That takes care of the majority of ->sendmsg() instances - most of them
via memcpy_to_msg() or assorted getfrag() callbacks.  One place where we
still keep memcpy_fromiovecend() is tipc - there we potentially read the
same data over and over; separate patch, that...
Signed-off-by: NAl Viro <viro@zeniv.linux.org.uk>

patch is actually smaller than it seems to be - most of it is unindenting
the inner loop body in tcp_sendmsg() itself...

the bit in tcp_input.c is going to get reverted very soon - that's what
memcpy_from_msg() will become, but not in this commit; let's keep it
reasonably contained...

There's one potentially subtle change here: in case of short copy from
userland, mainline tcp_send_syn_data() discards the skb it has allocated
and falls back to normal path, where we'll send as much as possible after
rereading the same data again.  This patch trims SYN+data skb instead -
that way we don't need to copy from the same place twice.
Signed-off-by: NAl Viro <viro@zeniv.linux.org.uk>

One deployment requirement of DCTCP is to be able to run
in a DC setting along with TCP traffic. As Glenn Judd's
NSDI'15 paper "Attaining the Promise and Avoiding the Pitfalls
of TCP in the Datacenter" [1] (tba) explains, one way to
solve this on switch side is to split DCTCP and TCP traffic
in two queues per switch port based on the DSCP: one queue
soley intended for DCTCP traffic and one for non-DCTCP traffic.

For the DCTCP queue, there's the marking threshold K as
explained in commit e3118e83 ("net: tcp: add DCTCP congestion
control algorithm") for RED marking ECT(0) packets with CE.
For the non-DCTCP queue, there's f.e. a classic tail drop queue.
As already explained in e3118e83, running DCTCP at scale
when not marking SYN/SYN-ACK packets with ECT(0) has severe
consequences as for non-ECT(0) packets, traversing the RED
marking DCTCP queue will result in a severe reduction of
connection probability.

This is due to the DCTCP queue being dominated by ECT(0) traffic
and switches handle non-ECT traffic in the RED marking queue
after passing K as drops, where K is usually a low watermark
in order to leave enough tailroom for bursts. Splitting DCTCP
traffic among several queues (ECN and non-ECN queue) is being
considered a terrible idea in the network community as it
splits single flows across multiple network paths.

Therefore, commit e3118e83 implements this on Linux as
ECT(0) marked traffic, as we argue that marking all packets
of a DCTCP flow is the only viable solution and also doesn't
speak against the draft.

However, recently, a DCTCP implementation for FreeBSD hit also
their mainline kernel [2]. In order to let them play well
together with Linux' DCTCP, we would need to loosen the
requirement that ECT(0) has to be asserted during the 3WHS as
not implemented in FreeBSD. This simplifies the ECN test and
lets DCTCP work together with FreeBSD.

Joint work with Daniel Borkmann.

  [1] https://www.usenix.org/conference/nsdi15/technical-sessions/presentation/judd
  [2] https://github.com/freebsd/freebsd/commit/8ad879445281027858a7fa706d13e458095b595fSigned-off-by: NFlorian Westphal <fw@strlen.de>
Signed-off-by: NDaniel Borkmann <daniel@iogearbox.net>
Cc: Glenn Judd <glenn.judd@morganstanley.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Current behavior only passes RTTs from sequentially acked data to CC.

If sender gets a combined ACK for segment 1 and SACK for segment 3, then the
computed RTT for CC is the time between sending segment 1 and receiving SACK
for segment 3.

Pass the minimum computed RTT from any acked data to CC, i.e. time between
sending segment 3 and receiving SACK for segment 3.
Signed-off-by: NKenneth Klette Jonassen <kennetkl@ifi.uio.no>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

With TLP, the peer may reply to a probe with an
ACK+D-SACK, with ack value set to tlp_high_seq. In the current code,
such ACK+DSACK will be missed and only at next, higher ack will the TLP
episode be considered done. Since the DSACK is not present anymore,
this will cost a cwnd reduction.

This patch ensures that this scenario does not cause a cwnd reduction, since
receiving an ACK+DSACK indicates that both the initial segment and the probe
have been received by the peer.

