

tcp_tso_should_defer() can return true in three different cases :

 1) We are cwnd-limited
 2) We are rwnd-limited
 3) We are application limited.

Neal pointed out that my recent fix went too far, since
it assumed that if we were not in 1) case, we must be rwnd-limited

Fix this by properly populating the is_cwnd_limited and
is_rwnd_limited booleans.

After this change, we can finally move the silly check for FIN
flag only for the application-limited case.

The same move for EOR bit will be handled in net-next,
since commit 1c09f7d0 ("tcp: do not try to defer skbs
with eor mark (MSG_EOR)") is scheduled for linux-4.21

Tested by running 200 concurrent netperf -t TCP_RR -- -r 60000,100
and checking none of them was rwnd_limited in the chrono_stat
output from "ss -ti" command.

Fixes: 41727549 ("tcp: Do not underestimate rwnd_limited")
Signed-off-by: NEric Dumazet <edumazet@google.com>
Suggested-by: NNeal Cardwell <ncardwell@google.com>
Reviewed-by: NNeal Cardwell <ncardwell@google.com>
Acked-by: NSoheil Hassas Yeganeh <soheil@google.com>
Reviewed-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

TCP loss probe timer may fire when the retranmission queue is empty but
has a non-zero tp->packets_out counter. tcp_send_loss_probe will call
tcp_rearm_rto which triggers NULL pointer reference by fetching the
retranmission queue head in its sub-routines.

Add a more detailed warning to help catch the root cause of the inflight
accounting inconsistency.
Reported-by: NRafael Tinoco <rafael.tinoco@linaro.org>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

If available rwnd is too small, tcp_tso_should_defer()
can decide it is worth waiting before splitting a TSO packet.

This really means we are rwnd limited.

Fixes: 5615f886 ("tcp: instrument how long TCP is limited by receive window")
Signed-off-by: NEric Dumazet <edumazet@google.com>
Acked-by: NSoheil Hassas Yeganeh <soheil@google.com>
Reviewed-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Previously the SNMP counter LINUX_MIB_TCPRETRANSFAIL is not counting
the TSO/GSO properly on failed retransmission. This patch fixes that.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Most linux hosts never setup TCP MD5 keys. We can avoid a
cache line miss (accessing tp->md5ig_info) on RX and TX
using a jump label.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

We can remove the loop and conditional branches
and compute wscale efficiently thanks to ilog2()
Signed-off-by: NEric Dumazet <edumazet@google.com>
Acked-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Jean-Louis reported a TCP regression and bisected to recent SACK
compression.

After a loss episode (receiver not able to keep up and dropping
packets because its backlog is full), linux TCP stack is sending
a single SACK (DUPACK).

Sender waits a full RTO timer before recovering losses.

While RFC 6675 says in section 5, "Algorithm Details",

   (2) If DupAcks < DupThresh but IsLost (HighACK + 1) returns true --
       indicating at least three segments have arrived above the current
       cumulative acknowledgment point, which is taken to indicate loss
       -- go to step (4).
...
   (4) Invoke fast retransmit and enter loss recovery as follows:

there are old TCP stacks not implementing this strategy, and
still counting the dupacks before starting fast retransmit.

While these stacks probably perform poorly when receivers implement
LRO/GRO, we should be a little more gentle to them.

This patch makes sure we do not enable SACK compression unless
3 dupacks have been sent since last rcv_nxt update.

Ideally we should even rearm the timer to send one or two
more DUPACK if no more packets are coming, but that will
be work aiming for linux-4.21.

Many thanks to Jean-Louis for bisecting the issue, providing
packet captures and testing this patch.

Fixes: 5d9f4262 ("tcp: add SACK compression")
Reported-by: NJean-Louis Dupond <jean-louis@dupond.be>
Tested-by: NJean-Louis Dupond <jean-louis@dupond.be>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

FQ pacing guarantees that paced packets queued by one flow do not
add head-of-line blocking for other flows.