The following packetdrill test, from Neal Cardwell, validates this patch:

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

+0    < S 0:0(0) win 32792 <mss 1000,sackOK,nop,nop,nop,wscale 7>
+0    > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 6>
+.020 < . 1:1(0) ack 1 win 257
+0    accept(3, ..., ...) = 4

// Send 1 packet.
+0    write(4, ..., 1000) = 1000
+0    > P. 1:1001(1000) ack 1

// Loss probe retransmission.
// packets_out == 1 => schedule PTO in max(2*RTT, 1.5*RTT + 200ms)
// In this case, this means: 1.5*RTT + 200ms = 230ms
+.230 > P. 1:1001(1000) ack 1
+0    %{ assert tcpi_snd_cwnd == 10 }%

// Receiver ACKs at tlp_high_seq with a DSACK,
// indicating they received the original packet and probe.
+.020 < . 1:1(0) ack 1001 win 257 <sack 1:1001,nop,nop>
+0    %{ assert tcpi_snd_cwnd == 10 }%

// Send another packet.
+0    write(4, ..., 1000) = 1000
+0    > P. 1001:2001(1000) ack 1

// Receiver ACKs above tlp_high_seq, which should end the TLP episode
// if we haven't already. We should not reduce cwnd.
+.020 < . 1:1(0) ack 2001 win 257
+0    %{ assert tcpi_snd_cwnd == 10, tcpi_snd_cwnd }%

Credits:
-Gregory helped in finding that tcp_process_tlp_ack was where the cwnd
got reduced in our MPTCP tests.
-Neal wrote the packetdrill test above
-Yuchung reworked the patch to make it more readable.

Cc: Gregory Detal <gregory.detal@uclouvain.be>
Cc: Nandita Dukkipati <nanditad@google.com>
Tested-by: NNeal Cardwell <ncardwell@google.com>
Reviewed-by: NYuchung Cheng <ycheng@google.com>
Reviewed-by: NEric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: NSébastien Barré <sebastien.barre@uclouvain.be>
Acked-by: NEric Dumazet <edumazet@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Signed-off-by: NAl Viro <viro@zeniv.linux.org.uk>

Signed-off-by: NAl Viro <viro@zeniv.linux.org.uk>

Commit c3ae62af ("tcp: should drop incoming frames without ACK
flag set") was created to mitigate a security vulnerability in which a
local attacker is able to inject data into locally-opened sockets by
using TCP protocol statistics in procfs to quickly find the correct
sequence number.

This broke the RFC5961 requirement to send a challenge ACK in response
to spurious RST packets, which was subsequently fixed by commit
7b514a88 ("tcp: accept RST without ACK flag").

Unfortunately, the RFC5961 requirement that spurious SYN packets be
handled in a similar manner remains broken.

RFC5961 section 4 states that:

   ... the handling of the SYN in the synchronized state SHOULD be
   performed as follows:

   1) If the SYN bit is set, irrespective of the sequence number, TCP
      MUST send an ACK (also referred to as challenge ACK) to the remote
      peer:

      <SEQ=SND.NXT><ACK=RCV.NXT><CTL=ACK>

      After sending the acknowledgment, TCP MUST drop the unacceptable
      segment and stop processing further.

   By sending an ACK, the remote peer is challenged to confirm the loss
   of the previous connection and the request to start a new connection.
   A legitimate peer, after restart, would not have a TCB in the
   synchronized state.  Thus, when the ACK arrives, the peer should send
   a RST segment back with the sequence number derived from the ACK
   field that caused the RST.

   This RST will confirm that the remote peer has indeed closed the
   previous connection.  Upon receipt of a valid RST, the local TCP
   endpoint MUST terminate its connection.  The local TCP endpoint
   should then rely on SYN retransmission from the remote end to
   re-establish the connection.

This patch lets SYN packets through the discard added in c3ae62af,
so that spurious SYN packets are properly dealt with as per the RFC.

The challenge ACK is sent unconditionally and is rate-limited, so the
original vulnerability is not reintroduced by this patch.
Signed-off-by: NCalvin Owens <calvinowens@fb.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Use the more common dynamic_debug capable net_dbg_ratelimited
and remove the LIMIT_NETDEBUG macro.

All messages are still ratelimited.

Some KERN_<LEVEL> uses are changed to KERN_DEBUG.

This may have some negative impact on messages that were
emitted at KERN_INFO that are not not enabled at all unless
DEBUG is defined or dynamic_debug is enabled.  Even so,
these messages are now _not_ emitted by default.

This also eliminates the use of the net_msg_warn sysctl
"/proc/sys/net/core/warnings".  For backward compatibility,
the sysctl is not removed, but it has no function.  The extern
declaration of net_msg_warn is removed from sock.h and made
static in net/core/sysctl_net_core.c

Miscellanea:

o Update the sysctl documentation
o Remove the embedded uses of pr_fmt
o Coalesce format fragments
o Realign arguments
Signed-off-by: NJoe Perches <joe@perches.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Ueki Kohei reported that when we are using NewReno with connections that
have a very low traffic, we may timeout the connection too early if a
second loss occurs after the first one was successfully acked but no
data was transfered later. Below is his description of it:

When SACK is disabled, and a socket suffers multiple separate TCP
retransmissions, that socket's ETIMEDOUT value is calculated from the
time of the *first* retransmission instead of the *latest*
retransmission.

This happens because the tcp_sock's retrans_stamp is set once then never
cleared.

Take the following connection:

                      Linux                    remote-machine
                        |                           |
         send#1---->(*1)|--------> data#1 --------->|
                  |     |                           |
                 RTO    :                           :
                  |     |                           |
                 ---(*2)|----> data#1(retrans) ---->|
                  | (*3)|<---------- ACK <----------|
                  |     |                           |
                  |     :                           :
                  |     :                           :
                  |     :                           :
                16 minutes (or more)                :
                  |     :                           :
                  |     :                           :
                  |     :                           :
                  |     |                           |
         send#2---->(*4)|--------> data#2 --------->|
                  |     |                           |
                 RTO    :                           :
                  |     |                           |
                 ---(*5)|----> data#2(retrans) ---->|
                  |     |                           |
                  |     |                           |
                RTO*2   :                           :
                  |     |                           |
                  |     |                           |
      ETIMEDOUT<----(*6)|                           |

(*1) One data packet sent.
(*2) Because no ACK packet is received, the packet is retransmitted.
(*3) The ACK packet is received. The transmitted packet is acknowledged.

At this point the first "retransmission event" has passed and been
recovered from. Any future retransmission is a completely new "event".

(*4) After 16 minutes (to correspond with retries2=15), a new data
packet is sent. Note: No data is transmitted between (*3) and (*4).

The socket's timeout SHOULD be calculated from this point in time, but
instead it's calculated from the prior "event" 16 minutes ago.

(*5) Because no ACK packet is received, the packet is retransmitted.
(*6) At the time of the 2nd retransmission, the socket returns
ETIMEDOUT.

Therefore, now we clear retrans_stamp as soon as all data during the
loss window is fully acked.

Reported-by: Ueki Kohei
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Signed-off-by: NMarcelo Ricardo Leitner <mleitner@redhat.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Tested-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This patch allows to set ECN on a per-route basis in case the sysctl
tcp_ecn is not set to 1. In other words, when ECN is set for specific
routes, it provides a tcp_ecn=1 behaviour for that route while the rest
of the stack acts according to the global settings.

One can use 'ip route change dev $dev $net features ecn' to toggle this.

Having a more fine-grained per-route setting can be beneficial for various
reasons, for example, 1) within data centers, or 2) local ISPs may deploy
ECN support for their own video/streaming services [1], etc.

There was a recent measurement study/paper [2] which scanned the Alexa's
publicly available top million websites list from a vantage point in US,
Europe and Asia:

Half of the Alexa list will now happily use ECN (tcp_ecn=2, most likely
blamed to commit 255cac91 ("tcp: extend ECN sysctl to allow server-side
only ECN") ;)); the break in connectivity on-path was found is about
1 in 10,000 cases. Timeouts rather than receiving back RSTs were much
more common in the negotiation phase (and mostly seen in the Alexa
middle band, ranks around 50k-150k): from 12-thousand hosts on which
there _may_ be ECN-linked connection failures, only 79 failed with RST
when _not_ failing with RST when ECN is not requested.