After TCP GSO conversion, increasing limit_output_bytes to 1 MB is safe,
since this maps to 16 skbs at most in qdisc or device queues.
(or slightly more if some drivers lower {gso_max_segs|size})

We still can queue at most 1 ms worth of traffic (this can be scaled
by wifi drivers if they need to)

Tested:

# ethtool -c eth0 | egrep "tx-usecs:|tx-frames:" # 40 Gbit mlx4 NIC
tx-usecs: 16
tx-frames: 16
# tc qdisc replace dev eth0 root fq
# for f in {1..10};do netperf -P0 -H lpaa24,6 -o THROUGHPUT;done

Before patch:
27711
26118
27107
27377
27712
27388
27340
27117
27278
27509

After patch:
37434
36949
36658
36998
37711
37291
37605
36659
36544
37349
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

tcp_tso_should_defer() first heuristic is to not defer
if last send is "old enough".

Its current implementation uses jiffies and its low granularity.

TSO autodefer performance should not rely on kernel HZ :/

After EDT conversion, we have state variables in nanoseconds that
can allow us to properly implement the heuristic.

This patch increases TSO chunk sizes on medium rate flows,
especially when receivers do not use GRO or similar aggregation.

It also reduces bursts for HZ=100 or HZ=250 kernels, making TCP
behavior more uniform.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Acked-by: NSoheil Hassas Yeganeh <soheil@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

tcp_tso_should_defer() last step tries to check if the probable
next ACK packet is coming in less than half rtt.

Problem is that the head->tstamp might be in the future,
so we need to use signed arithmetics to avoid overflows.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Acked-by: NSoheil Hassas Yeganeh <soheil@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Applications using MSG_EOR are giving a strong hint to TCP stack :

Subsequent sendmsg() can not append more bytes to skbs having
the EOR mark.

Do not try to TSO defer suchs skbs, there is really no hope.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Acked-by: NSoheil Hassas Yeganeh <soheil@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

With EDT model, SRTT no longer is inflated by pacing delays.

This means that RTO and some other xmit timers might be setup
incorrectly. This is particularly visible with either :

- Very small enforced pacing rates (SO_MAX_PACING_RATE)
- Reduced rto (from the default 200 ms)

This can lead to TCP flows aborts in the worst case,
or spurious retransmits in other cases.

For example, this session gets far more throughput
than the requested 80kbit :

$ netperf -H 127.0.0.2 -l 100 -- -q 10000
MIGRATED TCP STREAM TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to 127.0.0.2 () port 0 AF_INET
Recv   Send    Send
Socket Socket  Message  Elapsed
Size   Size    Size     Time     Throughput
bytes  bytes   bytes    secs.    10^6bits/sec

540000 262144 262144    104.00      2.66

With the fix :

$ netperf -H 127.0.0.2 -l 100 -- -q 10000
MIGRATED TCP STREAM TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to 127.0.0.2 () port 0 AF_INET
Recv   Send    Send
Socket Socket  Message  Elapsed
Size   Size    Size     Time     Throughput
bytes  bytes   bytes    secs.    10^6bits/sec

540000 262144 262144    104.00      0.12

EDT allows for better control of rtx timers, since TCP has
a better idea of the earliest departure time of each skb
in the rtx queue. We only have to eventually add to the
timer the difference of the EDT time with current time.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Andrey reported the following warning triggered while running CRIU tests:

tcp_clean_rtx_queue()
...
	last_ackt = tcp_skb_timestamp_us(skb);
	WARN_ON_ONCE(last_ackt == 0);

This is caused by 5f6188a8 ("tcp: do not change tcp_wstamp_ns
in tcp_mstamp_refresh"), as we end up having skbs in retransmit queue
with a zero skb->skb_mstamp_ns field.

We could fix this bug in different ways, like making sure
tp->tcp_wstamp_ns is not zero at socket creation, but as Neal pointed
out, we also do not want that pacing status of a repaired socket
could push tp->tcp_wstamp_ns far ahead in the future.

So we prefer changing tcp_write_xmit() to not call tcp_update_skb_after_send()
and instead do what is requested by TCP_REPAIR logic.

Fixes: 5f6188a8 ("tcp: do not change tcp_wstamp_ns in tcp_mstamp_refresh")
Signed-off-by: NEric Dumazet <edumazet@google.com>
Reported-by: NAndrey Vagin <avagin@openvz.org>
Acked-by: NSoheil Hassas Yeganeh <soheil@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

When TCP implements its own pacing (when no fq packet scheduler is used),
it is arming high resolution timer after a packet is sent.