It's unclear though, how much equipment in the wild actually marks CE
when buffers start to fill up.

We thought about a fallback to non-ECN for retransmitted SYNs as another
global option (which could perhaps one day be made default), but as Eric
points out, there's much more work needed to detect broken middleboxes.

Two examples Eric mentioned are buggy firewalls that accept only a single
SYN per flow, and middleboxes that successfully let an ECN flow establish,
but later mark CE for all packets (so cwnd converges to 1).

 [1] http://www.ietf.org/proceedings/89/slides/slides-89-tsvarea-1.pdf, p.15
 [2] http://ecn.ethz.ch/

Joint work with Daniel Borkmann.

Reference: http://thread.gmane.org/gmane.linux.network/335797Suggested-by: NHannes Frederic Sowa <hannes@stressinduktion.org>
Acked-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDaniel Borkmann <dborkman@redhat.com>
Signed-off-by: NFlorian Westphal <fw@strlen.de>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Challenge ACK is described in RFC 5961, fix typo.
Signed-off-by: NSowmini Varadhan <sowmini.varadhan@oracle.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Signed-off-by: NStephen Hemminger <stephen@networkplumber.org>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

While testing upcoming Yaogong patch (converting out of order queue
into an RB tree), I hit the max reordering level of linux TCP stack.

Reordering level was limited to 127 for no good reason, and some
network setups [1] can easily reach this limit and get limited
throughput.

Allow a new max limit of 300, and add a sysctl to allow admins to even
allow bigger (or lower) values if needed.

[1] Aggregation of links, per packet load balancing, fabrics not doing
 deep packet inspections, alternative TCP congestion modules...
Signed-off-by: NEric Dumazet <edumazet@google.com>
Cc: Yaogong Wang <wygivan@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

We worked hard to improve tcp_ack() performance, by not accessing
skb_shinfo() in fast path (cd7d8498 tcp: change tcp_skb_pcount()
location)

We still have one spurious access because of ACK timestamping,
added in commit e1c8a607 ("net-timestamp: ACK timestamp for
bytestreams")

By checking if sk_tsflags has SOF_TIMESTAMPING_TX_ACK set,
we can avoid two cache line misses for the common case.

While we are at it, add two prefetchw() :

One in tcp_ack() to bring skb at the head of write queue.

One in tcp_clean_rtx_queue() loop to bring following skb,
as we will delete skb from the write queue and dirty skb->next->prev.

Add a couple of [un]likely() clauses.

After this patch, tcp_ack() is no longer the most consuming
function in tcp stack.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Cc: Willem de Bruijn <willemb@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Van Jacobson <vanj@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Suggested by Stephen. Also drop inline keyword and let compiler decide.

gcc 4.7.3 decides to no longer inline tcp_ecn_check_ce, so split it up.
The actual evaluation is not inlined anymore while the ECN_OK test is.
Suggested-by: NStephen Hemminger <stephen@networkplumber.org>
Signed-off-by: NDaniel Borkmann <dborkman@redhat.com>
Signed-off-by: NFlorian Westphal <fw@strlen.de>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

After Octavian Purdilas tcp ipv4/ipv6 unification work this helper only
has a single callsite.

While at it, convert name to lowercase, suggested by Stephen.
Suggested-by: NStephen Hemminger <stephen@networkplumber.org>
Signed-off-by: NDaniel Borkmann <dborkman@redhat.com>
Signed-off-by: NFlorian Westphal <fw@strlen.de>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

DataCenter TCP (DCTCP) determines cwnd growth based on ECN information
and ACK properties, e.g. ACK that updates window is treated differently
than DUPACK.

Also DCTCP needs information whether ACK was delayed ACK. Furthermore,
DCTCP also implements a CE state machine that keeps track of CE markings
of incoming packets.