But in many cases (like TCP_RR kind of workloads), this high resolution
timer expires before the application attempts to write the following
packet. This overhead also happens when the flow is ACK clocked and
cwnd limited instead of being limited by the pacing rate.

This leads to extra overhead (high number of IRQ)

Now tcp_wstamp_ns is reserved for the pacing timer only
(after commit "tcp: do not change tcp_wstamp_ns in tcp_mstamp_refresh"),
we can setup the timer only when a packet is about to be sent,
and if tcp_wstamp_ns is in the future.

This leads to a ~10% performance increase in TCP_RR workloads.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

In commit fefa569a ("net_sched: sch_fq: account for schedule/timers
drifts") we added a mitigation for scheduling jitter in fq packet scheduler.

This patch does the same in TCP stack, now it is using EDT model.

Note that this mitigation is valid for both external (fq packet scheduler)
or internal TCP pacing.

This uses the same strategy than the above commit, allowing
a time credit of half the packet currently sent.

Consider following case :

An skb is sent, after an idle period of 300 usec.
The air-time (skb->len/pacing_rate) is 500 usec
Instead of setting the pacing timer to now+500 usec,
it will use now+min(500/2, 300) -> now+250usec

This is like having a token bucket with a depth of half
an skb.

Tested:

tc qdisc replace dev eth0 root pfifo_fast

Before
netperf -P0 -H remote -- -q 1000000000 # 8000Mbit
540000 262144 262144    10.00    7710.43

After :
netperf -P0 -H remote -- -q 1000000000 # 8000 Mbit
540000 262144 262144    10.00    7999.75   # Much closer to 8000Mbit target
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

sk_pacing_rate has beed introduced as a u32 field in 2013,
effectively limiting per flow pacing to 34Gbit.

We believe it is time to allow TCP to pace high speed flows
on 64bit hosts, as we now can reach 100Gbit on one TCP flow.

This patch adds no cost for 32bit kernels.

The tcpi_pacing_rate and tcpi_max_pacing_rate were already
exported as 64bit, so iproute2/ss command require no changes.

Unfortunately the SO_MAX_PACING_RATE socket option will stay
32bit and we will need to add a new option to let applications
control high pacing rates.

State      Recv-Q Send-Q Local Address:Port             Peer Address:Port
ESTAB      0      1787144  10.246.9.76:49992             10.246.9.77:36741
                 timer:(on,003ms,0) ino:91863 sk:2 <->
 skmem:(r0,rb540000,t66440,tb2363904,f605944,w1822984,o0,bl0,d0)
 ts sack bbr wscale:8,8 rto:201 rtt:0.057/0.006 mss:1448
 rcvmss:536 advmss:1448
 cwnd:138 ssthresh:178 bytes_acked:256699822585 segs_out:177279177
 segs_in:3916318 data_segs_out:177279175
 bbr:(bw:31276.8Mbps,mrtt:0,pacing_gain:1.25,cwnd_gain:2)
 send 28045.5Mbps lastrcv:73333
 pacing_rate 38705.0Mbps delivery_rate 22997.6Mbps
 busy:73333ms unacked:135 retrans:0/157 rcv_space:14480
 notsent:2085120 minrtt:0.013
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

In EDT design, I made the mistake of using tcp_wstamp_ns
to store the last tcp_clock_ns() sample and to store the
pacing virtual timer.

This causes major regressions at high speed flows.

Introduce tcp_clock_cache to store last tcp_clock_ns().
This is needed because some arches have slow high-resolution
kernel time service.

tcp_wstamp_ns is only updated when a packet is sent.

Note that we can remove tcp_mstamp in the future since
tcp_mstamp is essentially tcp_clock_cache/1000, so the
apparent socket size increase is temporary.