Therefore, extend the congestion control framework to provide these
event types, so that DCTCP can be properly implemented as a normal
congestion algorithm module outside of the core stack.

Joint work with Daniel Borkmann and Glenn Judd.
Signed-off-by: NFlorian Westphal <fw@strlen.de>
Signed-off-by: NDaniel Borkmann <dborkman@redhat.com>
Signed-off-by: NGlenn Judd <glenn.judd@morganstanley.com>
Acked-by: NStephen Hemminger <stephen@networkplumber.org>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

The congestion control ops "cwnd_event" currently supports
CA_EVENT_FAST_ACK and CA_EVENT_SLOW_ACK events (among others).
Both FAST and SLOW_ACK are only used by Westwood congestion
control algorithm.

This removes both flags from cwnd_event and adds a new
in_ack_event callback for this. The goal is to be able to
provide more detailed information about ACKs, such as whether
ECE flag was set, or whether the ACK resulted in a window
update.

It is required for DataCenter TCP (DCTCP) congestion control
algorithm as it makes a different choice depending on ECE being
set or not.

Joint work with Daniel Borkmann and Glenn Judd.
Signed-off-by: NFlorian Westphal <fw@strlen.de>
Signed-off-by: NDaniel Borkmann <dborkman@redhat.com>
Signed-off-by: NGlenn Judd <glenn.judd@morganstanley.com>
Acked-by: NStephen Hemminger <stephen@networkplumber.org>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This patch adds a flag to TCP congestion algorithms that allows
for requesting to mark IPv4/IPv6 sockets with transport as ECN
capable, that is, ECT(0), when required by a congestion algorithm.

It is currently used and needed in DataCenter TCP (DCTCP), as it
requires both peers to assert ECT on all IP packets sent - it
uses ECN feedback (i.e. CE, Congestion Encountered information)
from switches inside the data center to derive feedback to the
end hosts.

Therefore, simply add a new flag to icsk_ca_ops. Note that DCTCP's
algorithm/behaviour slightly diverges from RFC3168, therefore this
is only (!) enabled iff the assigned congestion control ops module
has requested this. By that, we can tightly couple this logic really
only to the provided congestion control ops.

Joint work with Florian Westphal and Glenn Judd.
Signed-off-by: NDaniel Borkmann <dborkman@redhat.com>
Signed-off-by: NFlorian Westphal <fw@strlen.de>
Signed-off-by: NGlenn Judd <glenn.judd@morganstanley.com>
Acked-by: NStephen Hemminger <stephen@networkplumber.org>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This patch is a cleanup which follows the idea in commit e11ecddf (tcp: use
TCP_SKB_CB(skb)->tcp_flags in input path),
and it may reduce register pressure since skb->cb[] access is fast,
bacause skb is probably in a register.

v2: remove variable th
v3: reword the changelog
Signed-off-by: NWeiping Pan <panweiping3@gmail.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Our goal is to access no more than one cache line access per skb in
a write or receive queue when doing the various walks.

After recent TCP_SKB_CB() reorganizations, it is almost done.

Last part is tcp_skb_pcount() which currently uses
skb_shinfo(skb)->gso_segs, which is a terrible choice, because it needs
3 cache lines in current kernel (skb->head, skb->end, and
shinfo->gso_segs are all in 3 different cache lines, far from skb->cb)

This very simple patch reuses space currently taken by tcp_tw_isn
only in input path, as tcp_skb_pcount is only needed for skb stored in
write queue.

This considerably speeds up tcp_ack(), granted we avoid shinfo->tx_flags
to get SKBTX_ACK_TSTAMP, which seems possible.

This also speeds up all sack processing in general.

This speeds up tcp_sendmsg() because it no longer has to access/dirty
shinfo.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Now that tcp_dma_try_early_copy() is gone nothing ever sets
copied_early.

Also reverts "53240c20 tcp: Fix possible double-ack w/ user dma"
since it is no longer necessary.