Fixes: 9799ccb0 ("tcp: add tcp_wstamp_ns socket field")
Signed-off-by: NEric Dumazet <edumazet@google.com>
Acked-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Previously TCP initial receive buffer is ~87KB by default and
the initial receive window is ~29KB (20 MSS). This patch changes
the two numbers to 128KB and ~64KB (rounding down to the multiples
of MSS) respectively. The patch also simplifies the calculations s.t.
the two numbers are directly controlled by sysctl tcp_rmem[1]:

  1) Initial receiver buffer budget (sk_rcvbuf): while this should
     be configured via sysctl tcp_rmem[1], previously tcp_fixup_rcvbuf()
     always override and set a larger size when a new connection
     establishes.

  2) Initial receive window in SYN: previously it is set to 20
     packets if MSS <= 1460. The number 20 was based on the initial
     congestion window of 10: the receiver needs twice amount to
     avoid being limited by the receive window upon out-of-order
     delivery in the first window burst. But since this only
     applies if the receiving MSS <= 1460, connection using large MTU
     (e.g. to utilize receiver zero-copy) may be limited by the
     receive window.

With this patch TCP memory configuration is more straight-forward and
more properly sized to modern high-speed networks by default. Several
popular stacks have been announcing 64KB rwin in SYNs as well.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NWei Wang <weiwan@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Reviewed-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Now TCP keeps track of tcp_wstamp_ns, recording the earliest
departure time of next packet, we can remove duplicate code
from tcp_internal_pacing()

This removes one ktime_get_tai_ns() call, and a divide.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

TCP keeps track of tcp_wstamp_ns by itself, meaning sch_fq
no longer has to do it.

Thanks to this model, TCP can get more accurate RTT samples,
since pacing no longer inflates them.

This has the nice effect of removing some delays caused by FQ
quantum mechanism, causing inflated max/P99 latencies.

Also we might relax TCP Small Queue tight limits in the future,
since this new model allow TCP to build bigger batches, since
sch_fq (or a device with earliest departure time offload) ensure
these packets will be delivered on time.

Note that other protocols are not converted (they will probably
never be) so sch_fq has still support for SO_MAX_PACING_RATE

Tested:

Test showing FQ pacing quantum artifact for low-rate flows,
adding unexpected throttles for RPC flows, inflating max and P99 latencies.

The parameters chosen here are to show what happens typically when
a TCP flow has a reduced pacing rate (this can be caused by a reduced
cwin after few losses, or/and rtt above few ms)

MIBS="MIN_LATENCY,MEAN_LATENCY,MAX_LATENCY,P99_LATENCY,STDDEV_LATENCY"
Before :
$ netperf -H 10.246.7.133 -t TCP_RR -Cc -T6,6 -- -q 2000000 -r 100,100 -o $MIBS
MIGRATED TCP REQUEST/RESPONSE TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to 10.246.7.133 () port 0 AF_INET : first burst 0 : cpu bind
 Minimum Latency Microseconds,Mean Latency Microseconds,Maximum Latency Microseconds,99th Percentile Latency Microseconds,Stddev Latency Microseconds
19,82.78,5279,3825,482.02

After :
$ netperf -H 10.246.7.133 -t TCP_RR -Cc -T6,6 -- -q 2000000 -r 100,100 -o $MIBS
MIGRATED TCP REQUEST/RESPONSE TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to 10.246.7.133 () port 0 AF_INET : first burst 0 : cpu bind
Minimum Latency Microseconds,Mean Latency Microseconds,Maximum Latency Microseconds,99th Percentile Latency Microseconds,Stddev Latency Microseconds
20,49.94,128,63,3.18
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Next patch will use tcp_wstamp_ns to feed internal
TCP pacing timer, so switch to CLOCK_TAI to share same base.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Switch internal TCP skb->skb_mstamp to skb->skb_mstamp_ns,
from usec units to nsec units.

Do not clear skb->tstamp before entering IP stacks in TX,
so that qdisc or devices can implement pacing based on the
earliest departure time instead of socket sk->sk_pacing_rate

Packets are fed with tcp_wstamp_ns, and following patch
will update tcp_wstamp_ns when both TCP and sch_fq switch to
the earliest departure time mechanism.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

TCP will soon provide earliest departure time on TX skbs.
It needs to track this in a new variable.

tcp_mstamp_refresh() needs to update this variable, and
became too big to stay an inline.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>