Cc: Ali Saidi <saidi@engin.umich.edu>
Cc: James Morris <jmorris@namei.org>
Cc: Patrick McHardy <kaber@trash.net>
Cc: Eric Dumazet <edumazet@google.com>
Cc: Alexey Kuznetsov <kuznet@ms2.inr.ac.ru>
Cc: Hideaki YOSHIFUJI <yoshfuji@linux-ipv6.org>
Cc: Neal Cardwell <ncardwell@google.com>
Reported-by: NDave Jones <davej@redhat.com>
Acked-by: NDavid S. Miller <davem@davemloft.net>
Signed-off-by: NDan Williams <dan.j.williams@intel.com>

Per commit "77873803 net_dma: mark broken" net_dma is no longer used
and there is no plan to fix it.

This is the mechanical removal of bits in CONFIG_NET_DMA ifdef guards.
Reverting the remainder of the net_dma induced changes is deferred to
subsequent patches.

Marked for stable due to Roman's report of a memory leak in
dma_pin_iovec_pages():

    https://lkml.org/lkml/2014/9/3/177

Cc: Dave Jiang <dave.jiang@intel.com>
Cc: Vinod Koul <vinod.koul@intel.com>
Cc: David Whipple <whipple@securedatainnovations.ch>
Cc: Alexander Duyck <alexander.h.duyck@intel.com>
Cc: <stable@vger.kernel.org>
Reported-by: NRoman Gushchin <klamm@yandex-team.ru>
Acked-by: NDavid S. Miller <davem@davemloft.net>
Signed-off-by: NDan Williams <dan.j.williams@intel.com>

In order to make TCP more resilient in presence of reorders, we need
to allow coalescing to happen when skbs from out of order queue are
transferred into receive queue. LRO/GRO can be completely canceled
in some pathological cases, like per packet load balancing on aggregated
links.

I had to move tcp_try_coalesce() up in the file above tcp_ofo_queue()
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

icsk_rto is a 32bit field, and icsk_backoff can reach 15 by default,
or more if some sysctl (eg tcp_retries2) are changed.

Better use 64bit to perform icsk_rto << icsk_backoff operations

As Joe Perches suggested, add a helper for this.

Yuchung spotted the tcp_v4_err() case.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Now we no longer rely on having tcp headers for skbs in receive queue,
tcp repair do not need to build fake ones.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

tcp_collapse() wants to shrink skb so that the overhead is minimal.

Now we store tcp flags into TCP_SKB_CB(skb)->tcp_flags, we no longer
need to keep around full headers.
Whole available space is dedicated to the payload.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

We can allow a segment with FIN to be aggregated,
if we take care to add tcp flags,
and if skb_try_coalesce() takes care of zero sized skbs.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Input path of TCP do not currently uses TCP_SKB_CB(skb)->tcp_flags,
which is only used in output path.

tcp_recvmsg(), looks at tcp_hdr(skb)->syn for every skb found in receive queue,
and its unfortunate because this bit is located in a cache line right before
the payload.

We can simplify TCP by copying tcp flags into TCP_SKB_CB(skb)->tcp_flags.

This patch does so, and avoids the cache line miss in tcp_recvmsg()

Following patches will
- allow a segment with FIN being coalesced in tcp_try_coalesce()
- simplify tcp_collapse() by not copying the headers.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

After commit 740b0f18 ("tcp: switch rtt estimations to usec resolution"),
we no longer need to maintain timestamps in two different fields.

TCP_SKB_CB(skb)->when can be removed, as same information sits in skb_mstamp.stamp_jiffies
Signed-off-by: NEric Dumazet <edumazet@google.com>
Acked-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

TCP_SKB_CB(skb)->when has different meaning in output and input paths.

In output path, it contains a timestamp.
In input path, it contains an ISN, chosen by tcp_timewait_state_process()

Lets add a different name to ease code comprehension.

Note that 'when' field will disappear in following patch,
as skb_mstamp already contains timestamp, the anonymous
union will promptly disappear as well.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Acked-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Upon timeout, undo (via both timestamps/Eifel and DSACKs) was
disabled if any retransmits were still in flight.  The concern was
perhaps that spurious retransmission sent in a previous recovery
episode may trigger DSACKs to falsely undo the current recovery.

However, this inadvertently misses undo opportunities (using either
TCP timestamps or DSACKs) when timeout occurs during a loss episode,
i.e.  recurring timeouts or timeout during fast recovery. In these
cases some retransmissions will be in flight but we should allow
undo. Furthermore, we should only reset undo_marker and undo_retrans
upon timeout if we are starting a new recovery episode. Finally,
when we do reset our undo state, we now do so in a manner similar
to tcp_enter_recovery(), so that we require a DSACK for each of
the outstsanding retransmissions. This will achieve the original
goal by requiring that we receive the same number of DSACKs as
retransmissions.

This patch increases the undo events by 50% on Google servers.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Fix TCP FRTO logic so that it always notices when snd_una advances,
indicating that any RTO after that point will be a new and distinct
loss episode.

Previously there was a very specific sequence that could cause FRTO to
fail to notice a new loss episode had started:

(1) RTO timer fires, enter FRTO and retransmit packet 1 in write queue
(2) receiver ACKs packet 1
(3) FRTO sends 2 more packets
(4) RTO timer fires again (should start a new loss episode)

The problem was in step (3) above, where tcp_process_loss() returned
early (in the spot marked "Step 2.b"), so that it never got to the
logic to clear icsk_retransmits. Thus icsk_retransmits stayed
non-zero. Thus in step (4) tcp_enter_loss() would see the non-zero
icsk_retransmits, decide that this RTO is not a new episode, and
decide not to cut ssthresh and remember the current cwnd and ssthresh
for undo.

There were two main consequences to the bug that we have
observed. First, ssthresh was not decreased in step (4). Second, when
there was a series of such FRTO (1-4) sequences that happened to be
followed by an FRTO undo, we would restore the cwnd and ssthresh from
before the entire series started (instead of the cwnd and ssthresh
from before the most recent RTO). This could result in cwnd and
ssthresh being restored to values much bigger than the proper values.
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Fixes: e33099f9 ("tcp: implement RFC5682 F-RTO")
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

tcp_tw_recycle heavily relies on tcp timestamps to build a per-host
ordering of incoming connections and teardowns without the need to
hold state on a specific quadruple for TCP_TIMEWAIT_LEN, but only for
the last measured RTO. To do so, we keep the last seen timestamp in a
per-host indexed data structure and verify if the incoming timestamp
in a connection request is strictly greater than the saved one during
last connection teardown. Thus we can verify later on that no old data
packets will be accepted by the new connection.

During moving a socket to time-wait state we already verify if timestamps
where seen on a connection. Only if that was the case we let the
time-wait socket expire after the RTO, otherwise normal TCP_TIMEWAIT_LEN
will be used. But we don't verify this on incoming SYN packets. If a
connection teardown was less than TCP_PAWS_MSL seconds in the past we
cannot guarantee to not accept data packets from an old connection if
no timestamps are present. We should drop this SYN packet. This patch
closes this loophole.

Please note, this patch does not make tcp_tw_recycle in any way more
usable but only adds another safety check:
Sporadic drops of SYN packets because of reordering in the network or
in the socket backlog queues can happen. Users behing NAT trying to
connect to a tcp_tw_recycle enabled server can get caught in blackholes
and their connection requests may regullary get dropped because hosts
behind an address translator don't have synchronized tcp timestamp clocks.
tcp_tw_recycle cannot work if peers don't have tcp timestamps enabled.

In general, use of tcp_tw_recycle is disadvised.

Cc: Eric Dumazet <eric.dumazet@gmail.com>
Cc: Florian Westphal <fw@strlen.de>
Signed-off-by: NHannes Frederic Sowa <hannes@stressinduktion.org>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

ACK timestamps are generated in tcp_clean_rtx_queue. The TSO datapath
can break out early, causing the timestamp code to be skipped. Move
the code up before the break.
Reported-by: NDavid S. Miller <davem@davemloft.net>

Also fix a boundary condition: tp->snd_una is the next unacknowledged
byte and between tests inclusive (a <= b <= c), so generate a an ACK
timestamp if (prior_snd_una <= tskey <= tp->snd_una - 1).
Signed-off-by: NWillem de Bruijn <willemb@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Add SOF_TIMESTAMPING_TX_ACK, a request for a tstamp when the last byte
in the send() call is acknowledged. It implements the feature for TCP.

The timestamp is generated when the TCP socket cumulative ACK is moved
beyond the tracked seqno for the first time. The feature ignores SACK
and FACK, because those acknowledge the specific byte, but not
necessarily the entire contents of the buffer up to that byte.
Signed-off-by: NWillem de Bruijn <willemb@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This commit reduces spurious retransmits due to apparent SACK reneging
by only reacting to SACK reneging that persists for a short delay.

When a sequence space hole at snd_una is filled, some TCP receivers
send a series of ACKs as they apparently scan their out-of-order queue
and cumulatively ACK all the packets that have now been consecutiveyly
received. This is essentially misbehavior B in "Misbehaviors in TCP
SACK generation" ACM SIGCOMM Computer Communication Review, April
2011, so we suspect that this is from several common OSes (Windows
2000, Windows Server 2003, Windows XP). However, this issue has also
been seen in other cases, e.g. the netdev thread "TCP being hoodwinked
into spurious retransmissions by lack of timestamps?" from March 2014,
where the receiver was thought to be a BSD box.

Since snd_una would temporarily be adjacent to a previously SACKed
range in these scenarios, this receiver behavior triggered the Linux
SACK reneging code path in the sender. This led the sender to clear
the SACK scoreboard, enter CA_Loss, and spuriously retransmit
(potentially) every packet from the entire write queue at line rate
just a few milliseconds before the ACK for each packet arrives at the
sender.

To avoid such situations, now when a sender sees apparent reneging it
does not yet retransmit, but rather adjusts the RTO timer to give the
receiver a little time (max(RTT/2, 10ms)) to send us some more ACKs
that will restore sanity to the SACK scoreboard. If the reneging
persists until this RTO then, as before, we clear the SACK scoreboard
and enter CA_Loss.

A 10ms delay tolerates a receiver sending such a stream of ACKs at
56Kbit/sec. And to allow for receivers with slower or more congested
paths, we wait for at least RTT/2.

We validated the resulting max(RTT/2, 10ms) delay formula with a mix
of North American and South American Google web server traffic, and
found that for ACKs displaying transient reneging:

 (1) 90% of inter-ACK delays were less than 10ms
 (2) 99% of inter-ACK delays were less than RTT/2

In tests on Google web servers this commit reduced reneging events by
75%-90% (as measured by the TcpExtTCPSACKReneging counter), without
any measurable impact on latency for user HTTP and SPDY requests.
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Since Yuchung's 9b44190d (tcp: refactor F-RTO), tcp_enter_cwr is always
called with set_ssthresh = 1. Thus, we can remove this argument from
tcp_enter_cwr. Further, as we remove this one, tcp_init_cwnd_reduction
is then always called with set_ssthresh = true, and so we can get rid of
this argument as well.

Cc: Yuchung Cheng <ycheng@google.com>
Signed-off-by: NChristoph Paasch <christoph.paasch@uclouvain.be>
Acked-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

The undo code assumes that, upon entering loss recovery, TCP
1) always retransmit something
2) the retransmission never fails locally (e.g., qdisc drop)

so undo_marker is set in tcp_enter_recovery() and undo_retrans is
incremented only when tcp_retransmit_skb() is successful.

When the assumption is broken because TCP's cwnd is too small to
retransmit or the retransmit fails locally. The next (DUP)ACK
would incorrectly revert the cwnd and the congestion state in
tcp_try_undo_dsack() or tcp_may_undo(). Subsequent (DUP)ACKs
may enter the recovery state. The sender repeatedly enter and
(incorrectly) exit recovery states if the retransmits continue to
fail locally while receiving (DUP)ACKs.

The fix is to initialize undo_retrans to -1 and start counting on
the first retransmission. Always increment undo_retrans even if the
retransmissions fail locally because they couldn't cause DSACKs to
undo the cwnd reduction.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>